bluuebunny/crossref_metadata_2025 huggingface dataset:
PE modeling of upslope propagation data.
Authors: Thomson David J. , Heard Garry J. , Brooke Gary H.
Published: October 2005

The analysis and interpretation of underwater acoustic propagation data measured in regions of complex bathymetry (e.g., near continental slopes) can be enhanced by using a full-wave, range-dependent propagation model. Such models are particularly important at low frequencies, where sub-bottom penetration of sound can be significant. With the recent development of a two-way parabolic equation (PE) formalism [e.g., M. D. Collins etal., J. Acoust. Soc. Am. 90, 2277 (1991)], fast and accurate modeling of forward- and backscattered waves in range-dependent media can be realized. For propagation in an idealized wedge-shaped waveguide, the outgoing field computed with this two-way PE model compares well with the total field given by a more computationally intensive, coupled-mode model. In this paper, the formulation of a two-way PE model is used to interpret shot data recorded during an upslope experiment off the west coast of Canada. The shot waveforms were received on a 1200-m towed line array that opened range along a 15-km track. Over this endfire propagation run, the water depth decreased from 1500 to 500 m. The measured data were processed as a function of frequency, range, and vertical angle and compared to the forward-scattered component of the two-way PE field.
Wave field splitting, invariant imbedding, and phase space analysis applied to ocean acoustic wave propagation modeling.
Authors: Fishman Louis
Published: October 2005

Due to the extremely large size of realistic three-dimensional ocean problems, there is an understandable desire to incorporate marching methods into the solution algorithm for the inherently elliptic (frequency-domain) problem. Recognizing that typical ocean propagation problems are essentially scattering problems in terms of a transition region and transversely inhomogeneous asymptotic half-spaces, wave field splitting, invariant imbedding, and phase space methods reformulate the problem in terms of an operator scattering matrix characteristic of the transition region. The subsequent equations for the reflection and transmission operators are first order in range, nonlinear (Riccati-like), and, in general, nonlocal. The system is well-posed, but stiff. The reflected and transmitted wave fields can be computed in a very efficient manner, while the wave field in the transition region (if desired) can be computed by essentially a layer-stripping algorithm. In principle, the transition region can be divided into subregions, allowing for parallel computations and subsequent recombination. Numerical examples will be discussed. [Work supported by NSF, AFOSR, ONR, ASEE, JEWC.]
The inhibition of speech perception by linguistic categories.
Authors: Labov William
Published: October 2005

In a number of empirical studies it is reported that speakers consistently make a distinction between certain word classes, but categorize them as ‘‘the same’’ in perception tests. This paper reports experiments that explore the ability of Philadelphians to categorize and discriminate /er/–/Λr/ in ferry, furry, etc. A first series of categorization experiments showed that Philadelphians are significantly worse than non-Philadelphians in the ability to label and interpret this distinction. A second experiment combined tests of categorization and discrimination. Stimuli separated by 100-Hz differences in F2 were used to construct pairs in a 4IAX model, where subjects judged which of two pairs were different. Non-Philadelphians showed a linear psychophysical response, from near 100% for a three-step discrimination to 78% correct for a one-step discrimination. Philadelphians are much inferior in discrimination for the three-step test, but the difference disappears as discrimination becomes more difficult in the one-step test. The linguistic decision that two sounds are ‘‘the same’’ appears to interfere with discrimination to the extent that the task is recognized as a linguistic one. [Work supported by NSF.]
The effects of high-level blast wave exposure on hearing in the chinchilla.
Authors: Hamernik Roger P. , Ahroon William A. , Davis Robert I.
Published: October 2005

A summary of audiometric and histological results from 423 chinchillas that were exposed to high-level (150 to 160 dB peak SPL) blast waves produced by four different sources will be presented. The four sources, in an anechoic environment, produced Friedlander waveforms whose A-weighted energy spectra peaked at 0.25, 1.0, 2.0, or 4.0 kHz. The experimental data were organized to illustrate the effects of sound exposure level (SEL) and spectral energy on auditory system trauma. SEL was increased through increases in peak SPL and number of impulse presentations. The results show that the rate at which pathology (audiometric or histological) grows with increasing SEL is strongly influenced by both the different spectra of the four sources and the spectral details of an individual source. The results can be interpreted to indicate that, for a given octave band, equivalent amounts of energy do not produce equivalent auditory effects but are influenced by the dominant energy-carrying band of the blast wave. [Work supported by USAARL.]
Frequency-time and wave-number-space processing tools for experimental studies of wave propagation in shells.
Authors: Williams Earl G.
Published: October 2005

Experimental measurements on point-driven, fluid-loaded cylindrical shells in the frequency region 0<ka<11 are processed using near-field acoustical holography and wave vector filtering in order to expose the details of wave motion on the structure. Signal processing tools along with the creation of various filters in the k-ω space domain are used to study impulsive wave packets as a function of space and time excited by a point drive inside the shell. This technique provides a very powerful tool for the study of experimental data. Special emphasis will be given to the frequency range that has not presented before, 4<ka<11. Significant is the evidence of shear and longitudinal waves for many circumferential orders. Signal processing methods are presented that include generalized time frequency representations (GTFR) used in the k–z domain, which provide the ability to display wave conversion at discontinuities, such as the ends of the shell. Of particular importance in this frequency range is the radiation from the near field of the point driver, as the WVF processing dramatically shows.
Estimation of plate-mode transfer functions near discontinuities via radon transform decomposition.
Authors: Fricke J. Robert
Published: October 2005

This paper considers the estimation of complex coupling coefficients, or transfer functions, that relate an incident plate wave encountering an impedance discontinuity to the resulting scattered waves. The discontinuity may be modeled as a linear-time-invariant (LTI) filter while the incident and scattered wave represent the filter input and output, respectively. Using the radon transform, the input and output plate modes are isolated in p-tau space, which facilitates the computation of the transfer function via Fourier transforms and complex division. An example is discussed showing the flexural-to-flexural transfer function that occurs at the end of a truncated steel plate. [Research supported by ONR.]
Scattering of impulse signal by elastic shells with internal structures.
Authors: Guo Yue-Ping
Published: October 2005

Sound scattering from an elastic shell can be significantly affected by structures loaded inside the shell. A major effect of internal loading has been shown to be the conversion of subsonic flexural waves in the shell into strongly radiating acoustic waves. In the frequency domain, this leads to a deeply scalloped scattering form function with the empty-shell result as its ‘‘mean value.’’ Since flexural waves in the shell are much slower than supersonic compressional leaky waves that dominate the scattering process for an empty shell, the conversion of flexural waves into sound by internal structural loading is well separated in time from other processes such as the initial ‘‘bubble-like’’ reflection and the phase-matched leaky wave re-radiation. Also, because of the dispersive nature of the flexural waves, the internal loading effect in time domain can be clearly identified; they distinguish themselves in the scattered field by a series of wave packets whose amplitudes decay due to both acoustic damping and dispersion. This is in contrast to the other two main components in the scattered time signal, one being the initial reflection that is basically the inverse of the incident signal and the other being the leaky wave contribution consisting of decaying pulses. All these are clearly demonstrated in this paper, which examines the scattering of impulse incidence from elastic shells with internal structures. The response of the loaded shell is also examined, which further reveals the interactions between the acoustic field, the shell, and its internals. [Work supported by ONR.]
Radiated sound energy of a fluid-loaded cylinder as a function of force time duration and multiple pulse time spacing.
Authors: Kruger Daniel H. , Mann J. Adin
Published: October 2005

Transfer functions measured by near-field acoustical holography (NAH) are used to simulate the sound radiated by a point-driven, fluid-loaded cylindrical shell with spherical end caps in the time domain for a variety of force inputs [J. Acoust. Soc. Am. 90, 1656–1664 (1991)]. The accuracy of time and frequency domain processing is reviewed. Radiated sound energy as a function of the time duration of a single pulse driving force is studied. The results are partially explained by looking at the output sound power as a function of frequency. Radiated sound energy as a function of the time spacing between multiple pulses is also studied. Sound radiation from the entire shell, the ends, and the driver alone are used to explain results. Frequencies determined by modal analysis as structurally resonant, acoustically radiating, both resonant and radiating, and neither resonant nor radiating were simulated for both single and multiple pulse forces. Instantaneous acoustic intensity, instantaneous power, and the time integral of these quantities are analyzed. Conclusions on the importance of time-domain analysis are made.
An update on the field of otoacoustic emissions.
Authors: Lonsbury-Martin Brenda L.
Published: October 2005

Otoacoustic emissions or sounds generated by the cochlea have been known for almost 15 yr. In that time, the beneficial features of using emissions as response measures of hearing function have been well-realized. Today, the field of otoacoustic emissions has expanded to include three major areas of study that can be described as basic, clinical, and technical applications of the technique. The primary purpose of this presentation is to provide an update on the contributions that otoacoustic emissions have made to our understanding of the basis of cochlear transduction, the screening of hearing and the differential diagnoses of hearing problems, and the straightforward monitoring of peripheral processes including the functional integrities of the acoustic reflex, ‘‘cochlear amplifier,’’ and cochlear-efferent system. The relevance of the work of Eberhard Zwicker and his colleagues on otoacoustic emissions will be used as a framework within which to discuss the current status of otoacoustic emissions in the hearing sciences. [Work supported by NIH.]
Modifications to nonlinear processing for recruitment of loudness.
Authors: Rutledge Janet C. , Tungthangthum Apichat
Published: October 2005

The time-varying frequency-dependent processing (TVFD) [J. C. Rutledge and M. A. Clements, Proc. IEEE ICASSP, 3641–3644 (1991)] was designed to adjust to changing speech input on both time and frequency basis. Due to inherent problems with the processing technique, some spectral smearing occurs [J. Acoust. Soc. Am. 89, 1974 (A) (1991)]. The processing has been modified to maintain the time-varying frequency-dependent nature, but reduces the smearing effect. A sinusoidal speech model is used with the compensation processing. Appropriate gains for the sinusoid amplitudes are calculated every 5 ms. In the modified processing scheme, the gain determined for the largest peak in each frequency band is applied to all components in that band. This maintains the relative amplitude ratio between neighboring peaks in the frequency spectrum. The compensation algorithm is presented here along with preliminary performance results based on the objective measures described in another paper at this meeting.
Auditorium acoustics simulation for the Macintosh.
Authors: Rieffel Marc , Everbach E. Carr
Published: October 2005

Geometrical ray-tracing techniques have been used as educational and analysis tools in a Macintosh program written in ThinkTM C. This program takes as an input file a standard DXF-format file created from any of several popular CAD programs. From this data, a wire-frame representation of the auditorium is displayed and the user is able to specify the acoustic reflection characteristics of each surface. The program then allows the user to specify the position, frequency characteristics, and directionality of sound sources in the hall, and to run a ray-tracing simulation represented in 3-D color graphics of the behavior of sound rays emanating for the sources. Diffraction effects and air attenuation are accounted for in the propagation algorithm. Sound level contours through arbitrary planes are available after the simulation terminates, as are a variety of acoustic descriptors for user-specified listener locations. Hopefully this normal-mode analysis for steady excitations can be incorporated into newer versions of the program.
Measurements of nonlinear phenomena in a vibrating wire.
Authors: Hanson Roger J. , Anderson James M. , Macomber H. Kent
Published: October 2005

If tensional changes and longitudinal motion of a vibrating string or wire are taken into account, the usual simple theory of vibrating strings is replaced by a complex theory involving coupled nonlinear partial differential equations. The resulting motion is correspondingly complex. Experimental measurements have been made of the motion of a harpsichord wire driven electromagnetically in a fixed direction perpendicular to the equilibrium position of the wire. Near resonance the amplitude of the motion perpendicular to the driving direction can become nearly as large as in the driving direction and harmonics of the driving frequency can be pronounced. The outputs of the two orthogonal optical motion detectors were fed to a dual-channel frequency analyzer and an X–Y oscilloscope. Measurements were made of frequency, amplitude, and phase of the motions in directions parallel and perpendicular to the driving force. Results will be presented for the nonlinear motion that has been observed for amplitudes of 5 mm down to less than a micron on a wire 0.7 m long. The larger amplitude motions will be exhibited on a video tape.
Remarks on the reed–air column system as coupled oscillators.
Authors: Weinreich Gabriel
Published: October 2005

The combination of a reed (outward- or inward-striking) and a resonant air column represents a pair of oscillators which mutually influence each other in a non-Hermitian way, allowing one to look for eigenfrequencies of the composite system and for criteria that give their imaginary parts the sign that corresponds to growing oscillations. This approach, although elementary, does not appear to have been used before. The computation is complicated by the fact that the imaginary part is a rather small fraction of the frequency, so that the usual reduction of a quartic to a quadratic secular equation provides at best a first approximation. Experiments are being designed to test these results for the simple case of an air ‘‘column’’ with but a single resonance; it is also expected that these calculations will be helpful in computer simulations of wind instrument systems. [Work supported by NSF.]
A high-resolution fundamental frequency determination based on phase changes of the FFT.
Authors: Brown Judith C.
Published: October 2005

The constant Q transform described recently [J. C. Brown and M. S. Puckette, ‘‘A Real Time Constant Q Transform,’’ submitted to IEEE Trans. Signal Process. (1992)] has been adapted so that it is suitable for tracking the fundamental frequency of extremely rapid musical passages. For this purpose a frame size of 25 ms or less is required; thus the calculation described previously has been modified so that it is constant resolution rather than constant Q for the lower frequency bins. This calculation as modified serves as the input for a fundamental frequency tracker similar to that described by Brown [J. C. Brown, ‘‘Musical Pitch Tracking using a Pattern Recognition Method,’’ submitted to J. Acoust. Soc. Am. (1992)]. This frequency tracker is designed for musical applications and is based on pattern recognition of the log frequency representation of a sound with harmonic frequency components. Once the fundamental frequency has been chosen by the tracker, an approximation for the phase change in the FFT for a time advance of one sample was used to obtain an extremely precise value for this frequency. Graphical examples are given for musical passages by a violin executing vibrato and glissando. Here the instrumental frequency changes are continuous, and an extremely accurate value is necessary for a precise determination of the fundamental frequency.
Nonoccupational noise exposure: Is it a problem?
Authors: Johnson Daniel L.
Published: October 2005

If the differences of hearing levels between males and females, as indicated in either the 1960–62 or 1971–75 Public Health Service Reports, are compared with the noise induced permanent threshold shifts (NIPTS) as predicted by ISO1999, these differences fit the NIPTS expected from an exposure of the males to 8 h levels from 89 to 93 dB. Since the number of male workers exposed to such noise is expected to be but a fraction of the total male population, nonoccupational noise exposure must be the most important contribution to this difference. Supporting figures and analysis will be presented to make this point. The role of dosimetry and noise exposure questionnaires in defining nonoccupational noise will also be discussed.
Assessment of the volume of environment noise SNL CNL and their utilization.
Authors: Cong Lin
Published: October 2005

In this paper the author has analyzed the relation between human subjective sense and various environmental noise level indexes and conducted a comprehensive study of the correlation between different sound levels and subjective annoying rates using the principle of applied mathematics and experimental psychology. Comprehensive indexes in sound environmental assessment, SNL and CNL, are introduced and the method of limited isolated sound level indexes has been changed into the calculation of psychological effect of sound over humans as assessment volume in environmental noise assessment so that the assessment index can reflect directly psychologically acceptable limits of humans to surrounding sound. This shows the percentage of people affected by noise under the surrounding sound. Meanwhile, the author also gives a detailed description of the application of the assessment index to the analysis and prediction of sound environmental quality, of construction project planning, and of the advantages of the assessment index.
Acoustic properties of materials with reduced Poisson ratio.
Authors: Narang Subhash C. , Nigam Asutosh
Published: October 2005

Reduced Poisson ratio materials present an interesting opportunity for altering acoustic properties. Relationship of material morphology with Poisson ratio will be explored. The effect of Poisson ratio on mechanical properties will be reviewed. Relationship with moduli, flexural rigidity, and fracture toughness will be examined. Reduced Poisson materials have low cutoff frequencies and low wave dispersion constants compared to analogous conventional materials. Development of a variety of materials for specific applications will be discussed. The generic approach to the fabrication of reduced Poisson ratio materials, such as foams, honeycombs, laminates, and composites, provides application engineers added flexibility in choosing appropriate materials.
A robust transfer matrix formulation for ultrasonic wave propagation in multilayered absorbing media.
Authors: Lévesque Daniel , Piché Luc
Published: October 2005

An improved version of the transfer matrix approach is presented where general expressions are obtained for reflection and transmission coefficients in either fluid or solid half-space and problems associated with numerical stability are solved efficiently. The formulation is applicable to longitudinal and shear input wave alike, at arbitrary incidence angles and for any sequence of solid or fluid layers. Also, allowance is made for viscoelastic behavior by means of relaxation functions and the response to arbitrary incident pulse shapes and beam profiles is obtained through a two-dimensional numerical Laplace inversion. For application to bond quality inspection, the interface between any two layers is handled by introducing a very thin interphase layer, rigidly bonded to its neighbors, with adjustable viscoelastic properties. In the case of a metal–polymer–metal structure, simulations show resonance splitting, which measures coupling strength between layers and is found to be very sensitive to small changes in the interfacial conditions. These theoretical results are confirmed by experiments and good estimates are given for the specific stiffness of the interface region.
Compressional waves in an anisotropic porous material: CELCOR.
Authors: Hsu Chaur-Jian , Plona Thomas J.
Published: October 2005

Ultrasonic experiments were made to study the compressional waves that propagate in a fluid-saturated, porous, anisotropic material, CELCOR, made by Corning. This material can be described as a regular packing of hollow cylinders, each with a square cross section. The material is anisotropic, both in the plane containing the cylinder axis and in the plane perpendicular to the cylinder axis. This cell structure has square openings of typically 0.05 in. and ceramic wall thicknesses of 0.01 in. The porosity of the system is about 70%–80%. Along the cylinder axis, two compressional waves are observed, one traveling near the fluid speed and one traveling near the ceramic speed. In the plane perpendicular to the cylinder axis, the compressional wave has an angular velocity variation consistent with the square-lattice nature of the cross section. Ultrasonic results of velocity and attenuation (centered at 500 kHz) will be shown for the cases when the material is saturated either with water or air. The two compressional wave behavior parallel to the cylinder axis will be interpreted in terms of the Biot theory.
The directional scattering factor: Experimental determination.
Authors: D’Antonio Peter
Published: October 2005

This paper suggests experimental procedures to determine a three-dimensional directional scattering factor (DSF) or directivity balloon, which fully represents the backscattered energy from any acoustical surface of arbitrary size, as a function of the frequency and direction cosine of the incident sound. The DSF is necessary in acoustical design, computer room modeling programs, and auralizations. D’Antonio [P. D’Antonio and J. Konnert, 79th AES Convention, Preprint No. 2295 (October 1985)] introduced a boundary measurement technique for evaluating directional scattering from absorptive, reflective, and diffusive surfaces using a pressure-zone microphone and the TEF analyzer. A more comprehensive automated polar mapping technique is suggested. This approach is capable of yielding the full 3-D backscattered directivity balloon using either the impulse response, transfer function, or intensity. A test sample approximately 20 m2 in size is placed on the floor boundary, a hemispherical or flat grid of appropriately spaced microphones is placed above this at approximately 5 m, and the test signal sound source is suspended 5 m above the microphone grid. The microphone grid records the backscattered energy at an appropriate angular increment for each angle of incidence. These data are then analyzed and corrected for the effects of finite sample area, speaker and microphone responses, and near-field/far-field differences.
Effect of a co-flowing annular outer flow on the flow and acoustics in a porous tube.
Authors: Ramamoorthy Sundar , Khodabakhsh Fariborz , Munukutla Sastry
Published: October 2005

One of the most important aspects of flow in a gas turbine combustor is the cooling airflow introduced through the combustor liner. The co-flowing annular cooling air affects the flow and the acoustic field of the main combustor. A generic study is in progress to study the effect of a co-flowing annular outer flow on the flow and acoustics in a porous tube. This work is an idealization of the actual gas turbine combustor flow. The results generated here will be used to validate the computational codes currently being used by the gas turbine industry to calculate these flow fields. In the present experimental work, a 6-in.-diam tube made out of perforated sheet is located coaxially in an 8-in.-diam outer tube. Airflows in the inner perforated tube, as well as in the annular space between the two tubes. Detailed measurements of the turbulence structure using hot wire anemometry and of the acoustic field using microphone transducers are being made. Effects of parameters such as porosity of the tube, relative areas of annular space and cross section of inner tube, and flow Reynolds number on the turbulence quantities and the acoustic field will be reported.
A maximal-length sequence technique for estimating the coefficients of a pole-zero parametric model of an acoustical system.
Authors: Kovitz Paul S. , Swanson David C.
Published: October 2005

A technique is described for estimating pole-zero models of real acoustical systems using an excitation signal known as a maximal-length sequence (MLS). The MLS is useful because for any sequence length, the autocorrelation (and corresponding power spectrum) is nearly the same as that for an infinite-length white-noise sequence. The pole-zero model for the system can be used to simulate the response of the actual acoustic system to any input. In addition, a pole-zero filter model can be synthesized from theoretical descriptions of the underlying physics such as Green’s functions. Therefore, pole-zero modeling of acoustic systems provides a convenient way to compare theory and experiment. An example is presented consisting of a one-dimensional waveguide excited by a loudspeaker that is driven by a MLS signal.
Application of a new signal processing technique in a multibeam sounder.
Authors: Chakraborty Bishwajit , Raju Y. Satayanarayana
Published: October 2005

The currently used beamforming method for multibeam sounding application is a delay-sum method, which is also known as a Fourier spectral estimation method applicable to directional estimation. But array size is one major difficulty that exists with this technique. A solution to such a problem is the application of a ‘‘high-resolution technique’’ for the beamforming method. In this paper beamforming techniques like the maximum likelihood method (MLM) and the maximum entrophy method (MEM) are presented for multibeam sounding application. The size of the array is chosen to be one-fourth and one-half of the conventional array at an input signal-to-noise ratio of 0 or 10 dB. Both high-resolution techniques are found to be suitable for multibeam sounding use at a 10-dB signal-to-noise ratio (input), even for a small array size of 16 elements. But relatively wide beamwidths for the 45° direction, when the input signal-to-noise ratio is 0 dB, does not support its utility for both techniques. The lowest value of the standard deviation angular error for MEM shows its preference over the MLM technique.
Effects of rate and bite-block manipulations on relationships between duration and variability in children’s and adults’ speech.
Authors: Smith Bruce L.
Published: October 2005

Based on the assumption that speakers are less skilled at talking under atypical conditions, segment duration and variability were studied in 5 and 11 yr olds and a group of adults as they produced short phrases in normal, fast-rate, and bite-block conditions. Preliminary analyses indicate that for all three groups, the bite-block task generally did not cause much difference in duration or variability from the normal condition. Although duration and variability both decreased in the fast-rate condition for all groups, relative changes in variability were typically less than those for duration. Several other comparisons of duration and variability also did not show very strong relationships between these two measures. For instance, variability was also evaluated as a function of intrinsic vowel duration. Although high vowels were approximately 30% shorter than low vowels for the three groups, variability was only about 15% less for high than for low vowels. For individual subjects, correlations between duration and variability were also generally rather low. Implications for speech production development will be discussed.
Age-related differences in processing dynamic acoustic cues to initial stops.
Authors: Fox Robert Allen , Gokcen Jeanne , Wall Lida G.
Published: October 2005

The present study examines possible age-related differences in the ability to process rapidly changing acoustic information in the identification and discrimination of initial stop consonants. As described in Fox etal. [J. Acoust. Soc. Am. 89, 1935(A) (1991)], older adults show lower accuracy rates in identifying both consonants and vowels in so-called ‘‘silent-center’’ tokens. Such results are compatible with the suggestion that older listeners have greater difficulty than younger listeners in processing dynamic acoustic cues, such as those represented by consonant transitions. Two sets of stimuli were created: one set represented a 7-step [beh]–[deh] continuum with a 40-ms transition followed by a 150-ms steady-state vowel; the second set was composed of a 7-step [beh]–[deh] continuum with the same transition followed by a 10-ms vowel. There were two sets of normal hearing listeners including 15 listeners aged 20–24 and 15 listeners aged 60–75. There was an identification task and two AX discrimination tasks (with either a 500- or 2000-ms ISI). Older listeners showed shallower slopes in their identification functions than did the younger listeners, particularly in the stimuli composed primarily of formant transitions. In the discrimination task, older listeners were significantly less sensitive to formant transition differences than were the younger listeners, but there were no significant differences related to interstimulus interval. [Work supported, in part, by an NIA grant to R. Fox.]
Articulatory compensation in hearing-impaired speakers.
Authors: Campbell Melanie M. , Boothroyd Arthur , McGarr Nancy S. , Harris Katherine S.
Published: October 2005

Previous studies have shown that normal adults compensate for a bite block at the onset of vowel production. It has thus been presumed that auditory feedback plays no role in on-line, speech-motor control. Some investigators have further concluded that while acoustic feedback may play some role, it is not a substantial component in the coordination of reciprocal articulatory movement. The role of auditory feedback is explored in this investigation of compensatory skills in oral subjects with congenital hearing loss. Six severely hearing-impaired and six profoundly hearing-impaired adults were recorded. The stimuli were repetitions of a carrier phrase containing a target word with one of three vowels (/i/, /i/, /ae/) in mixed, randomly selected sequences in four conditions: normal (with hearing aids), masking noise (without hearing aids), bite block (with hearing aids), bite block plus masking noise (without hearing aids). Compensation was measured by comparison of the first three vowel formants of each test condition with those of the normal condition. Formants were acoustically analyzed using LPC techniques. Preliminary data suggest that the first few tokens produced with bite block result in some changes in F0 and loudness. However, there is not convincing evidence of failure of compensation. [This work was supported by NINCDS Grant No. DC00121-29.]
Articulatory and acoustic study of consonant production in a traumatically brain injured speaker.
Authors: Bell-Berti Fredericka , Rossetti Maria , Oshima Kiyoshi , Coelho Carl
Published: October 2005

This study examined the clinical utility of instrumental studies of the speech of neurologically impaired persons, as well as documented the particular articulatory patterns of one individual, a right-handed 36-yr-old female, 5-yr post onset of TBI, with right hemiparesis. Communicative impairments included relatively mild cognitive deficits and moderate-to-severe hypokinetic dysarthria. The speech sample included repetitions of isolated sustained continuant consonants, of VCV disyllables, and of sentences containing consonant clusters. The articulatory data were electropalatographic (EPG) signals recorded using a RION soft artificial palate with a sampling rate of 64 fps. The acoustic signals were recorded simultaneously with the EPG signals and subjected to spectral analysis. Vowel environment was found to have a strong influence on lingual contact patterns for /s/. In general, within vowel height groups, the extent of lingual contact for /s/ was greater in the VCV context than in words produced in sentences. Spectral patterns associated with differences in lingual contact for /s/ will be described. [Work supported by NIH Grant No. DC-00121 to Haskins Lab.]
Holographic array processing in a range-dependent ocean.
Authors: Al-Kurd Azmi A. , Porter Robert P. , Mourad Pierre D.
Published: October 2005

Imperfect knowledge of the salient characteristics of the propagation medium limits the performance of acoustic array processors at long ranges in the ocean. Holographic and phase conjugation techniques can be used to diminish the range-integrated effect of the medium and reconstruct the wave front in the vicinity of a scatterer or other signal source. Then, using a back-propagation technique that focuses sound at the position of the unknown source, the location of the source can be determine. In this paper, the WKB approximation is implemented for a range-dependent ocean with both N2 bilinear and exponential stratification of the sound-speed profile. The results of the analysis and simulation show the possibility of localizing a source at large distance with great accuracy.
Acoustic monitoring of beams by time-frequency analysis.
Authors: Robin Pierre Y. , Jansen Ben H. , Cheng Xiu-Ting , Finch Robert D.
Published: October 2005

Acoustic vibrations can be used to monitor the structural integrity of metallic beams. A current technique involves impacting a beam with a hammer and analyzing the induced vibrations. However, the generated vibrations exhibit transient and nonstationary characteristics such as rapid decay of amplitude and time-dependent frequency content. This observation limits the use of classical spectral analysis which requires stationarity. Time-frequency representations, such as the Wigner distribution, provide a more general framework to analyze transient and nonstationary signals. Actual beam vibrations, analyzed with time-frequency representations, will be presented. In particular, it will be shown that this analysis enhances the appearance and disappearance with time of specific frequency components as well as changes of the resonant frequencies. These observations can ultimately be used as discriminative features in a pattern recognition scheme classifying good and cracked beams. [Work supported by NSF Grant No. MSS-9024224.]
Transient acoustic wave modeling with higher-order WKBJ asymptotic expansions and symbolic manipulation.
Authors: Verweij Martin D.
Published: October 2005

A combination of the WKBJ asymptotic expansion and the Cagniard–De Hoop method is used to determine the total space-time domain acoustic wave field in a continuously layered configuration. In the transform domain, a recurrence scheme for the coefficients of the WKBJ asymptotic expansion is derived from the integral equations of the problem. This scheme is well suited for implementation in a symbolical manipulation program, e.g., Mathematica. The WKBJ asymptotic expansion is transformed back to the space-time domain by applying the Cagniard–De Hoop method in a particularly efficient way. For various configurations numerical results are presented, which show that the presented approach yields accurate results for instants ranging up to many times the arrival time.
Variations on E. Zwicker’s theme of frequency discrimination.
Authors: Zwislocki Jozef J.
Published: October 2005

Eberhard Zwicker once suggested that human frequency discrimination was controlled by the apical cutoff of cochlear excitation rather than by the maximum of excitation [E. Zwicker and R. Feldtkeller, DasOhralsNachrichtenempfänger (Hirzel, Stuttgart, 1967)]. Physiological evidence supporting Zwicker’s suggestion has been obtained. Microelectrode recordings from the cochlear inner and outer hair cells have revealed that the sound frequency associated with their maximum excitation at a given cochlear location changes with SPL, whereas the subjective pitch remains nearly constant. In Mongolian gerbils, at the 2-kHz cochlear location, the frequency is lowered by about an octave when the SPL is increased from 20 to 80 dB [J. J. Zwislocki, Acta Otolaryngol. 111, 256–262 (1991)]. Over the same SPL range, the human pitch changes by only a few percent. A similar near invariance in the high-frequency cutoff of the hair-cell excitation has been found. As a result of the known spatial frequency mapping in the cochlea, this cutoff corresponds to the apical cutoff of hair-cell excitation. The results suggest that the cutoff rather than the excitation maximum constitutes the place code for pitch, especially since the excitation maximum is not sharp enough to account for the great sensitivity of humans and animals to frequency changes. This is consistent with Zwicker’s hypothesis. [Work supported by NIDCD.]
The number and nature of OAE generators.
Authors: Norton Susan J. , Stover Lisa J.
Published: October 2005

Professor Zwicker immediately appreciated the importance of David Kemp’s discovery of otoacoustic emissions (OAE), and devoted much of his efforts in recent years to their study. OAEs are sounds generated by nonlinear, highly tuned, physiologically vulnerable biomechanical elements within the cochlea. They can be measured in the external ear canal, and thus provide an exquisite noninvasive tool for studying cochlear mechanics, particularly the ‘‘cochlear amplifier,’’ invivo. There are several OAE phenomena, distinguished primarily by the stimuli used to evoke them. While all OAEs are generally absent if there is a cochlear hearing loss involving absence of outer hair cells, how these phenomena relate to one another and normal cochlear function is not entirely clear. For a given evoking stimulus, OAEs differ in their ‘‘thresholds,’’ growth rate and maximum output across audiologically normal ears. In addition, within an ear there are frequencies and presumably cochlear regions with relatively stronger emissions than others. Similarly, within an ear different OAE phenomena in the same frequency region may have different thresholds, growth rates, and maximum output. Such differences within and across ears and OAE phenomena may reflect differences in generators or may be the result of procedural differences which require some discussion before more esoteric issues can be addressed. [Work supported by NIDCD and DRF.]
Intelligent signal processing for source location in anisotropic materials.
Authors: Chittayil Karthikeyan
Published: October 2005

Active as well as passive source location methods are commonly used in industry, in nondestructive testing (NDT) of materials and structures. Most of these methods cater to materials or media that are isotropic; methods for source location in anisotropic media are the subject of a few recent investigations [Castagnede etal., J. Acoust. Soc. Am. 86, 1161 (1989); Steinberg etal., J. Acoust. Soc. Am. 90, 2081 (1991)]. In the present work, a method of source location in anisotropic media is introduced, which uses the learning ability of a neural-networklike computational algorithm to model the source location problem. During the learning phase, the correlation of the source location with the multisensor data is determined and stored in a ‘‘memory matrix.’’ This matrix is then used for the prediction of the location of any unknown source using multisensor input data, by the method of autoassociative recall. The results of the application of this algorithm, to a specific problem of estimating the location of a source in a composite material plate, is presented and discussed.
On the possibility of delaying shock formation in finite amplitude sound beams through alteration of the source signal.
Authors: Ginsberg Jerry H. , Too Gee-Pin
Published: October 2005

It was pointed out by Rudnick [J. Acoust. Soc. Am. 30, 564–567 (1958)] that if a plane wave is generated in the form of a backward sloping wave through appropriate phasing of the higher harmonics, then the amplitude of the fundamental frequency received at a given location would be enhanced. Burns [J. Acoust. Soc. Am. 45, 210–213 (1969)] further noted that such a wave would form a shock at a farther distance than that for monochromatic input at the level of the fundamental. Rogers [private communication (1987)] suggested that this effect would be especially beneficial for a sound beam. If the region for shock formation could be moved from the near field, where the signal is quasiplanar, to the far field, where it propagates in a spherical manner, then the result would be a substantial enhancement in the shock formation distance, because of the slower rate at which nonlinear effects accumulate in the far field. The present work uses the NPE code modified for sound beams to test this possibility. The input for the analysis is a baffled piston whose vibration consists of a fundamental at fixed amplitude plus a second harmonic at arbitrary amplitude and relative phase. Results for the fundamental and second harmonic at various locations indicate that the improvement is negligible because of the relative phase shift associated with transition from near-field to far-field behavior.
Fractionation of fine particle suspensions by coordinated planar ultrasonic and laminar flow fields in a rectangular acoustic chamber.
Authors: Mandralis Z. I. , Feke D. L.
Published: October 2005

Acoustics fields are being used to manipulate fine particles (0.1–20 μm) in liquid suspensions held within a thin rectangular channel. Ultrasonic fields are applied across the channel, resulting in particle migration in this direction. Particles with different physical characteristics (size, density, compressibility, shape) respond with different speeds to the sound field [Z. I. Mandralis and D. L. Feke, Fluid/Part. Sep. J. 3, 115–121 (1990)], resulting in partial separation across the channel. These partial separations are amplified into useful separations by applying cyclic, coordinated, bidirectional laminar flows. The response of the particles to the standing wave field will be described. A mathematical model that simulates the fractionation scheme has been developed. Results from the simulation demonstrate the effectiveness of the technique. Experimental results from a batch fractionation of 325 mesh polystyrene particles are presented. Scaleup and continuous operation schemes will be discussed.
Active control of radiation from vibrating structures.
Authors: Sommerfeldt Scott D.
Published: October 2005

In many applications, one faces the challenge of controlling the sound field that is produced by a vibrating structure. There are two general approaches that can be taken to control such a sound field. The first approach involves directly controlling the acoustic field by means of additional acoustic sources, such as loudspeakers, which are controlled in such a way so as to minimize the noise in the acoustic field. The second approach involves controlling the acoustic field indirectly, in that one can control the structural vibration field in a manner so as to minimize the radiation from the structure. This paper will outline some of the possible methods of using active vibration control of the structure so as to minimize the acoustic radiation from the structure. The physical mechanisms that lead to acoustic radiation and that can be controlled by means of an active vibration control system will be discussed. Research issues related to each active control approach, and the conditions that may make one particular approach for implementing active control preferable over other choices will also be discussed.
Active suppression of radiated sound from a vibrating structure using adaptive modal control.
Authors: Thi James , Morgan Dennis R.
Published: October 2005

Active control for silencing sound and vibration has recently become the subject of much investigation and is now being enhanced by the advent of adaptive filtering techniques. If the active control system is to suppress the sound field at locations where acoustic measurements are available, then the adaptive filtered-x least-mean-square (LMS) algorithm can be used. An alternative approach that does not require acoustic measurements controls the vibrational energy so as to minimize the radiated acoustic field. There are two approaches that can be used to achieve this. The first approach suppresses the total vibrational energy by minimizing an objective function that represents either the sum of the output sensor powers or the sum of all dominant vibrational modal powers. The second approach employs an adaptive modal control system to suppress selected high-efficiency radiating modes from the vibrating structure. In this paper, these two methods are compared and illustrated by numerical example for a simply supported vibrating cantilever beam. Calculations of vibrational energy and acoustic energy with and without control are presented and compared for both approaches. [Work supported in part by ONR/DARPA.]
Stabilization of feedback-type active control systems in finite ducts.
Authors: Swanson David C.
Published: October 2005

Use of feedback-type control architectures in active noise cancellation can be very difficult due to the extreme sensitivity of this approach to instability, but are sometimes advantageous because the need for the reference sensor is eliminated. Instability results when the feedback loop time delay causes an additional phase shift of 180° at some frequencies allowing high-gain positive feedback to occur. The classical approach to this problem is to design a compensation filter which reduces the feedback gain to less than unity at the frequency(s) where the loop delay causes instability. In a finite-length duct, the feedback instabilities are found to be related to the transducers and the distance between them as well as their positions within the duct. The potential for instability is present whether adaptive or fixed active noise control is done using a feedback-type controller. The compensation filter design is determined directly from the physical parameters of the finite waveguide and the transducers. However, the performance of the feedback controller is restricted by the compensation filter.