bluuebunny/crossref_metadata_2025 huggingface dataset:
On formulation of the high-order boundary conditions for stratified media
Authors: Osipov Audrey V.
Published: October 2005

Boundary conditions containing spatial derivatives of high orders, Â(∂x,∂y)(∂U/∂z)+ikB̂(∂x, ∂y)U‖z=0=0, where z is the coordinate orthogonal to a plane surface of a layered medium, Â, B̂ are the differential operators with respect to the tangent coordinates, U is a scalar function representing acoustic pressure in the region above the surface (z≥0), k is the wave number, are widely used to describe the complicated process of interaction of sound waves with thin elastic plates or stratified slabs. The high-order boundary conditions are usually obtained from the equations of motion of the elastic structures by expanding the wave fields into power series according to the slab thickness. This paper describes a new approach to formulation of the high-order boundary conditions for isotropic stratified media. Via Fourier transform the problem is rigorously reduced the approximation of certain entire functions by polynomials in the complex plane. Choosing an appropriate method of approximation such boundary conditions are constructed that are specially adjusted to model the structure of interest. The technique proposed is regular; it does not require the separation of symmetrical and antisymmetrical processes, and it includes the slim layer theories as a special case. For structures placed between two fluid half-spaces, the scalar boundary conditions must be replaced by matrix ones combining the values of U and its derivatives on the opposite sides of the slab.
The contribution of a reduced visual image to speech perception in noise
Authors: Johnson Jennifer A. , Rosenblum Lawrence D. , Saldaña Helena M.
Published: October 2005

It has long been known that seeing a talker’s face can improve the perception of speech in noise [A. MacLeod and Q. Summerfield, Br. J. Audiol. 21, 131–141]. Yet little is known about which characteristics of the face are useful for embellishing the degraded signal. Recently, a point-light technique has been adopted to help isolate the salient aspects of a visible articulating face [Saldaña etal., J. Acoust. Soc. Am. 92, 2340(A) (1992)]. In this technique, a speaker’s face is darkened and reflective dots are arranged on the lips, teeth, tongue, cheeks, and jaw. The actor is videotaped speaking in the dark so that when shown to subjects, only the moving dots are seen. In order to determine whether these reduced images could contribute to the perception of degraded speech, noise-embedded sentences were dubbed with point-light images at various signal-to-noise ratios. It was found that these images could improve comprehension depending on the number and location of points used. Implications of these results for theories of audiovisual integration, models of lip-reading, and telecommunications systems will be discussed. [Work supported by NSE.]
Using a multi-talker database to identify sentence- and talker-dependent correlates of speech intelligibility: Preliminary results
Authors: Bradlow Ann R. , Pisoni David B.
Published: October 2005

Intelligibility tests of a database of 2000 recorded sentences (100 sentences produced by 20 talkers), show considerable variation in intelligibility across both sentences and talkers. This paper presents preliminary results of a study designed to identify the lexical and talker characteristics that correlate with speech intelligibility scores. To identify sentence-specific correlates of intelligibility, lexical characteristics of the words that comprise the most versus the least intelligible sentences were compared. Results show that high-intelligibility sentences have a higher proportion of function to content words, and consequently, a higher mean word frequency than low-intelligibility sentences. Furthermore, lexical characteristics of the most common word-level substitution errors indicate the importance of word frequency in overall sentence intelligibility. To identify talker-specific correlates of intelligibility, acoustic measurements of sentences that show the most inter-speaker variability in intelligibility were compared. For example, durational characteristics of the most common segment-level errors indicated that the relative timing between segments is an important aspect of talker intelligibility. Future directions and general implications of this database project will be discussed. [Work supported by NIH Training Grant No. T32 DC-00012 and NIH Research Grant No. R01 DC-00111.]
Beyond visemes: Using disemes in synthetic speech with facial animation
Authors: Henton Caroline
Published: October 2005

Realistic on-screen computer ‘‘assistants’’ need synthetic visible speech that has accurate, pleasing articulation; they also need to run in real-time on a personal computer. Previous visible speech systems have used 9–32 ‘‘visemes,’’ minimal contrastive units of visible articulation. Viseme-based animation has been choppy, inaccurate, and insufficiently plastic. In contrast, concatenative speech synthesis utilizes a large inventory of diphone units. Therefore, two improvements are needed: expansion beyond simple visemes, and reduction of the number of diphones so that they may be mapped to the facial images in real-time. To this end, visible and articulatory archiphones and diphone ‘‘aliases’’ were formalized, using standard phonological distinctive features, and a system of disemes was created. Disemes begin during one viseme (phone) in an archiphonic family and end somewhere during the following viseme (phone), in another archiphonic family. In this way, many transitions that occur in approximately 1800 diphones (for General American English) can be visually depicted by the same diseme, due to their similarity in lip, teeth, and tongue image positions. The effectiveness of mapping these disemes may be demonstrated, using a variety of non-screen agents and faces.
Perceptual assimilation of voiceless Mandarin fricatives and affricates by American English speakers
Authors: Ling Xiange , Strange Winifred
Published: October 2005

This study investigated how voiceless Mandarin consonants (MC) are assimilated to American English consonants (EC) by monolingual English adult listeners. Nine Mandarin fricatives and affricates that are articulated in middle alveolar, blade-palatal, and back palatal regions were used. Subjects listened to a sequence of Mandarin CV syllables (produced by two native Mandarin speakers) and responded by selecting the most similar English consonants. The main results were: (1) MC’s tended to be assimilated to consonant categories with the same place of articulation and manner. However, place of articulation tended to take priority over manner class. (2) When there was not an identical place of articulation, categorization was influenced by the vowel context and the results were more variable. Further, the MC tended to be categorized to an adjacent place of articulation within the same manner class. (3) Voiceless aspirated MC’s were almost always assimilated to voiceless EC categories, while voiceless unaspirated MC’s were assimilated to voiced EC categories. Graphs of EC categorization frequencies for each MC will be presented. [Work supported by NIDCD.]
Three new diagnostic intelligibility tests for communication systems
Authors: Voiers William D.
Published: October 2005

In one form or another, the diagnostic rhyme test (DRT) has been in use for 30 years. This paper describes the latest version of the DRT, DRT-V, and two other diagnostic intelligibility tests based on the same principles and having the same general format as the DRT. These are the diagnostic medial consonant test (DMCT) and a new version of the diagnostic alliteration test (DALT-II). All three tests use the two-alternative, forced-choice paradigm to measure the discriminability of six distinctive features in consonant phonemes. They differ in the position of the equivocal phoneme, which, respectively, occurs initially, intervocalically, and terminally. In all three tests, vowel environment is systematically varied, and the discriminability of each critical feature is tested under systematically varied conditions of other, noncritical, features in the equivocal phoneme. These design characteristics provide a diversity of diagnostic scoring options. Comparative results are presented for selected types of speech degradation.
Acoustic scattering from immersed axisymmetric objects. Cylindrical shell limited by hemispherical endcaps
Authors: Décultot Dominique , Lecroq Florence , Maze Gérard , Ripoche Jean
Published: October 2005

The study of the acoustic scattering from axisymmetric objects is based on the works that have been previously carried out in this field for objects with ‘‘simple’’ shapes (plates, cylinders, cylindrical shells). The objects under investigation consist of a cylindrical shell limited by two hemispherical shells. The experimental results presented in the form of resonance spectra are obtained by the MIIR (method of isolation and identification of resonances). A complete study for various excitation angle allows one to locate particular directions where the resonant phenomenon is important. For axial excitation, the results are interpreted by means of the condition of resonance formation (deduced from the phase matching) in the meridian plane of the objects. This condition takes into account the phase velocities of the waves propagating on the hemispherical parts and on the cylindrical part. For normal excitation experimental results are compared to those obtained from infinite cylindrical shell. Other particular directions of excitation are highlighted in this study. These results obtained for axisymmetric objects (stainless steel) allow the understanding of the influence of the shape of the objects on the surface wave propagation [Décultot etal., J. Acoust. Soc. Am. 94, 2916–2923 (1993)].
Influence of the incidence angle on the scattering by a circular cylindrical shell
Authors: Maze Gérard , Léon Fernand , Veksler Naum , Ripoche Jean
Published: October 2005

The resonance scattering theory (RST) developed by Flax etal. [J. Acoust. Soc. Am. 63, 723–731 (1978)] allows one to explain the structure of the spectra at scattering by a cylindrical shell at normal incidence. The resonances are related to the circumferential waves propagating around the shell. In this presentation, the problem of scattering of an obliquely incident plane acoustic wave by a circular cylindrical shell is considered. The resonance spectra obtained by the MIIR ‘‘in propagation’’ show resonance peaks related to the A, S0, T0, and T1 waves. The wide deep dips observed on the reflection spectra are particularly analyzed using the calculation of modal resonance components. These dips are attributed to a wave with modal resonance components having a small Q factor and situated sufficiently near each other with an important overlap. The position of dips depends on the incidence angle. With incidence angles larger than the second critical angle, resonance peaks are clearly observed on the reflection spectra that are related to the A wave and wide dips related to modal resonance components of the second type of the A0 wave; when the incidence angle increases the central frequency of these dips decreases.
Coupling coefficient for leaky waves on thick spherical shells from elasticity theory
Authors: Kargl Steven G. , Marston Philip L.
Published: October 2005

An approximation of leaky wave scattering amplitudes for thick spherical shells in water was previously demonstrated for backscattering and (through the optical theorem) the total cross section [S. G. Kargl and P. L. Marston, J. Acoust. Soc. Am. 88, 1103–1113 (1990); 89, 2545–2558 (1991)]. In the convention of that formulation, it was sufficient to approximate the complex leaky wave coupling coefficient Gl for the thick shell by 8πβlc/cl over the frequency range considered. In the present work, calculations are given pertaining to that approximation that are based on full elasticity theory and a generalization of Watson transformation results for solid spheres. The computations [S. G. Kargl, Ph.D. dissertation, Wash. State Univ. (1990)] show that while ‖Gl‖ is well approximated as noted, φl=arg(Gl) tends to negative values for low ka [roughly in proportion to (ka)−1] for both the s0 wave and the supersonic region of the a0 wave. At large ka, φl becomes small in agreement with the approximation. Over a range of ka above the coincidence frequency the φl(ka) curves for the a0 and s0 waves are similar and ‖φl‖<1 rad. [Work supported by ONR.]
Resonance scattering theory for strongly overlapping resonances
Authors: Guicking D. , Peine H.
Published: October 2005

The classification of submerged bodies with respect to internal properties (material, etc.), rather than geometrical shape, should be possible by means of their scattering resonances. This approach is known as the resonance scattering theory (RST). An essential extension of the RST theory with respect to solving practical problems is possible by applying to the acoustical scattering problem the so-called R-matrix theory (RMT), which has been developed by Wigner and Eisenbud to describe nuclear reactions. This theory includes in a natural way the linewidths, the coupling of the different resonant modes, and the overlapping of the resonances. Properly defined resonances provide the basis from which the whole formalism is constructed, while in the former theory the resonances are introduced after computing the scattering cross section. This allows an extension of the ‘‘acoustical spectroscopy’’ of isolated resonances as introduced by Brill to the practically important case of strongly overlapping resonances. However, isolated resonances are included in the RMT theory as special cases as well. A model function is proposed that can be matched—without knowledge of the scattering cross section—to measured differential cross sections in order to extract the resonance features.
Acoustics of the hammered dulcimer, its history, and recent developments
Authors: Peterson David R. , Rossing Thomas D. , Canfield Gregory H.
Published: October 2005

Hammered dulcimers (e.g., santir, psanterim, psalterion) have been known in the Near East for about 5000 years. From the Near East, the instrument traveled west to Europe and east to China and Korea. The most elaborate of dulcimers is the cimbalom, developed in Hungary in the 19th century and used by Hungarian gypsies as a concert instrument. Brought from England to the New World in colonial times, the hammered dulcimer became popular in New England, the Appalachians, the lumber camps of Maine and Michigan, and the southern states. The hammered dulcimer, an ancestor of the piano, has changed significantly in the last 15 years. Since the instrument does not have mechanical dampers, builders have attempted to reduce decay times by changing bridge caps, slotting bridges, reducing weight and stiffness, and reducing soundboard impedance. Changes in string arrangements and soundboard bracing have resulted in stronger sound in the bass. Unlike a piano, in which the hammers consistently strike the same position on the strings, the hammered dulcimer player strikes the strings at varying distances from the bridge. During the brief contact time, this creates a short string segment that drives the bridge at frequencies considerably higher than the string fundamental. Thus the attack characteristics and spectral frequencies are greatly influenced by the hardness of the hammer and the strike position.
String vibrations in hammered dulcimers
Authors: Peterson David R.
Published: October 2005

A unique characteristic of the hammered dulcimer is that the middle bridge splits string courses into a length ratio of 3:2, so the two string segments have an interval of a musical fifth. When the longer segment is struck, coupling through the bridge to the shorter segment causes every third harmonic to have two modes of vibration with slightly different frequencies, which interact to produce vibrations with slowly varying amplitudes, which are determined by bridge impedance, decay rates, and slight tuning differences. A theoretical model has been developed, which applies to any number of strings tuned to rational multiples. The distinct sound of the hammered dulcimer depends upon the initial transient, which is strongly influenced by the hardness of the hammer and the strike position.
Hearing aids, noise, and the FDA
Authors: Villchur Edgar
Published: October 2005

Recent statements by the Food and Drug Administration imply that the only way a hearing aid can improve speech recognition in noise is to reduce the relative level of the noise. Hearing-aid circuits have not been designed that can effectively separate speech from noise, especially from the most common type of noise, competing speech. There is, however, experimental evidence—cited in this paper—that signal processing that restores lost speech cues to the hearing impaired can improve their speech recognition in any kind of noise, even though the signal-to-noise ratio is degraded. With such processing, hearing-impaired persons can better afford to lose cues to noise because they start out with more cues. They retain more of the redundant information in speech after masking has taken its toll, which helps them separate desired from undesired signals (a process that Broadbent calls, in normal listeners, ‘‘selective listening.’’) [D. E. Broadbent, PerceptionandCommunication (Pergamon, New York, 1958)]. A recording of speech plus interference, processed to simulate recruitment, will be played both with and without compensation processing to illustrate how speech intelligibility can be improved in the face of increased relative noise.
The role of amplitude modulation in auditory motion perception
Authors: Strybel Thomas Z.
Published: October 2005

When two discrete sound sources are presented in succession and appropriately timed, a single sound image is heard to move across the gap between the sources. Research in this area of auditory motion perception has shown that motion is heard even under monaural listening conditions [Strybel etal., Percept. Psychophys. 45, 371–377 (1989)], suggesting that motion perception may not require the detection of a change in location over time. The present experiment tested this idea by producing apparent motion from stimuli which were emitted from a single loudspeaker. The stimuli consisted of sine waves (0.5, 1, and 5 kHz) of 100–250 ms duration. Here 10–150 ms after signal onset, the amplitude of the signal was either increased or decreased at a rate of 0.6 dB/ms for a period of 10 ms. Four subjects were instructed to classify their perception of the stimuli into one of five categories. Two of these categories were for moving sounds and two were for stationary sounds. The fifth category was used when the listener’s perception did not fit into any of the other categories. Continuous motion was reported at all frequencies only when an amplitude change occurred 60–100 ms after signal onset.
Binaural hearing aid algorithms for noisy and reverberant environments
Authors: Peissig J. , Kollmeier B.
Published: October 2005

In noisy and reverberant acoustical environments the human hearing system is capable of concentrating on a desired speaker, while attenuating undesired sounds like noise or reverberation. This so called ‘‘cocktail-party processing’’ is reduced in hearing-impaired persons. New hearing aid strategies try to imitate binaural processing principles to suppress interfering noise or reverberation in order to yield maximum speech intelligibility for the impaired listener. Two algorithms are described that reduce unwanted signal energies by attenuating sound that arises from the sides (lateral suppression) and that remove reverberation and short reflections. They were combined and implemented in real time on a multiprocessor setup to allow for adaptive adjustment of parameters and an optimal fit to the individual hearing loss. The noise reduction performance of the algorithms was tested with normal- and hearing-impaired subjects. Subjective speech quality and speech intelligibility assessments as well as speech intelligibility measurements show that both groups benefit significantly from the preprocessing strategies in anechoic and reverberant acoustical situations with interfering speakers.
Detection of a mistuned component in a harmonic complex
Authors: Lee Jungmee , Green David M.
Published: October 2005

The ability to detect a mistuned component in a harmonic complex was measured as a function of harmonic number using three different stimulus procedures. The harmonic complex and target component were presented either simultaneously to the same ear (simultaneous monotic), successively to the same ear (successive monotic), or simultaneously to different ears (simultaneous dichotic). For harmonic numbers above the third, the thresholds obtained in the simultaneous-monotic procedure were much lower than those obtained in the other two procedures. It is argued that detection in the simultaneous-monotic procedure is primarily based on a cue associated with the lower-frequency components in the power spectrum of the stimulus envelope. The envelope cue seemed to be weakened by either decreasing the stimulus duration from 410 ms or increasing the fundamental frequency from 200 Hz. Thresholds did not differ for successive-monotic and simultaneous-dichotic procedures, suggesting that the detection of the mistuned component was mediated above the level of the cochlea. [Work supported by NIH.]
Intrasubject reliability of pure-tone thresholds
Authors: Zhou Bin , Green David M.
Published: October 2005

The effect of standing waves in the ear canal on the intrasubject reliability of pure-tone thresholds was investigated. Sound was transmitted to the ear canal through a hard-wall source and a 3-cm plastic tube. This created a standing wave field between the hard-wall source and the eardrum. In response to a broadband stimulus, the measured sound pressure at the entrance to the ear canal showed several maxima and minima for frequencies between 7000 and 14 000 Hz. Thresholds were measured at three frequencies where successive maxima and minima occurred and also at 1000 Hz. Four subjects were tested. The intrasubject reliability of the thresholds was tested across five sessions with the hard-wall source located at the same position or moved by as much as ±0.5 cm. Thresholds were not significantly different across the sessions when the hard-wall source had the same position. The movement of the hard-wall source across the five sessions resulted in significant changes in the high-frequency thresholds (±3 dB), but not in the low-frequency thresholds (±0.5 dB). [Research supported by NIH.]
The effects of age and gender on the leakage of dB SPL from beneath binaural headphones
Authors: Bilger R. C. , Brestal K. L. , Lesniak J. M. , Petrow K. L. , Meyer T. A.
Published: October 2005

The purpose of the present study was to investigate the possibility that the low-frequency leakage of sound from beneath supra-aural earphone cushions increases as adults age, giving the appearance of a low-frequency hearing loss. The subjects were four groups (n=6) defined by age (younger than 30 and older than 65 years) and gender. The probes for ER-7C microphones were placed 0.5 in. into each of a subject’s ear canals. Continuous pure tones at 11 1/3-oct frequencies (100–1000 Hz) were presented via TDH-39 headphones mounted in MX-41/AR cushions. Tones were presented at 80 dB SPL. The procedure was replicated once. The variable analyzed was the difference between sound pressure level measured in the ear canal and that measured in an NBS 9A coupler. The low-frequency leak was present in our results in that the average leak increased at a rate of 12 dB/oct, beginning at 400 Hz. The slope of the leak, however, was different for right and left ears and was different for the four age-by-sex groups. These results strongly suggest that caution be used in interpreting sensitivity differences obtained with supra-aural earphones. [Work supported by DC00174. Etymotic Research loaned us one of the two ER-7 microphones used.]
The time course of effects on intensity discrimination following monaural fitting of hearing aids
Authors: Robinson Ken , Gatehouse Stuart
Published: October 2005

Systematic effects on intensity discrimination occur in listeners who have experienced long-term unilateral amplification [Robinson and Gatehouse, Br. J. Audiol. (in press)]. For a bandlimited tone complex centered at 3 kHz, the normally aided ear was more sensitive than the unaided ear at levels greater than 85 dB SPL, yet was less sensitive at lower levels. No effects were found at 250 Hz. Intensity discrimination for these same stimuli was prospectively investigated for six listeners with bilateral symmetric sensorineural hearing impairment following provision of unilateral amplification. Weber functions for both ears were similar at both frequencies prior fitting, and 4 to 12 weeks post-fitting. From 4 to 12 weeks, the normally aided ear became more sensitive than the unaided ear for the 3-kHz stimulus presented at 95 dB SPL. No effects were observed at 250 Hz, or for the unaided ear. Hence, changes in intensity coding occur in the normally aided ear following unilateral amplification, and exhibit similar characteristics to the changes observed for speech identification in noise [Gatehouse, J. Acoust. Soc. Am. 92, 1258–1268 (1992)]. The effects may be due to long-term exposure to the frequency gain characteristic of the hearing aid, and can be explained by a perceptual relearning process that maximizes sensitivity across the dynamic range.
A sensor for measuring low-frequency surface vibration of a fluid-loaded compliant structure
Authors: McClearly Alan D. , Young A. Mark , Klippel Patrick J. , Trivett D. H.
Published: October 2005

Presently, the most common method of measuring the surface vibration of a fluid-loaded structure is through the use of accelerometers mounted on the surface. When the surface of the structure consists of a low-density, compliant material it is necessary to use small, low-mass accelerometers to avoid mass loading of the surface. There are a variety of commercially available, low-mass accelerometers all of which have low sensitivities. For high-frequency measurements the low sensitivities are not a major problem and the accelerometers are generally acceptable. However, at low frequency, where the acceleration is small, the signal-to-noise ratio obtained with the low-sensitivity accelerometers is generally unacceptable. To address the problem of making low-frequency measurements of the vibration of a submerged compliant surface the Underwater Sound Reference Detachment of the Naval Research Laboratory has developed a class of neutrally buoyant capacitive displacement sensors. The dynamic mass of these sensors is equal to the mass of the displaced fluid, thus the sensor does not add any additional loading to the surface. Since the sensors respond to the displacement of the surface these sensors are ideally suited for low-frequency measurements. Both the theory of operation as well as experimental results will be presented. [Work supported by ONR Code 452.]
Waves on fluid-loaded plates with equally spaced identical ribs
Authors: Wang Zhang , Norris Andrew
Published: October 2005

A low-frequency asymptotic dispersion relation for a fluid-loaded plate with infinite number of equally spaced identical ribs is derived, from which an equation of motion for the plate is inferred that is valid also at low frequencies. Both the asymptotic and the exact dispersion curves are computed and compared. A relatively simple formulation is obtained to compute the positions and widths of the stopping bands. The dispersion curve and the stopping bands for a plate in vacuum with infinite number of equally spaced identical ribs are also presented and compared with those for the fluid-loaded plate. A scattering matrix method is employed to solve wave propagation problems on infinite fluid-loaded plates with finite numbers of equally spaced identical ribs. The response to plane wave incidence of fluid-loaded plates with infinite and finite numbers of equally spaced identical ribs are compared. [Work supported by ONR.]
Selection of a shell model for a cylindrical shell immersed in fluid
Authors: Manley Martin G.
Published: October 2005

There are many thin shell models available to the investigator studying the vibrations of thin-walled, circular, cylindrical shells. A rational method for selecting a model for any given problem is needed. An asymptotically exact dispersion relation for waves on fluid-loaded, elastic, hollow cylinders derived from the exact equations of elasticity previously [M. Manley, J. Acoust. Soc. Am. 93, 2335(A) (1993)] is applied to determine dominant terms for dispersion of axial waves. This is compared with dispersion relations obtained from shell models. The simplest shell model retaining the dominant terms in the dispersion relation is recommended. Some examples are presented. [The author acknowledges the advice of A. D. Pierce.]
Turbulent boundary layer wall pressure spatial filtering with sensors constructed of polyvinylidene fluoride (PVDF)
Authors: Gershfeld Jonathan L. , Blake William K. , Gilheany John J. , Casarella Mario
Published: October 2005

Piezoelectric pressure transducers constructed of electrically polarized polyvinylidene fluoride (PVDF) were assessed as spatial filters for a spatially homogeneous turbulent boundary layer induced wall pressure field. Rectangular-shaped transducers were constructed of lengths ranging from 0.25 to 7.3 times the displacement thickness of the boundary layer in the streamwise direction and widths of 0.25 the displacement thickness. Measurements of the power spectral levels of the wall pressure fluctuations are compared for each transducer with the predicted levels determined by the integration over the spatial extent of each transducer of the Corcos model of the wall pressure cross spectrum. [Work supported by ONR.]
Mean flow effects on the vibroacoustic behavior of a plate backed by a cavity in light and heavy fluids
Authors: Sgard Franck , Atalla Noureddine , Nicolas Jean
Published: October 2005

The purpose of this work is to present a general formulation dealing with the vibroacoustic behavior of a baffled plate, immersed on one side in a mean flow and coupled on the other side to a rectangular cavity with arbitrary wall impedances. The plate is excited by a point force, a plane wave convected by the mean flow, or a turbulent boundary layer. The formulation is based on a finite element method for the calculation of the transverse vibrations of the plate. Fluid loading on each side of the plate is taken into account through an impedance matrix approach. The plate–cavity impedance matrix is calculated using a boundary element method. The amount of computation is decreased by using a two indices Green’s function for the rigid cavity. The external impedance matrix accounting for mean flow, is calculated using the extended form of Kirchhoff’s integral equation and a boundary element method. This formulation is compared with a finite element approach for the calculation of the acoustic field inside the cavity. The mean flow effects are investigated through the calculation of different vibroacoustic indicators like the mean-square velocity of the plate, the internal and external acoustic radiated power, the radiation efficiency of the plate inside the cavity and the transmission loss of the plate. Coupling mechanisms are explained in terms of added mass, stiffness, and damping.
Latent spatial resonances at thin-walled structure vibration and acoustic radiation in convecting field of random pressure fluctuations
Authors: Efimtsov B. M.
Published: October 2005

Fundamental phenomena defining thin-walled structure vibration and acoustic radiation in uniform and essentially nonuniform convecting fields of random pressure fluctuations are investigated. Investigated, in particular, are the new effects of vibration increase, which are revealed at some definite relations between spatial scales of correlation, nonuniformity, phase velocity of the pressure fluctuation field, elastic wavelengths in the structure, and their propagation velocity. The new effects of the vibration increase can be considered as latent spatial resonances. Just those resonances determine the main regularities in vibration of arbitrary thin-walled structure in the convecting field of random pressure fluctuation. Moreover, those resonances determine acoustic radiation at some moderate (usually realized) energy dissipation in the structure. In this case the radiation is associated with the resonance excitation of the structure. The well-known aerodynamic coincidence effect appears only in the case when spatial scales of correlation and nonuniformity exceed the elastic wavelength in the structure. At the high degree of vibroabsorption the acoustic radiation associated with the nonresonance excitation of the structure (their inertial behavior) can become dominant. In this case similar effects of the acoustic radiation increase become dominant. They are revealed at some definite relations between spatial scales of correlation and nonuniformity, phase velocity of pressure fluctuation, sound wavelength, and its propagation velocity in a medium, where the sound field is formed.
The isolation of scattering mechanisms
Authors: Dragonette Louis R.
Published: October 2005

One of the outstanding benefits that resulted from Flax’s work on the scattering from simple shapes was the impetus given to isolating and understanding basic scattering mechanisms. For example, his pioneering work on the development of acoustic resonance theory described methods for the separation of elastic and geometric contributions to the scattered field, and for differentiating between elastic methods. As the targets considered become more complicated, isolation of generally overlapping mechanisms becomes increasingly more difficult. Standard time/frequency approaches and newer techniques such as wave packet analysis can be applied to these problems, with confidence obtained from their applications to the well understood canonical shapes. Examples of the isolation of scattering mechanisms derived from targets of increasing complexity are presented.
The acoustical background for submerged elastic targets and resonance scattering phenomena
Authors: Werby M. F.
Published: October 2005

The pioneering paper on resonant acoustical scattering from elastic solids by Flax etal. [J. Acoust. Soc. Am. 63, 723–731 (1978)] (FDU) formed the basis for resonance scattering theory; proposing the decomposition of the elastic response into resonance and nonresonance ‘‘background’’ components. The resonance component was equivalent to the Breit–Wigner form from quantum mechanics, in which Flax and Uberall were trained. The ‘‘background’’ for elastic solids was that of a rigid scatterer. In a series of papers, FDU introduced many resonance classification schemes based on actual physical mechanisms. The FDU development bore much fruit in target classification physics in over a decade of lively research. Flax, also interested in elongated targets and elastic shells, initiated theoretical studies involving Waterman’s T-matrix method. Here, some recent work stemming from Flax’s interests stimulated by our early interaction with him is reported. The best backgrounds for generally shaped evacuated and fluid-filled elastic shells are discussed; and their use to interpret complicated resonance phenomena are outlined. Also demonstrated is that the FDU theory can be derived formally from an abstract mathematical structure. [Work sponsored by the Naval Research Laboratory 6.1 Program Element 0601153N.]
Structural resonance contributions to scattering cross sections of cylindrical shells: Mode and frequency dependence
Authors: Junger Miguel C.
Published: October 2005

H. Lamb’s compact formulation of resonance contributions to the scattered field of elastic, lossless spheres (1900) was generalized by P. W. Smith, Jr. to structurally damped scatterers of arbitrary shape (1962). Excluding axisymmetric modes, modal configurations of cylindrical shells are identified by a pair of axial/circumferential wave numbers associated with a predominantly flexural, poorly radiating resonance and two supersonic (membrane and shear) resonances. Implementing Smith’s solution for the latter two yields a simple expression for the familiar ‘‘chalice’’ resonance peak loci in the aspect/frequency plane. A peak’s height is proportional to the square of the ratio of the modal radiation resistance over the combined radiation and structural resistance. With increasing wave numbers, modal configurations gradually become tangential, thereby reducing radiation damping as compared to total damping (an effect overlooked in this speaker’s 1993 publication, which unrealistically neglected structural as compared to radiation resistance). Consequently, though associated with supersonic wave numbers, resonance peaks fade with increasing mode order and frequency. Results are compared with Rummerman’s (1993) detailed analysis and measurements. [Work sponsored by Naval Undersea Warfare Center.]
Linear time-invariant approaches to feedforward, multifrequency control
Authors: Vipperman Jeffrey S. , Clark Robert L.
Published: October 2005

There are many applications where the unwanted noise or vibration response of a system is composed of multiple harmonics. Unless care is taken in selecting the size of the adaptive control filter and the sampling rate, the LMS algorithm will converge to a time-varying solution, which oscillates about the optimal control solution for the system. Further, the response due to each harmonic can vary in magnitude by more than an order of magnitude, causing a much higher signal-to-noise ratio for the weaker frequencies. In such a case, it is best to implement a separate independent reference signal and corresponding independent adaptive filter for each frequency contained in the system. The resulting filters have a stationary optimal solution and the dynamic range of the control system is improved. Experiments were performed on a simply supported steel beam to demonstrate the effects of sampling rate, filter size, and signal-to-noise ratio for multifrequency and higher harmonic control applications. [Work supported by the Structural Acoustics Branch of NASA-Langley.]
Practical considerations for designing shaped modal sensors from polyvinylidene fluoride
Authors: Clark Robert L. , Burke Shawn E.
Published: October 2005

The purpose of this work is to explore practical methods of achieving distributed sensors on two-dimensional structures and illuminate difficulties associated with positioning sensors on structures in general. A method of designing ‘‘modal family’’ sensors is introduced whereby shaped sensors, similar to those employed on one-dimensional structures, are utilized to monitor the response of a group of modes on two-dimensional structures. Limits on sensor width with respect to the trace structural wavelength along the width of the sensor are provided. In addition, positioning of distributed strain sensing material is shown to be critical to the performance of the sensor over a specified bandwidth, specifically in the vicinity of structural resonances.
A directional adaptive LMS acoustic array for hearing enhancement
Authors: McKinney Elaine D. , DeBrunner Victor E.
Published: October 2005

Acoustic fixed arrays are designed to match a focused array gain pattern, while acoustic adaptive arrays are designed to respond to changing environments. However, as currently constructed, adaptive arrays cannot stay focused with the limited number of microphones available in a cosmetically acceptable hearing aid. In this paper, a technique that combines fixed and adaptive arrays in a system that enhances the desired signal while effectively attenuating interference speech and background noises is discussed. In particular, the design and performance of a directional adaptive four-supercardioid microphone array is examined. This idea was formed independently from the use by Soede etal. [J. Acoust. Soc. Am. 94, 785–798 (1993)] of fixed arrays with cardioid microphones. Thus it is confirmed that using directional microphones improves array directivity in the hearing aid application. Simulations show that the directivity of this adaptive array is improved to 7.15–7.71 dB. The array reduces interferences from different directions by about 15–30 dB and provides a SNR improvement of about 9–12 dB over a single omni-directional microphone. A sensitivity analysis is also discussed. [Work partially supported by Deafness Research Foundation.]
Preventing reverberation-induced target cancellation in adaptive-array hearing aids
Authors: Greenberg Julie E. , Zurek Patrick M.
Published: October 2005

Previous work has shown that adaptive-array hearing aids based on the Griffiths–Jim beamformer perform poorly in reverberant environments. Although relatively long adaptive filters successfully reduce reverberant environments. Although relatively long adaptive filters successfully reduce reverberant interference, they also cause substantial cancellation of the desired (target) signal [Greenberg and Zurek, J. Acoust. Soc. Am. 91, 1662–1676 (1992)]. In this work, analysis of the adaptive noise canceller shows that target cancellation is eliminated when the delay in the primary channel (D) is shorter than the interval between the arrival of the direct wave and first reflection at the microphones. This generalizes the observation that setting D=0 prevents target cancellation due to reverberation [Hoffman, Ph.D. thesis, Univ. of Minnesota (1992)]. Computer simulations with several sets of source-to-microphone impulse responses show that target cancellation is most pronounced when there are a small number of reflections and they arrive at the array within D samples of the direct wave. Target cancellation is minimal when there are many reflections and only a small fraction arrive within D samples of the direct wave. Other benefits derived from using nonzero primary delays suggest that it is advisable to use a primary delay that is nonzero, but relatively small. [Work supported by NIDCD.]
Method for fitting binaural hearing aids
Authors: Gao Shawn , Sullivan Jean , Jayaraman Sriram , Soli Sigfrid D.
Published: October 2005

For a hearing aid wearer to perform binaural sound localization and to utilize directional hearing in noisy environments, it may be important to maintain at audible levels the binaural cues (i.e., interaural time and level differences) present without the hearing aid(s) in place. A method of achieving this hearing aid fitting goal for use with a prototype digital signal processing hearing aid has been developed. The method includes two major steps: hearing aid equalization (HAE) and hearing loss compensation (HLC). HAE is achieved with an FIR filter, which equalizes the amplitude and phase insertion effects of the hearing aids and maintains the binaural cues with the hearing aid(s) in place. The HAE filter coefficients are obtained from insitu probe tube measures of aided and unaided test signals using optimal filter calculations. HLC is also achieved with an FIR filter and associated gain. The HLC filter coefficients are obtained by using a weighted least-squares filter design technique. The target response for the HLC filter is determined from measures of electrical signal levels in the hearing aid circuit during threshold tests and during reference signal presentations in the sound field. The HAE and HLC filters are convolved to produce a single filter. A description of the fitting procedure will be provided, as well as examples of its use with hearing impaired individuals.
Amplitude envelope cues for consonant recognition with simulated loudness recruitment
Authors: Freyman Richard L. , Nerbonne G. Patrick , Tharp Diane , Stanford Emily
Published: October 2005

The perceptual consequences of expanding the amplitude variations in speech to simulate loudness recruitment were studied under conditions in which spectral information was obscured to varying degrees, but audibility was largely preserved. Signal correlated noise samples, created individually to follow the amplitude envelopes of 22 VCV nonsense words, were used as maskers of those words, with S-N ratios of −15 to 0 dB. Amplitude expansion was by a factor of 3.0 in terms of decibels; e.g., a difference in levels of 10 dB between adjacent consonants and vowels was expanded to approximately 30 dB. Presentation level of speech peaks was 80 dB SPL. Many consonant levels were substantially less intense, but almost all were still audible. Recognition performance for the expanded speech by 50 normal-hearing listeners was as much as 30 percentage points poorer than unexpanded speech, and the types of errors were dramatically different, particularly in the midrange of S-N ratios. Follow-up testing as a function of level confirmed that differences between conditions were due largely to expanded amplitude envelopes rather than differences in audibility. The data suggest that amplitude envelope cues are vulnerable to the effects of expansion of the type seen in loudness recruitment. [Work supported by NIH ♯DC01533.]
Spectral enhancement using a two-filter model of impaired frequency selectivity
Authors: Stone Michael A. , Moore Brian C. J.
Published: October 2005

Neural tuning curves can be described as comprising two shapes: a narrow ‘‘tip’’ filter with high sensitivity, and a wider ‘‘tail’’ filter with lower sensitivity. When cochlear function is impaired, the sensitivity of the tip filter decreases. The tail filter characteristics are difficult to measure in normal-hearing subjects using psychophysical techniques such as the notched-noise method. However, with mild to severe cochlear hearing loss, the tail filter has a greater influence on the results, and becomes easier to measure. Auditory filter shapes were estimated using the notched-noise method over a wide range of center frequencies for subjects with differing degrees of hearing loss. These data were used to construct a model that predicts typical filter shapes as a function of frequency and hearing loss. The model is being used to improve the accuracy of a spectral contrast enhancement method intended to compensate for the effects of reduced frequency selectivity. The method is similar to one also being developed in this laboratory [T. Baer and B. C. J. Moore, this meeting], but uses a filter bank rather than overlap-add FFTs. Results of intelligibility tests using speech in speech-shaped noise, enhanced by this method and tested on hearing-impaired subjects, will be described. [Work supported by the MRC.]
Spectral enhancement to compensate for reduced frequency selectivity
Authors: Baer Thomas , Moore Brian C. J.
Published: October 2005

A new method of spectral enhancement has been developed to compensate for reduced frequency selectivity in people with sensorineural hearing impairment. The goal is to modify a sound such that it evokes as nearly as possible the same excitation patterns in an ear with reduced frequency selectivity as would be evoked by the unmodified sound in a normal ear. Processing is performed in the spectral domain, using an overlap-add procedure. Enhanced spectra are calculated using a nonnegative-least-squares algorithm for matching excitation patterns. When compensating for a moderate degree of auditory-fiber broadening, this procedure dramatically increases spectral contrast while preserving relatively normal sound quality. To examine the effects of enhancement on the intelligibility of speech in speech-shaped noise (at −3 dB SNR), a pilot experiment was performed with normal-hearing subjects, using a spectral smearing method [T. Baer and B. C. J. Moore, J. Acoust. Soc. Am. 94, 1229–1241 (1993)] to simulate some aspects of reduced frequency selectivity. Results show that the enhancement compensates partially for the effects of spectral smearing and should thus improve the intelligibilty of speech in noise for the hearing-impaired. Experiments are presently underway to test the effects of enhancement on speech perception with hearing-impaired subjects. [Work supported by the MRC.]