bluuebunny/crossref_metadata_2025 huggingface dataset:
Active control of broadband structural vibration using the adaptive LMS algorithm.
Authors: Vipperman Jeffrey S. , Burdisso Ricardo A. , Fuller Chris R.
Published: October 2005

Adaptive feedforward control algorithms have been successfully implemented to attenuate structural vibration and its sound radiation when subjected to single and multiple sinusoidal excitations. Reduction of broadband noise in ducts and enclosures using both finite and infinite impulse response adaptive filters has been reported in the literature. However, feedforward control of structural response under broadband excitation has not been experimentally demonstrated. Here, an adaptive feedforward controller based on the filtered-x LMS algorithm has been developed for the active control of broadband vibration in structures. The control signal is obtained by filtering the reference signal through an adaptive finite impulse filter (FIR). The filtered-x LMS algorithm requires an estimate of the transfer function between the control input and the error output. This transfer function is represented by an infinite impulse response (IIR) filter. Two control configurations are then experimentally investigated on a simply supported beam. A power reduction of up to 15 dB is demonstrated. [Work supported by ONR/DARPA.]
An energy-based control strategy for minimizing the sound field in enclosures.
Authors: Nashif Peter J. , Sommerfeldt Scott D.
Published: October 2005

Squared pressure has generally been the quantity chosen to minimize when controlling the sound pressure level (SPL) in enclosures. However, difficulty in selecting the best possible microphone location makes this method unattractive for arbitrary enclosures. As an alternative control strategy, minimizing the total energy density in an enclosure has been investigated. This method is a logical alternative because the total energy in the field is composed of both pressure and velocity components, as opposed to simply a pressure component. Computer simulations and experimental measurements of reducing the SPL in a rigid-walled duct have been conducted to compare the two strategies. The results of the experiments show that minimizing the total energy provides greater attenuation of the total potential energy with the added advantage of being less sensitive to microphone placement than when minimizing the squared pressure.
Enhanced machine recognition of speech through the integration of handwriting side information.
Authors: Bellegarda Jerome R. , Kanevsky Dimitri
Published: October 2005

At last year’s ASA Fall Meeting the authors proposed a new approach for the correction of errors introduced by an automatic speech recognizer (ASR) during the decoding of difficult, large vocabulary speech tasks. This approach involves overwriting the incorrectly decoded words (or character strings) using a pressure-sensitive pen, on a transparent paperlike interface (PLI) capable of recognizing handwriting on-line. This solution combines the best features of both keyboard entry and mouse selecting: It is appropriate even for out-of-vocabulary words, yet the user’s attention remains concentrated on the screen at all times. In fact, further analysis shows that four intrinsic forms of complementarity between speech and handwriting add to the viability of this solution. Here evidence of complementarity in information, performance, functionality, and robustness is presented. Such complementarity contributes to the relative speed and ease of use of PLI to correct ASR errors. This in turn suggests an extension of this idea toward a jointuse of both ASR and PLI inputs. In this situation, ASR and PLI are used simultaneously, and the results of handwriting recognition are fed to the speech recognition module as apriori information to help reject unlikely possibilities and thereby guide the selection of promising candidates. Experimental results indicate that this approach may lead to substantially higher message recognition accuracy. As it requires users to speak and write consistent messages, psychological experiments that demonstrate this willingness are reported provided the increase in recognition accuracy effectively eliminates the need for error correction.
Spectral characterization of microevents in speech.
Authors: Berry David A. , Strong William J.
Published: October 2005

In a previous meeting, the authors presented an algorithm for extracting microevents (e.g., pitch periods, explosive bursts, packets of energy during frication) from continuous speech [D. A. Berry and W. J. Strong, J. Acoust. Soc. Am. 89, 1893 (A) (1991)]. After initially characterizing microevents by their one-third octave spectra, two methods were studied for plotting microevents in reduced, two-dimensional spectral spaces. The first method involved principal components in which the first two components captured approximately 60% of the spectral variance. However, microevents clustered into relatively small regions of the space and the resulting plots were often counterintuitive and difficult to interpret. The second method was a ‘‘trispectral’’ approach adapted from previous tristimulus studies (H. F. Pollard and E. V. Jansson, Acustica 51, 162–171). Low-, mid-, and high-frequency one-third octave filters were grouped so as to obtain a fairly uniform distribution of microevents throughout the trispectral space. The resulting plots were intuitively easy to interpret, with the distribution of vowel microevents being analogous to typical F1-F2 plots. Moreover, because the trispectral approach was based on filter outputs rather than formants, it permitted voiced and unvoiced events to be plotted on the same graph. Microevent distributions and trajectories from continuous speech will be presented in both spectral spaces.
Graphical editor for marking spectrograms (GEMS).
Authors: Talley Jim
Published: October 2005

This paper presents a new computer system, GEMS (or the Graphical Editor for Marking Spectrograms), for interactively collecting frequency and timing data from speech samples. GEMS was originally developed as a data collection aid for on-going research in vowel formant transition patterns, but it potentially has wider applicability. It takes digitized speech as input, generates a spectrographic display as the background, and yields as output time, frequency, and continuity data on the points (and sequences) marked as interesting by the user. This system, as its name suggests, is primarily a graphical editor for such markings. However, given the editing substrate, it is possible to incorporate less than perfect algorithms (true of most) to automate selection/marking of interesting features. The user immediately sees the results of such automatic techniques (e.g., LPC analysis) and has an opportunity to modify the outcome in context before proceeding. A middle road is, thereby, taken between completely automatic, but error prone, data production, and time consuming manual techniques of data collection. This framework can also serve as an environment for improvement of algorithms with a learning component. The system is currently implemented in pascal under VAX/VMS. It is designed with a client/(DSP) server architecture, and implements some automated techniques such as a new heuristic voice pulse localization algorithm. [Work supported by NSF.]
An efficient algorithm for articulatory synthesis.
Authors: Ananthapadmanabha T. V. , Jayasimha H. N.
Published: October 2005

One of the methods for efficient computation of articulatory synthesis is to discretize the area function into a small number of sections such as 10 or 12 resulting in a poor spatial resolution. The articulatory data have to be represented by 30 or more sections for a good spatial resolution requiring a large number of computations. However, good quality speech synthesis has been obtained using formant or LPC-based methods with filters of 10 or 12 coefficients for voiced sounds and 4 or 6 coefficients for unvoiced sounds. It is proposed that a computationally efficient method be used to obtain high quality synthesis using an articulatory model with a large number of sections (30 or more). It can be shown that when the adjacent sections are of equal area, the computations can be pruned without any loss of accuracy. For a given computational load, using an effective quantization rather than discretization of area function, one can achieve much lower spectral error in the transfer function. Trade-off relationships between spectral error and computational load as a function of quantization and discretization are reported.
Comparing human and neural network lip readers
Authors: Yuhas Ben P. , Goldstein Moise H.
Published: October 2005

Performance in identifying nine vowels from static images of a speaker’s mouth was studied experimentally for neural networks and human subjects. Data obtained indicated that they had essentially the same rate of correct identifications overall and vowel by vowel. The estimate of the probability of correct identification and 95% confidence limits were determined for each of the nine vowels.
Characterization of multiple piezoelectric actuators for structural excitation
Authors: Clark Robert L. , Fuller Chris R. , Wicks Al
Published: October 2005

The thrust of the present work is to analytically and experimentally study the response of a simply supported beam driven by multiple piezoelectric actuators in an effort to understand distributed excitation of the structure. The results indicate that the theoretical model provides the basis for a viable means of determining appropriate locations for piezoelectric actuators for exciting desired modal distributions in the structural response.
Plate wave acoustic emission
Authors: Gorman Michael R.
Published: October 2005

Plate theory is more easily applied to the analysis of composite laminates than exact three-dimensional elasticity theory. Under conditions such that plate theory is applicable, it is suggested that plate waves are useful for understanding acoustic emission (AE) phenomena. To test this idea, pencil leads were broken on aluminum plates and composite plates, and the resulting waves were detected with a broadband ultrasonic transducer. Both the fundamental extensional and flexural modes were observed. Their characteristics are described and the implications for AE source location are discussed as well. Several transducers, commonly used for acoustic emission measurements, are compared with regard to their ability to reproduce the characteristic shapes of plate waves. Their different responses show why similar test specimens and test conditions can yield disparate results.
The scattering of acoustic waves by small, surficial inhomogeneities
Authors: Lerche I.
Published: October 2005

Based on mean field methods a technique is presented for examining the coherent and incoherent reflection of acoustic waves from a spatially randomly variable surface with constant impedance mismatch along the surface. The power in the scattered component is shown to be proportional to the fourth power of frequency at low frequencies (the Rayleigh limit) and is roughly a constant fraction of the input power at high frequencies. The spatial wave-number variation of the scattered field depends sensitivity on the wave-number spectrum of the spatial fluctuations of the random surface, as well as on the mean position of the surface. The effect on the coherent component of the reflected acoustic field is as though the interface moved its mean position as a function of frequency, with no displacement at low frequencies tending toward a constant displacement at high frequencies. The technique is applicable to more complex wave/media interactions.
A boundary element formulation for acoustic shape sensitivity analysis
Authors: Kane J. H. , Mao S. , Everstine G. C.
Published: October 2005

The Helmholtz integral equation forms a conventional basis for acoustic boundary element analysis (BEA). Implicit differentiation of the discretized Helmholtz integral equation is shown to yield an effective approach for the computation of rates of change (sensitivities) of acoustic response quantities with respect to changes in the shape of an acoustic model. A theoretical formulation is presented that allows for the reuse of the factorization of the overall BEA system left-hand side matrix formed in a previous analysis, thus obviating the need to factor perturbed matrices in the sensitivity analysis process. The singularity strengths of the new kernel functions utilized to compute sensitivities of matrix coefficients required by this approach are shown to be the same as those present in ordinary BEA. An indirect approach for the computation of the diagonal contributions to sensitivity matrices associated with these new kernels is also discussed. The sensitivity analysis formulation includes surface pressures and normal gradients (velocities), surface tangential pressure gradients, and pressure and pressure gradients at arbitrary domain sample points. It is concluded that the overall efficiency of implementations of these formulations is significant. Numerical results for a series of example problems are presented to quantify the accuracy and efficiency of this approach.
Effects of peak pressure and energy of impulses
Authors: Patterson James H.
Published: October 2005

Peak pressure has been one of the key parameters of impulse noise used to assess the hazard to hearing. It is used in most international noise exposure limits. France uses an A-weighted energy limit. There is a rough correspondence between peak pressure and the hazard to hearing for a given type of impulse noise. However, when the effects of different types of impulses are compared, this correspondence breaks down. One of the alternate measures of impulse intensity is weighted energy. Weighted energy is appealing for a number of reasons. It does not depend on details of the pressure-time history such as the peak pressure and the more common duration measures. It should be easier to integrate with continuous or intermittent noise standards. It would make it easier to use standard hearing protector attenuation to estimate the hazard when a specific hearing protector is worn. Results of previously published articles and reports will be discussed. These reports lead to the conclusion that weighted energy is a more potent determiner of hearing hazard than peak pressure if spectral effects are controlled.
Evaluation of orthogonal polynomial compression
Authors: Levitt Harry , Neuman Arlene C.
Published: October 2005

In orthogonal polynomial compression, the short-term speech spectrum is first approximated by a family of orthogonal polynomials. The coefficients of each polynomial, which vary over time, are then adjusted in terms of their average value and range of variation. These adjustments can be used to compress (or expand) temporal variations in the average level, slope, and various forms of curvature of the short-term speech spectrum. The analysis and reconstruction of the short-term speech spectrum using orthogonal polynomials was implemented using a digital master hearing aid. This method of compression was evaluated on eight sensorineurally hearing-impaired listeners. Speech recognition scores were obtained for a range of compression conditions and input levels with and without frequency shaping. The results showed significant advantages over conventional linear amplification when temporal variations in the average level of the short-term spectrum were compressed, a result comparable to that obtained with conventional amplitude compression. A subset of the subjects showed further improvement when temporal variations in spectrum slope were compressed, but these subjects also showed similar improvements when frequency shaping was combined with level-only compression. None of the subjects showed improved speech recognition scores when variations in quadratic curvature were compressed in addition to level and slope compression.
Some further investigations of the seat dip effect
Authors: Bradley J. S.
Published: October 2005

The attenuation of low-frequency sound, passing at near grazing incidence over audience seating, was investigated by examining spectra of the direct and early arriving sound. Detailed measurements in one hall and on scale model seats showed the frequency of the principal attenuation minimum to be related to the vertical angle of incidence as well as to the presence of additional early arriving sound energy. Octave band measurements in 11 halls demonstrated that the low-frequency attenuation was strongly influenced by the geometry of the halls and in particular by the average ceiling height. Measurements of a string bass showed large level changes due to the seat dip attenuation, and this phenomena is concluded to play an important role in determining the perceived quality of a concert hall. These new results can be used to help design halls to minimize the negative effects of the seat dip attenuation.
Human temporary threshold shift (TTS) and damage risk
Authors: Melnick William
Published: October 2005

Information regarding the relation of human temporary threshold shift (TTS) to properties of steady-state and intermittent noise published since the 1966 appearance of the CHABA damage risk contours is reviewed. The review focuses on results from four investigative areas relevant to potential revision of the CHABA contours including effects of long-duration exposure and asymptotic threshold shifts (ATS); equivalent quiet and /or safe noise levels; effects of intermittency; and use of noise-induced temporary threshold shift (NITTS) to predict susceptibility to noise-induced permanent threshold shift (NIPTS). These data indicate that two of three major postulates on which the original contours were based are not valid. First, recovery from TTS is not independent of the conditions that produced the TTS as was assumed. Second, the assumption that all exposures that produce equal TTS2 are equally hazardous is not substantiated. The third postulate was that NIPTS produced by 10 years of daily exposure is approximately equal to the TTS2 produced by the same noise after an 8-h exposure. Based upon several TTS experiments showing that TTS reaches an asymptote after about 8 h of exposure, the third CHABA postulate can be reworded to state the hypothesis that ATS produced by sound of fixed level and spectrum represents an upper bound on PTS produced by that sound regardless of the exposure duration or the number of times exposed. This hypothesis has a strong, logical foundation if ATS represents a true asymptote for TTS, not a temporary plateau, and if threshold shifts do not increase after the noise exposure ceases.
Recent studies of temporary threshold shift (TTS) and permanent threshold shift (PTS) in animals
Authors: Clark William W.
Published: October 2005

It is well known that excessive exposure to noise results in temporary and/or permanent changes in hearing sensitivity in both human and animal subjects. The purpose of this review is to describe the major findings from laboratory studies of experimentally induced hearing losses, both temporary and permanent, resulting from exposure to noise in animal subjects which have been published since the report of Kryter et al. (1966). The data reviewed support the following general statements: (1) The chinchilla is the most widely used and most appropriate animal model for studies of noise-induced hearing loss; (2) with continuous exposures to moderate-level noise, thresholds reach asymptotic levels (ATS) within 18–24 h; (3) permanent threshold shifts, however, depend upon the level, frequency, and the duration of exposure; (4) below a ‘‘critical level’’ of about 115 dB, permanent threshold shift (PTS) and cell loss are generally related to the total energy in continuous exposures; (5) periodic rest periods inserted in an exposure schedule are protective and result in less hearing loss and cochlear damage than equal energy continuous exposures; and (6) under some schedules of periodic exposure, threshold shifts increase over the first few days of exposure, then recover as much as 30 dB as the exposure continues.
F
stylization in speech: Straight lines versus parabolas
Authors: ’t Hart Johan
Published: October 2005

This is a brief report on an experiment, intended to show that a piecewise linear approximation of an F0 curve in speech is, perceptually, not inferior to an approximation by means of fragments of parabolas, which gives—visually at least—a better fit to the original F0 curve than does the rectilinear approximation. (More details can be read in IPO Rep. no. 816, available on request.) Stimuli consisted of two pairs of linearly or parabolically frequency modulated pulse trains, one of which contained identical, the other different members. The subjects had to indicate whether it was the first, or the second pair that contained different members. The results showed that even the best performers were hardly ever able to distinguish the parabolic from the rectilinear shapes, provided the latter contained a flattened peak.
On new features in the computation of reflection and transmission factors for an arbitrary number of liquid and elastic layers, in arbitrary order
Authors: Cervenka Pierre
Published: October 2005

This Letter is in response to the Letter to the Editor by J. George entitled ‘‘Comments on ‘A new efficient algorithm to compute the exact reflection and transmission factors for plane waves in layered absorbing media (liquid and solids)’ ’’ [J. Acoust. Soc. Am., this issue]. It is pointed out that this comment is not relevant.
Responses of ‘‘lower-spontaneous-rate’’ auditory-nerve fibers to speech syllables presented in noise. II: Glottal-pulse periodicities
Authors: Geisler C. Daniel , Silkes Steven M.
Published: October 2005

The responses of single cat auditory-nerve fibers to natually spoken voiced sounds (the vowels [a, i, u] and the murmur [m]) presented at normal intensity (70 dB SPL) in various levels of speech-shaped noise were analyzed for the encoding of the glottal-pulse (fundamental) period. To quantify the strength of this fundamental-period encoding, selected segments of the response histograms were autocorrelated, rectified, and fitted with the best-fitting sinusoid of the fundamental frequency. The magnitude of this best-fitting sinusoid was taken as the magnitude of synchronization. In most cases, it was found that the ‘‘lower-SR’’ fibers (those with spontaneous discharge rates <20/s) encoded the fundamental periodicity more strongly and more robustly than did the ‘‘high-SR’’ fibers (those with spontaneous discharge rates >20/s). When either a single strong spectral peak or a relatively ‘‘flat’’ spectrum excited a fiber, it showed poor synchronization to the fundamental period, regardless of its spontaneous-rate class. Judging from a few examples, the glottal-pulse synchronization appears to be intensity dependent, with the relative performance of the high-SR fibers improving at lower intensities. A conceptual model is given which accounts for the general characteristics of the data.
A comparison of spatial interpolation and transformation matrices in the coherent signal subspace for bearing estimation of multiple broadband signals
Authors: Chen Yuan-Hwang , Yu Feng-Peng
Published: October 2005

The coherent signal-subspace (CSS) algorithm proposed by Wang and Kaveh [IEEE Trans. Acoust. Speech Signal Process. ASSP-33, 823–831 (1985)] for high-resolution directions-of-arrival (DOAs) of multiple coherent broadband signals focuses the signal subspaces at various frequencies into a common one through transformation matrices preprocessing. The preliminary angle estimates of DOAs using a spatial periodogram are often required to obtain the transformation matrices, and these angle estimates, if they are inappreciately selected especially in multigroup case, sometimes result in an estimation error in the transformation matrices and the performance of DOAs is thus degraded. Therefore, an interpolation preprocessing concept [Bienvenu et al., Proc. ICASSP, pp. 2799–2802 (1989)] was introduced to replace the transformation matrices without the need of initial angle estimates. The objective of this paper is to further compare the statistical characteristics and DOAs estimation performances of the two preprocessings coupled with CSS in the multigroup case. Computer simulations show that each of the two preprocessings has its advantages and disadvantages.
Effects of a single reflection with varied horizontal angle and time delay on speech intelligibility
Authors: Nakajima Tatsumi , Ando Yoichi
Published: October 2005

Previously, almost all physical measures for estimating speech intelligibility in a room have been derived from only temporal-monaural criteria. This paper shows that speech intelligibility for a sound field with a single reflection depends not only on the temporal-monaural factor but also on the spatial-binaural factor of the sound field. Articulation tests for sound fields simulated with a single reflection of delay time Δt1 after the direct sound were conducted changing the horizontal incident angle ξ of the reflection. Remarkable findings are as followings: (1) speech intelligibility (SI) decreases with increasing delay time Δt1, (2) SI increases when ξ approachs 90°; the horizontal angle of the reflection causes a significant effect on SI, and (3) the analysis of variance for articulation test scores clearly demonstrated that the effects of both Δt1 and ξ on SI are fully independent. Concerning result (2), if listeners get a spatial separation of signals at the two ears, then the listener’s capability for speech perception is assumed to be improved due to ‘‘adding’’ further information to the temporal pattern recognition.
Linear vibration analysis of an ultrasonic cleaning problem
Authors: Geers Thomas L. , Hasheminejad Mohamad
Published: October 2005

A micron-size spherical particle is bound to an elastic half-space by van der Waals forces. The particle and the surface of the half-space are immersed in an acoustic fluid and excited by harmonic plane waves propagating normally to the surface from either direction. The low-amplitude vibration of the particle is unidirectional and linear about an equilibrium position determined by van der Waals attraction and Hertz contact. An equation of motion is formulated for this problem that incorporates the effects of particle and fluid inertia, radiation and material damping, and surface and particle compliance. Numerical results are presented for a silicon particle adhering to a silicon half-space both in air and in water. Highly resonant response is found, the product of long-wavelength scattering in the resonance-frequency region. This result points to ‘‘resonance cleaning’’ as a promising method for efficient particle removal.
A two-dimensional downslope propagation model based on coupled wedge modes
Authors: Primack Harel , Gilbert Kenneth E.
Published: October 2005

A ‘‘wedge-mode’’ representation of the acoustic field has been implemented in a two-dimensional downslope propagation model that can treat realistic sound-speed profiles. Wedge-mode calculations are compared to parabolic equation (PE) calculations for an isospeed profile and profiles typical of the Eastern Mediterranean in winter and summer. For an isospeed profile over a penetrable bottom, coupling between the wedge modes is negligible. For a realistic depth-dependent profile, however, the mode coupling is significant and cannot be neglected. When mode coupling is accounted for, good agreement is obtained with the PE results. The wedge-mode calculation is considerably faster than the reference PE calculation. On a general-purpose workstation, for example, a typical wedge-mode calculation required 10 min while the corresponding finite-difference PE calculation required 8.5 h. As with all mode calculations, the wedge-mode computation is straightforward to implement on a parallel computer. Using off-the-shelf parallel processors (‘‘transputers’’) and a personal computer, the parallel calculation is as fast as the sequential calculation on a general-purpose workstation. Hence it appears, possible to do significant calculations in the field using a personal computer configured with parallel processors.
Acoustic scattering from finite cylindrical elastic objects
Authors: Maze Gérard , Lecroq Florence , Décultot Dominique , Ripoche Jean , Numrich Susan K. , Überall Herbert
Published: October 2005

The acoustic scattering from infinite elastic cylinders or spheres is well known. It is possible to characterize these targets by their resonance spectra. The resonances are established by the generation of surface waves that propagate around the circumference of the targets. The resonances originate from the phase matching of repeatedly circumnavigating surface waves. Experimentally, it is possible to characterize a target with a complicated shape, but it is not easy to explain the spectra theoretically because the geometry is not separable and the usual analytical methods to calculate the far-field pressure cannot be used. In this paper, resonance spectra and angular diagrams obtained from a target consisting of a finite cylinder with hemispherical endcaps are obtained experimentally. To explain the resonance spectra, an integral phase matching condition is used. Upon incidence normal to the cylinder axis, resonances due to the phase matching of surface waves traveling along a circumference or along a meridian have been observed simultaneously.
Environmental correlates of Arctic ice-edge noise
Authors: Makris Nicholas C. , Dyer Ira
Published: October 2005

Temporal variations of low-frequency, broadband ambient noise measured in early summer under drifting ice floes of the marginal ice zone (MIZ) are cross correlated with local environmental forces and ice field descriptors. Surface gravity wave forcing is the primary correlate of the noise; its interaction with ice floes generates sound, most likely via flexural floe failure, and unloading motion, within a few kilometers of the ice edge. Ice concentration is also well correlated with the noise, most likely parametrically. That is, as ice concentration increases, so does the density of potential sound sources in the ice field. Ice stress, ice moment, and wind stress magnitude, while highly correlated with low-frequency noise in the fully ice-covered Arctic, are poor correlates in the MIZ. Lateral melt rate, as a surrogate for thermally induced ice stress, is also poorly correlated with low-frequency MIZ noise. When surface wave forcing is weak, about 1/2 the time in this experiment, episodes of high noise are sometimes observed. These episodes are roughly 3–6 h in duration and 12 h in periodicity. Evidence suggests that they are related to the formation and advection of bands of highly concentrated ice by internal waves during off-ice winds.
Scattering of acoustic waves by a circular disc in the interface between two fluids
Authors: Boström Anders , Peterson Lars
Published: October 2005

The scattering of acoustic waves by a sound-hard circular disc situated in the planar interface between two different fluids is considered. The incident wave is taken as a plane wave and a Hankel transform representation of the scattered field is used. After some manipulations using the boundary conditions, this leads to an integral equation over the disc for the potential jump across the disc. This jump is expanded in a series of Legendre functions which fulfill the correct edge condition and the integral equation is projected on the same set of Legendre functions. The far fields are calculated and the implications of energy conservation are explored and used as a numerical check. A few numerical computations showing the disc-scattered far-field amplitude and the total scattered energy are given and are shown to compare favorably with the expectations from simple ray theory at higher frequencies.
Evaluation of the scanning intensity method for sound power measurements.
Authors: Chu W. T.
Published: October 2005

Despite the recent advances in both instrumentation and application of a sound intensity method for acoustic measurements, there are still standardized methods lacking for sound power measurements using the more convenient scanning method. In this paper, sound power measurement results obtained in both an anechoic chamber and a deadened reverberation room using the intensity scanning method will be presented. Four speaker sources, arranged both as monopole and dipole sources, and a modified ILG source were used. The results were compared with those determined by the reverberation room method according to the ANSI S1.31 standard in the frequency range 100–6.3 kHz. The intensity measurements were taken with the Norwegian electronic type 216 p–v probe. A method of checking the calibration of this probe in an impedance tube will also be discussed.
New approaches to the use of acoustic radiation forces in the calibration of high-frequency transducers.
Authors: Apfel Robert E. , Holland Christy K.
Published: October 2005

A drop (of radius a) of one liquid acoustically levitated in another ‘‘host’’ liquid can be displaced from its equilibrium position by the imposition of the acoustic field of a transducer. The displacement is related to the radiation force which, in turn, can allow one to compute the acoustic pressure and thus absolutely calibrate the transducer. This approach is a variation of the sphere radiometer technique. For high-frequency calibration, however, the form function of the acoustic radiation force varies rapidly with frequency for ka≳1, where k is the wave number in the host liquid; therefore, the properties of drop and host must be known very accurately if the inversion process is to yield good estimates. If, however, the drop material is nearly impedance matched with the host, although the densities and sound velocities individually differ, the ‘‘resonances’’ in the form function are pushed to a much higher ka, and unambiguous inversion of the data over a wider frequency range to give accurate pressures is possible. Theoretical and experimental results will be presented, as will other possible acoustic radiation force configurations that overcome some of the limitations of levitation and sphere radiometry techniques. [Work supported, in part, by a U.S. National Institutes of Health Grant No. 5-RO1-CA-39374.]
Localization of narrow-band sounds in the medial sagittal plane.
Authors: Rakerd Brad , Hartmann William Morris
Published: October 2005

Many experiments have shown that sources in the medial sagittal plane are localized by human listeners according to spectral cues. Therefore, narrow-band sounds in different frequency ranges are perceived as originating from different vertical locations. To further elucidate the cues involved in this effect, listeners were asked to localize two kinds of sound; a sine tone and one-third-octave band of noise. Stimuli were presented, at random, from one of three sources: front, overhead, and rear. The listener’s task was to identify the source. The experimental results supported the spectral theory of localization in that, for both kinds of sound, judgments depended more upon frequency than upon actual source location. However, the results also showed that judgments were considerably more stable and consistent with the one-third-octave bands of noise than with the sine tones. It is suspected, therefore, that the spectral theory of vertical localization is true but that it is incomplete. The results are interpreted to mean either that listeners are capable of using head-related transfer-function cues on a band that is as narrow as one-third octave or else that listeners gain vertical-plane localization information from spectral fluctuations, present in the noise but not in the sine tone. [Work supported by the NIDCD, DC00181.]
Suppressor duration effects in forward masking.
Authors: Thibodeau Linda M. , Champlin Craig A. , Stiritz Lori
Published: October 2005

It has been shown that as the duration of a suppressor increases, the amount of suppression increases in experiments where the suppressor duration was equal to or less than the masker [D. L. Weber and D. M. Green, J. Acoust. Soc. Am. 64, 1392–1399 (1978); R. V. Shannon, ibid. 59, 1460–1470 (1976)]. This suggests that the degree of suppression may be related to the time of interaction between the masker and suppressor. The purpose of this experiment was to evaluate the effects of suppressor duration when the time of masker–suppressor interaction was held constant. Forward-masked thresholds for the 2000-Hz probe were measured with a 2000-Hz masker only (M), and as a function of suppressor duration when the onset of the suppressor preceded the masker (M+S). When the suppressor preceded the masker by 200 ms, the amount of suppression was minimal or absent. The possibility that the threshold change resulted from increased masking by the longer duration suppressor was evaluated by measuring masked threshold for the suppressor alone and by delaying the probe onset relative to the masker–suppressor offset.
How many tests should an auditory performance battery contain?
Authors: Canevet Georges , Divenyi Pierre L.
Published: October 2005

One primary objective of performance batteries is to select a set of tests that maximizes differences among subjects with a minimum of time investment. The present study proposes a method apparently well suited for making such a selection. Auditory performance of 51 normal-hearing young volunteers was assessed in 24 tests designed to evaluate spectral, temporal, intensity, sequential, and binaural/spatial processing. From the data, 15 measures were extracted. By means of a hierarchical cluster analysis, subjects were divided into four groups, on the basis of the 15 performance measures. Discriminant analyses, with an increasingly restrictive criterion for the retention of measures in the three canonical discriminant functions were applied to the same data in order to identify the smallest subset of measures that could classify the subjects into the four groups with an acceptable (≳95%) success rate. The operation unveiled a subset of eight measures that yielded error-free classification of our subject sample, thereby suggesting that the other seven measures could be bypassed, thus requiring a substantially shorter testing time. [Work supported by the Direction des Constructions et Armes Navales, Toulon, France, the Department of Veterans Affairs, and the National Institutes on Aging.]
Acoustic analysis of vowels produced by young and elderly women in spontaneous speech.
Authors: Brundage Shelley B. , Liss Julie M.
Published: October 2005

Recent investigations suggest that analysis of spontaneous speech, rather than citation form speech, may have greater explanatory power for models of speech production and perception. The present study sought to establish a preliminary database of selected acoustic characteristics of vowels produced by women. Acoustic analyses were performed on the spontaneous and citation form speech of five elderly and five young female subjects. Measurements included segment durations and F1 and F2 temporal midpoints for monophthong vowels, and formant trajectory characteristics for selected diphthongs. Vowels were grouped for analysis by syllabic and phrase level stress and by phonemic environment. Acoustic characteristics of vowels will be illustrated by F1–F2 plots and distribution histograms. Results will be discussed relative to speaker characteristics, speaking condition, and how these acoustic measures may relate to speech motor control.
Combined effects on vowel duration.
Authors: Behne Dawn M. , Nygaard Lynne C.
Published: October 2005

The duration of a vowel is dependent on its inherent quality, and can be affected by speaking rate, sentence position, focal stress, and voicing of a following consonant. The goal of this study was twofold: (1) to confirm the independent effects of each of these factors on vowel duration; and (2) to describe their effects when combined in the same sentence. Using /i, o, a, æ/, conversations were developed with /kVt/ and /kVd/ as target words. In each conversation a target word occurred in initial or final sentence position, and was either focused or nonfocused by the discourse. Subjects produced each conversation at three speaking rates. The results indicate that (1) as expected, vowel duration differed among vowel qualities and was affected by speaking rate, sentence position, focal stress, and voicing of a following consonant; and (2) when converging on a vowel, the investigated factors influence vowel duration independently. These findings suggest that the effects are simultaneously reflected in vowel duration by being superimposed on one other.
Description of low-frequency acoustic scattering from ocean surface features using coupled normal modes and integral series representation.
Authors: Hayward Thomas J.
Published: October 2005

The coupled normal mode description developed by Boyles for acoustic propagation and scattering from a pressure-release surface in a waveguide is applied to the description of scattering from ocean surface features. An exact transformation leading to equations for the forward- and backward-propagating acoustic fields is utilized. These equations are solved using standard numerical methods for ordinary differential equations. Alternatively, an integral series representation of the solution of the coupled mode equations permits both assessment of the validity of single-scatter approximations and the computation of small-ka approximations. The mode-based calculations are compared with previous scattering descriptions for simple geometric shapes of the scatterers. Simulations based on these calculations test the applicability of matched-field techniques to localization of boundary scatterers using both point-source and extended-scatterer replicas. [Work supported by the ONR Arctic Sciences Program.]
Exact scattering theory applied to more realistic 2-D sea surfaces.
Authors: Keiffer Richard S. , Dubberly John , Resio Don , Novarini J. C.
Published: October 2005

The notion that the scattering at the air–sea interface is insufficient to account for experimentally determined backscattering strengths at low grazing angles is based almost exclusively on the application of various scattering theories to sea surfaces that are described by steady-state, essentially 1-D sea surface spectra. In reality, and particularly for higher wind speeds where the steady state is rarely achieved, these spectra may not be representative of the seas encountered during the acoustics experiments and certainly do not describe seas that are due to time and spatially varying forcing conditions. For example, during the Chapman–Harris experiment wind speeds ranged from 0 to 30 kn over a 52-h period. Recently, NOARL funded OCTI to model the sea surface spectra encountered during the Critical Sea Test-4 (CST-4), which took place in the Gulf of Alaska. OCTI’s wave mode, WAVAD, takes into account time varying local and regional forcing conditions and predicts the 2-D sea surface spectrum. From these spectra, 2-D realizations are generated via a linear filter technique. The scattering theory employed is an exact integral equation approach [E. I. Thorsos, J. Acoust. Soc. Am. 83, 78–92 (1988)] that has been applied to 2-D surfaces. In this paper, a preliminary report is made on the low-frequency backscatter from realistic 2-D sea surfaces. [Work supported by ONR and ONT.]
Fine-scale biological and physical structure in stratified waters.
Authors: Donaghay Percy L.
Published: October 2005

A new sampling technique is described that simultaneously allows collection of centimeter scale vertical profiles of plankton and the physical, chemical, and biological factors that can influence those distributions. A high-resolution electronic profiler is first used to define the vertical structure on centimeter scales of a series of electronically sensed parameters, then real-time analysis of data from the profiler is used in a second cast to collect pumped and siphoned samples from discrete layers for chemical and biological analysis. Application of these techniques in a permanently stratified estuary indicates that vertical distributions of zooplankton, their food resources, and potentially controlling biological, chemical, and physical factors can vary by up to an order of magnitude per 10 cm. A numerical analysis of these measured fine-scale structures indicates that conventional coarse scale sampling techniques can miss entire communities, grossly underestimate food resource availability, underestimate the sharpness in chemical and biological gradients that limit the vertical distributions of zooplankton, and confound efforts to evaluate acoustic sensor performance in the ocean. [Work supported by ONR and Sea Grant.]
Acoustical mapping of zooplankton patchiness in the sea.
Authors: Greene Charles H. , Wiebe Peter H.
Published: October 2005

Sound scattering by zooplankton is a major source of high-frequency, volume reverberation in the sea. The importance of zooplankton in the formation of sound-scattering layers (SSLs) has long been recognized. Only recently, however, has this sound scattering been quantified to map and characterize the distributions of marine zooplankton. Here, new methods of acoustically mapping and analyzing the fine- to coarse-scale structure (‘‘patchiness’’) of zooplankton distributions are reported. Examples of two-dimensional mapping and statistical analyses will be drawn from submersible-collected data of the fine-scale horizontal patchiness within krill SSLs in the Gulf of Maine; examples of three-dimensional mapping (i.e., three-dimensional volume rendering) and statistical analyses will be drawn from ship survey data of coarse-scale zooplankton patchiness in the waters above Fieberling Guyot in the north Pacific. [Work supported by ONR Oceanic Biology.]
Fast field program analysis of sound propagation from C-135A aircraft runups.
Authors: Raspet Richard , Lundberg Wayne
Published: October 2005

The Armstrong Laboratory Noise Effects Branch has taken extensive data from C-135A aircraft ground runups over a 2400-m path at Wright-Patterson Air Force Base. The temperature was measured at 0.3- , 1.5- , and 9.1-m height and the wind velocity and direction were measured near 2.0-m height. These data were analyzed using multivariate analysis and have been used to develop excess sound attenuation algorithms for use in the NOISEMAP model for predicting noise impacts around airports. This paper reports on the analysis of selected data from these tests with the fast field program (FFP). The emphasis of this study is to determine the usefulness of the FFP for this type of data analysis and to establish procedures for developing sound velocity profiles from the limited meteorological data. The FFP predictions agree well with the data for downward refraction conditions when the sound levels are high, and are poorest for upward refraction conditions. Since high levels dominate the average noise impact, the FFP should be useful predicting environmental impacts around airports. [Work supported by System Research Laboratory.]
Annoyance caused by different simultaneous environmental sounds.
Authors: Vos Joos
Published: October 2005

To prevent or reduce the annoyance in residential areas with two or more different sound sources, it is relevant to develop a quantitative model in which the sound exposure levels from the separate sources are combined into one overall sound exposure measure that adequately represents total noise annoyance. The literature shows that many field studies have had difficulty obtaining valid total annoyance ratings, and that the more persuasive data are too rough to model total annoyance. In the present laboratory experiments, (total) annoyance ratings were obtained for various combinations of impulse, road-traffic, and aircraft sounds. Separate dose-response relationships for the three isolated sounds were also determined. The results support a relatively simple model in which the impulse and/or aircraft sound exposure is expressed in the A-weighted Leq of equally annoying road-traffic sound by applying level-dependent penalties, and the total sound exposure level is obtained by a summation of the corrected Leq’s of the various single sources. Advantages of the method used and applications of the present model will be discussed.