bluuebunny/crossref_metadata_2025 huggingface dataset:
An analytical decomposition of the Biot–Tolstoy edge diffraction model
Authors: Svensson U. Peter , Andersson Roger , Vanderkooy John
Published: July 2002

The sound field for a rigid, infinite wedge which is irradiated from a point source has an explicit impulse response solution which has been given by Biot and Tolstoy. This solution has been the basis for methods to calculate sound fields by summing up all specular reflections from finite surfaces and edge diffraction components from finite edge segments. However, these methods have usually relied on assumptions about the solution for finite edges. In this paper an edge diffraction model is presented which uses directional edge sources and analytical expressions for the directivities are derived to give exactly the Biot–Tolstoy solution for an infinite wedge. Unlike other models, with this new model explicit expressions can be found for certain geometries, such as the on-axis scattering from a rigid circular disc, for which both the first-order and the second-order edge diffraction can be derived. Furthermore, this model can be used to incorporate low-order edge diffraction into room acoustics prediction programs, which should improve them significantly at low frequencies.
Uncertainty of acoustic calibrations performed at different ambient pressures
Authors: Frederiksen Erling
Published: July 2002

The sensitivity of measurement microphones and the sound pressure produced by acoustic calibrators can be highly influenced by the ambient static pressure. Therefore such transducers should ideally be calibrated at a static pressure which is essentially equal to that existing at the site where the sound measurement is performed. However, in general, this is not possible and not necessary. Most often it is sufficient to know the typical pressure coefficients valid for the applied type of microphone and calibrator and to use those either for correction of the measurement results or just for estimation of the uncertainty. For few applications, like noise approvals of aircraft, it might be worth the costs and efforts to calibrate the transducers at the measurement conditions. This paper describes how the Brüel & Kjaer Reciprocity Calibration System Type 9699 which includes a pressure chamber was used for accurate and quite fast determination of pressure influence on laboratory and working standard microphones and on sound level calibrators. Calibration results and estimated uncertainties will be presented for some frequently applied types of microphone and calibrator.
Precision grade sound power measurement using the sound intensity scanning technique: Comparison of methods for accuracy estimation
Authors: Keith Stephen E. , Chiu Vincent , Bly Stephen H. P.
Published: July 2002

To assist the development of a precision scanning intensity standard for sound power determinations, this study investigated two accuracy estimation methods proposed by ISO Working Group 25. One method used the F4 sampling criterion from the precision grade discrete point intensity standard [ISO 9614-1:1993], adapted for the scanning intensity technique. The other method assumed that measurement accuracy was related to the difference between measurements using two different scanning paths. This method was a variation of the repeatability test from the engineering grade scanning intensity standard [ISO 9614-2:1996]. The study compared the estimated accuracy from each of the two methods to a third accuracy estimate, the difference between a sound power determination using the scanning intensity technique and a precision grade measurement using sound pressure [ISO 3745:1977]. All measurements were made under hemi-anechoic conditions and all intensity scannning was done manually. Results were obtained for the following three sound sources: a Bruel & Kjaer reference source, a vacuum cleaner, and a loudspeaker. The results suggested that both of the proposed methods can be of use in achieving precision grade accuracy.
A study of sonic boom effects on seals breeding on Sable Island, Canada
Authors: Perry Elizabeth A. , Boness Daryl J. , Insley Stephen J.
Published: July 2002

Researchers were on Sable Island, Canada (440°N, 600°W), for the 1997 grey seal (Halichoerus grypus) breeding season to study the affects of sonic booms on seal behavior. The physical parameters of 36 sonic booms associated with the Concorde’s trans-Atlantic flights were recorded. The average minimum overpressure was −0.568 PSF and mean maximum overpressure was 0.935 PSF. Data were collected on frequency of aggression, movements, and distances moved in 10-min periods before and after booms to detect changes in general agitation of pups and adults. The number of mother-pup pairs nursing and number of nursing bouts interrupted during the minute of sonic booms were compared to those in randomly chosen minutes to measure immediate startle responses of seals. There was no difference in the behavior of adults nor pups between 10-min periods before and after booms, nor did their behavior differ in the minute of booms when compared to randomly chosen control minutes. Female heart rates did not vary before, during and after booms but pup heart rates were variable. A similar study on Sable Island harbour seals (Phoca vitulina) is being conducted for comparison.
Bubble size distribution near a pressure antinode
Authors: Mettin Robert , Luther Stefan , Ohl Claus-Dieter , Lauterborn Werner
Published: July 2002

When strong ultrasonic standing wave fields are generated in liquid-filled resonators, cavitation bubbles emerge and build dynamic structures, typically near the pressure antinodes. For technical applications like sonochemical reactions taking place in the resonator, the size distribution of the clustering bubbles is an important parameter. New photographic measurements of bubble sizes are presented, along with numerical size estimates. These are based on Bjerknes force and shape stability calculations for spherical bubbles. [Work partially supported by ‘‘Graduiertenkolleg Strömungsinstabilität und Turbulenz.’’]
Viability of plant cells in ultrasonic fields
Authors: Böhm Hannes , Lowe Kenneth C. , Briarty L. Gregory , Power J. B. , Davey Michael R. , Benes Ewald
Published: July 2002

The influence of ultrasonic waves on suspension plant cell cultures (e.g., petunia and carrot) has been investigated. The dependency of viability, cell density, miototic division, and regeneration, on the controlled acoustic parameters’ frequency of sound, energy density, treatment time, and wave type, is presented. Special focus is on the investigation of the influence of propagating plane waves, because these allow a spatially almost invariable acoustic pressure amplitude and, as a consequence, the same acoustic parameter values for all cells in the treated suspension volume. [Work supported in part by the European Commission, Contract No. ERBFMRXCT970156.]
Acoustic deinking of paper: Pulse effectiveness in particle detachment
Authors: Madanshetty Sameer I. , Chandran Jogesh B.
Published: July 2002

Deinking remains an important step in environmentally conscious manufacturing of paper. Removing xerographic ink from a print sample using acoustic microcavitation provides an easy method to observe context for studying cavitation-induced erosion at surfaces. Experiments indicate that microcavitation evolves microimplosions which are effective in causing deinking preferentially at ink sites. Acoustic microcavitation is brought about by low megahertz acoustic fields giving rise to micron-size bubbles that live a few microseconds. In exposing a surface to continuous waves for a defined duration, one could obtain cavitation effects in an average, overall sense; the details of nucleation, evolution of inertial events, and the precise interplay of field parameters in effecting cavitation, however, get glossed over. Studying pulsed cavitation using tone bursts at low duty cycles, instead of CW insonification, reveals interesting details of the initiation and evolution of acoustic microcavitation. Specifically, it is found that short-pulse deinking is more effective than long-pulse deinking. This fact seems to raise an anomalous question of how the sample knows it is being visited by a short pulse or a long pulse. Recent experiments operationally demonstrate why short-pulse deinking is more effective. [Work supported by NSF.]
Time series analysis of cavitation bubble fields
Authors: Luther Stefan , Parlitz Ulrich , Lauterborn Werner
Published: July 2002

Acoustical cavitation bubble fields exhibit spatio–temporal structure formation on multiple scales involved. The temporal scales are the fast, medium, and slow scales given by the period of the driving, the typical relaxation time of the individual bubble motion, and the drift of the filaments, respectively. The spatial scales are the microscopic, mesoscopic, and macroscopic scales given by the bubbles’ mean radii, typical bubble distances, and the wavelength of macroscopic disturbances (e.g., sound waves). Prominent examples of the observed structures are, for example, period doubling on a fast temporal scale and a slowly drifting filamentlike bubble distribution on the macroscopic spatial scale. These structures are experimentally investigated using simultaneously measured sequences of images (CCD camera) and acoustic time series (hydrophones). Based on these data the experimentally observed patterns are characterized using image processing techniques (e.g., principal component analysis) and by means of methods of nonlinear time series analysis (e.g., test for stationarity, dimension estimates, surrogate data analysis). The investigations focus on the correlation between spatial and temporal structures on different scales and transitions between them. [Work supported by ‘‘Graduiertenkolleg Strömungsinstabilität und Turbulenz.’’]
Effects of internal waves and bores on acoustic transmissions in the Strait of Gibraltar
Authors: Tiemann Christopher O. , Worcester Peter F. , Cornuelle Bruce D. , Send Uwe
Published: July 2002

The Strait of Gibraltar Acoustic Monitoring Experiment was conducted in April 1996 to determine the feasibility of using acoustic methods to make routine, rapidly repeated transport measurements in the Strait of Gibraltar, as well as to explore the acoustic scattering caused by the unique internal wave field in the Strait. The acoustic data from high-frequency (2 kHz) reciprocal transmissions across the Strait are unique in that they clearly isolate the acoustic effects of passing internal bores without the added complexity of surface and bottom interactions. Although the acoustic scattering caused by each passing internal bore is different, some common characteristics can be identified. The challenge is to explain these acoustic observations in terms of the physical processes occurring. Acoustic propagation models through range- and time-dependent sound-speed fields representing the Strait and perturbed by internal bore models are being used to understand these observations and to determine to what aspects of the bore the acoustics are most sensitive.
Performing room surfaces’ effects on the acoustical condition distributions
Authors: Erdem Aknesil Aye
Published: July 2002

To obtain suitable acoustic conditions for the important rooms from the room acoustics point of view, some parameters have to be evaluated. For example, reaching the acceptable reverberation time limits, not the place of surfaces but the total absorbtion, has an important role according to the size of surfaces and absorbability. On the other hand, while designing the first reflections or avoiding the constitution of acoustical defects like echo, the place of the suitable surfaces is important. When considering the importance of the distribution of acoustical conditions, the question is posed of how the surface affects this matter. In line with this observation, the aim of this study is to find the effects of the surfaces according their ratios.
Virtues and perils of an empiricist approach to speech perception
Authors: Kluender Keith R. , Lotto Andrew J.
Published: July 2002

Nearey’s “double-weak” approach [J. Acoust. Soc. Am. 102, 3241–3254 (1997)] advocates a pattern-recognition model in answer to fundamental problems of speech perception. Both theoretically and empirically, there is much to recommend double-weak. However, there is some question whether this approach avoids long-standing disagreement with respect to the objects of speech perception. In addition, the descriptive power of even relatively simple empiricist models such as Nearey’s can mislead with respect to fundamental articulatory and auditory processes underlying speech perception. Here, the positive contributions are celebrated, and several cautionary observations—germane to broader questions of experience and learning—are offered.
An L∞-norm estimator for environmentally robust, shallow-water source localization
Authors: Harrison Brian F.
Published: July 2002

In this paper, the L∞-norm estimator for robust matched-field source localization in the presence of environmental uncertainties is proposed. This estimator is derived from the maximum a posteriori (MAP) estimator by interpreting MAP as an exponentially weighted average over environmental realizations. In the presence of infinite averaging, MAP provides an optimal approach to robust matched-field source localization. However, in practice only a finite number of environmental realizations can be included in this average, resulting in a suboptimal processor. With finite environmental sampling, the L∞-norm estimator provides superior performance over that of MAP. Also introduced is the concept of wave number gradients which provide physical insight for the performance of the L∞-norm estimator. Simulation results from two shallow-water environments are presented, which show that at moderate signal-to-noise ratios, the proposed estimator provides nearly 90%-correct localization performance compared to less than 70% for MAP. Results using experimental data from the Mediterranean Sea are also presented.
Performance of a two-stack annular prime mover
Authors: Carter Brent A. , Atchley Anthony A.
Published: July 2002

In a previous study, numerical investigations of a two-stack annular thermoacoustic prime mover were made [H. T. Lin et al., J. Acoust. Soc. Am. 102, 3071(A) (1997)]. The conditions for onset were predicted. The subject of this paper is an experimental investigation of a two-stack annular prime mover. The prime mover, modeled using DeltaE and a matlab program, is constructed using an annular aluminum resonator, copper fin heat exchangers, and two parallel plate, glass stacks. The relative position of the two stacks is adjustable within the resonator. Fluid-cooled ambient heat exchangers, and cartridge-heater powered hot heat exchangers are used to apply independent temperature differences across the two stacks. The sound field and temperature within the resonator are measured with a series of microphones and thermocouples distributed along the resonator. The performance of the prime mover is investigated as a function of relative stack position, temperature differences applied to the stacks, and heater power levels. [Work supported by the Office of Naval Research and The Pennsylvania State University Applied Research Laboratory.]
How much pressure drop is necessary for heat transfer?
Authors: Rülich I. , Quack H.
Published: July 2002

In many technical applications, heat has to be transferred from a fluid to a wall. To do this efficiently in a short time and confined volume, the fluid has to be moved by applying a pressure difference. In refrigeration and power processes working with ideal gases, the pressure drop in the heat exchangers plays an important role in the overall efficiency of the cycle. The present work investigated whether the necessary pressure drop to obtain a certain amount of heat transfer can be derived from basic principles at least for Reynolds numbers below 2000. It was found that there exists a lower limit on the order of magnitude of 2 pressure heads per transfer unit. This figure is slightly dependent on the Prandtl number of the fluid. Real heat exchanger geometries have values higher than this minimum value, the magnitude depending on simple shape factors, which were derived from numerical fluid flow calculations and with published experimental data. This work identified how the ideal inner geometry of a regenerator for regenerative power and refrigeration processes should look like. [Work is supported by BMBF Grant No. 13N6619/0.]
Directivity of a uniform-strength, continuous circular-arc source phased to the spatial position of its diameter
Authors: Thompson William
Published: July 2002

An expression is derived for the in-plane directivity function of a uniform-strength, circular-arc source (or receiver) wherein the output of each element of the arc is phase shifted by the wavelength distance from that element to the spatial position of the diameter of the circle which is parallel to the chord joining the end points of the arc. Numerical results are presented for a selected set of arc sizes and values of the radius-to-wavelength ratio. The results resemble the directional characteristics of a true straight line source of the chord length. The special cases of a phased semicircular arc and a phased complete circular arc are discussed. The directivity function of a uniform-strength, unphased, continuous circular-arc source, a previously established result, is also presented for comparison.
Nonlinear pulse propagation in arbitrarily dispersive media: Tube waves in permeable formations
Authors: Johnson David Linton
Published: July 2002

An approximate quasistatic equation, analogous to the Burgers equation, is derived to account for the combined effects on tube wave propagation of (a) dispersion/attenuation in permeable formations, and (b) quadratic nonlinearity of the fluid and of the formation. Numerical results for weak nonlinearity and narrow-band pulses indicate that pulse self-demodulation does occur, but over relatively large distances because of the relatively low-frequency band relevant for tube wave propagation in characteristic borehole geometries (f<10 kHz). The self-demodulated pulse shape can be very significantly distorted from that predicted by the conventional Burgers equation, depending upon the choice of relevant parameters such as the permeability, the carrier frequency, and the mudcake membrane stiffness. Numerically exact analytical formulas for the self-demodulated pulse shape, as well as for the energy in the second harmonic band, are derived for cases in which the pulse duration is long and the nonlinearity is relatively weak. These formulas are valid for any arbitrary dispersion/attenuation mechanism, and not just tube waves in permeable formations, as long as the propagation wave vector may be specified uniquely as a function of frequency.
Sound scattering cross-section in a stratified moving atmosphere
Authors: Ostashev V. E. , Goedecke G. H. , Wood R. , Auvermann H. , Clifford S. F.
Published: July 2002

For a realistic model of a stratified moving atmosphere with arbitrary vertical profiles of the adiabatic sound speed and wind velocity vector, an equation is derived for the sound scattering cross-section per unit volume, σ, as a function of apparent scattering angle Θ0. The effects of these profiles on σ are studied numerically. It is shown that if the wind velocity is zero, but the adiabatic sound speed varies with height, then σ can be affected significantly for Θ0<110°. Furthermore, if the wind velocity varies with height, but the adiabatic sound speed does not, then σ can be affected significantly for Θ0<110° as well as Θ0 near 180°. These and other numerical calculations have shown that in many cases acoustic remote sensing of the structure parameters of temperature and wind velocity fluctuations in the atmosphere should be based on the derived equation for σ rather than on that used in the literature. The derived equation for σ is also compared to those obtained by Clifford and Brown [J. Acoust. Soc. Am. 55, 929–933 (1974)] and by Ye [J. Acoust. Soc. Am. 102, 754–758 (1997)] for a model of a stratified moving atmosphere as two homogeneous layers in relative motion. It is explained why the predictions by Clifford and Brown are different from those by Ye.
Comparing the naturalness of several approaches for generating F0 contours in German text-to-speech systems
Authors: Mixdorff Hansjrg , Mehnert Dieter
Published: July 2002

Generating near-to-natural F0 contours is an important issue in text-to-speech synthesis and contributes vastly to the quality of synthetic speech. In earlier studies by the authors, a model of German intonation was developed that is based on the quantitative Fujisaki model. A typical F0 contour is described as a sequence of major rises and falls, which are modeled by onsets and offsets of accent commands connected to accented syllables. The current paper addresses perception experiments comparing the intonational naturalness of a Fujisaki-model-based TTS and four other German TTS systems with comparably high segmental quality. Natural speech samples were used as a reference. Three of the TTS systems had PSOLA, and one LPC segmentals. Two types of experiments were conducted with 20 subjects: (1) a pair comparison of 15 isolated sentences, (2) a ranking test based on a news passage of about 15 sec produced with each of the systems. Preliminary results from experiment (1) show, that on a naturalness scale from 0 to 5, the natural speech samples reach a maximum score of 4.5, with values of 2.8 for the best synthesis, the LPC-based one. The system with Fujisaki-model-based intonation leads the group of PSOLA systems, which is closely clustered at a mean of 2.4.
Influences of various factors upon parameters of the command–response model for fundamental frequency contour generation
Authors: Ohno Sumio , Hara Yoshikazu , Fujisaki Hiroya
Published: July 2002

The command–response model by Fujisaki and his co-workers formulates the generation process of the fundamental frequency contour (henceforth F0 contour) in terms of a set of input commands carrying linguistic and paralinguistic information and the mechanisms that respond to these commands. The parameters of the mechanisms are time constant of the phrase control mechanism (α), that of the accent control mechanism (β), and the baseline frequency (Fb). The present study investigates the influences of speech rate, speaking style, and individual difference on these parameters as well as their variability. The speech material consisted of recordings of a short story by six native speakers of Japanese at three speech rates (slow, normal, and fast) and in two speaking styles (reading and conversational). The parameters were extracted by the method of analysis-by-synthesis, and the results were analyzed statistically. The analysis indicated that utterance-to-utterance variations are quite small in all three parameters for a given rate, style, and speaker. Among the three parameters, only α showed a small but systematic tendency to increase with the speech rate, while differences in speaking style did not affect these parameters. Individual differences were quite small in α and β, while Fb varied from speaker to speaker.
Application of the command–response model to the analysis, interpretation, and synthesis of fundamental frequency contours of speech of various languages
Authors: Fujisaki Hiroya , Ohno Sumio
Published: July 2002

A command–response model has been presented by Fujisaki and his co-workers initially for the process of generation of the fundamental frequency contour (henceforth F0 contour) of the common Japanese. It consists of a set of input commands carrying linguistic and paralinguistic information, and the mechanisms that respond to these commands to generate both phrase and accent components, which, together with a baseline value, constitute the actual F0 contour. Subsequent works have shown that the model is applicable also to F0 contours of some other dialects of Japanese as well as of other languages including Chinese, English, German, Greek, Spanish, and Swedish. It has been argued, however, that the model may not be able to generate certain contour types that are not found in Japanese but are commonly used in other languages [e.g., P. Tayler, Speech Commun. 15, 183]. This paper shows how these contour types can actually be generated by the same mechanisms, with certain language-specific timing, polarity and magnitude of the input commands, all within the framework of syntactic and prosodic constraints of each language. The results will be demonstrated by resynthesized speech, and testify the validity and usefulness of the model for multilingual text-to-speech synthesis.
On the waveforms and spectra of glottal flow models
Authors: Doval Boris , d’Alessandro Christophe R.
Published: July 2002

Several glottal flow models have been proposed for speech analysis and synthesis (e.g., LF, Rosenberg, R++, and Klatt). All these models do not use the same number of parameters, or the same name for similar parameters, and it appears difficult to compare their merits. Then, a unified framework for studying the time and frequency domain properties of glottal flow models is proposed. It is shown that all the models can be represented by a common set of five time-domain parameters: three scale parameters (T0, peak amplitude, open quotient), a shape parameter (asymmetry quotient), and a closure continuity parameter. A generating function is computed for each model by normalization of the model with respect to scale parameters and closure continuity parameter. The specific features of each model are represented in its generating function. The spectrum of generating functions is low pass, and its derivative can be characterized by a spectral maximum, coined ‘‘glottal formant.’’ The closure continuity parameter corresponds to a spectral tilt component. The scale parameters are interpreted using scaling properties of the Fourier transform. Then, the glottal flow spectra can be characterized by two breakpoints. Frequencies of these breakpoints can be computed analytically for each model parameter setting.
The representation of spectrally dynamic signals in primary auditory cortex
Authors: Shamma Shihab
Published: July 2002

Natural sounds, including speech, are traditionally characterized by their patterns of pitch, timbre, loudness, and modulation, all of which are closely related to the instantaneous spectrum of these signals. The central auditory system has developed elegant mechanisms to encode this spectro-temporal information. Thus, at the level of the primary auditory cortex (AI), the dynamic spectrum is repeatedly represented over a wide range of spectral and temporal resolutions. This multi-scale representation is derived from response patterns of cells selective to such parameters as local bandwidth, asymmetry of spectral peaks, and onset/offset transition rates. Physiological experiments measured the spectral and dynamic properties of neurons in AI using linear system analysis. One set of experiments used broadband signals with sinusoidally modulated spectral envelopes (‘‘ripples’’). Varying the density (or frequency), amplitude, phase, and velocity of the ripple pattern in a systematic fashion was used to derive a ripple transfer function. From this function it is possible to derive a spectro-temporal (impulse) response function via an inverse Fourier transform. A separate technique computed the neuronal impulse response (via reverse correlation) using signals with random and dynamic spectral characteristics. This presentation summarizes the application of these two analytical techniques to characterizing AI neuronal responses and discusses the scientific and technical implications of these multi-scale, cortical representations.
Representation of consonant–vowel syllables in mammalian auditory cortex: Effects of background noise and speaking rate
Authors: Schreiner Christoph E. , Wong Sarah , Bonham Ben , Cheung Steve , Bedenbaugh Purvis
Published: July 2002

The representation of synthetic and naturally spoken consonant–vowel (CV) syllables was investigated in the auditory cortex of the cat and the squirrel monkey. Spatially discrete and temporally synchronized patterns of neuronal activation are systematically distributed across the tonotopic axis. This activation pattern is largely determined by the dynamics of the signal’s spectro-temporal envelope (such as the energy modulation associated with plosive consonants and the onset of voicing). The spatio-temporal activity is strongly influenced by sound pressure level, resulting in significant changes in neuronal response strength and distribution (analogous to those associated with sinusoidal signals). In contrast, background noise has a much smaller impact on the excitation pattern, except at levels sufficiently high as to cause extensive suppression of overall activity. Varying the speaking rate (i.e., shortening vocalic durations and the interval separating CV syllables) was systematically investigated. Forward masking and adaptation affect the magnitude, latency, and spatial distribution of neuronal activity for both slow and rapid speaking rates. These effects are differentially distributed across cortical fields. [Work supported by grants from NINDS and NSF.]
Phase conjugation and matched-field processing
Authors: Kuperman W. A. , Hodgkiss W. S. , Song H. C. , Thode Aaron , Akal T. , Ferla C.
Published: July 2002

Matched-field processing (MFP) can be implemented with a backpropagation algorithm. The method requires the same knowledge of the environment as the conventional MFP replica search approach. Rather than use numerical backpropagation, the phase conjugation (PC) approach (also referred to as a time-reversal mirror) uses physical sources at each receiver position to backpropagate the phase-conjugated data on the receive array. The acoustic field then focuses at the position of the original radiating source that is the same position of the main lobe of the analogous MFP ambiguity surface. No a priori environmental information is required. Two experiments implementing PC in the ocean have been successfully conducted in 1996 and 1997. These experiments and the physics of PC provide new insight into MFP.
Optimal amplitude shading for arrays of irregularly spaced or non-coplanar elements
Authors: Gallaudet Timothy C. , de Moustier Christian P.
Published: July 2002

The majority of optimal amplitude shading methods for arrays of irregularly spaced or non-coplanar elements rely on numerical optimizations and iterative techniques to compute the desired weighting function because analytic solutions generally do not exist. Optimality is meant here in the Dolph–Chebyshev sense to provide the narrowest mainlobe width for a given sidelobe level. A simpler and more efficient technique to compute the shading weights for arbitrary line array shapes or element spacings is presented and it is shown that it is sufficient to sample the optimal Dolph–Chebyshev window, computed for a uniform line array of equivalent aperture length, at the element position of the nonuniform array. Examples are given for narrow-band plane-wave beamforming with curved arrays in which phase compensation is achieved by projecting the elements on a line tangent to the array. For the same mainlobe width, the resulting peak sidelobe levels are within 3 dB of a 30-dB Dolph–Chebyshev weighted uniform line array of equal aperture length and number of elements. Results are presented for computer simulations and for data collected at sea with the Toroidal Volume Search Sonar by the Coastal Systems Station, Panama City, Florida. [Work sponsored by ONR-NRL (Contract No. N00014-96-1-G913).]
Robust Kalman-based DOA estimator with variable fading memory
Authors: Kim Han-Su , Sung Koeng-Mo , Lim Jun-Seok
Published: July 2002

In this paper, VFM (variable fading memory) is introduced to the robust Kalman-based DOA (direction of arrival) estimator [Han-Su Kim and Koeng-Mo Sung, Proc. 16th ICA/135th ASA 1, 1327–1328 (1998)] for DOA estimation of uncorrelated narrow-band plane waves under the impulsive noise environment. To introduce VFM properly into the Kalman filter, the proposed algorithm is derived from a modified least-squares criterion with VFM for DOA estimation. Since the derived VFM can be adaptively calculated with respect to the estimation error, the proposed algorithm has the effectiveness to the impulsive noise in stationary or nonstationary situations. The introduction of VFM enables the time-varying DOA to be estimated properly and prevents the estimated error caused by the impulsive outlier from propagating into future time instants, adaptively decreasing the importance of past observations. The proposed method can make the small perturbation to the parameters using the Huber function, named M-estimation, and adaptively reduce the estimation error propagation caused by past observations of the impulsive noise using VFF. Compared to the conventional Kalman algorithms, simulations show that the proposed algorithm is robuster than any other conventional Kalman algorithms to the impulsive noise.
High-resolution modal mapping in a complex shallow-water environment
Authors: Becker Kyle M. , Frisk George V.
Published: July 2002

During March 1997, a modal mapping experiment (MOMAX) was conducted to measure the spatial variability of low-frequency sound propagation in shallow water [G. V. Frisk et al., J. Acoust. Soc. Am. 103, 3028(A) (1998)]. The fields created by cw sources in the frequency range 50–300 Hz were measured using an array of freely drifting hydrophone buoys equipped with GPS navigation. The resulting data consist of complex pressure mapped onto a high-resolution geospatial grid. Based on the tenets of normal mode theory, the wave number content of the field was extracted using a high-resolution implementation of the Hankel tranform technique [K. M. Becker et al., J. Acoust. Soc. Am. 103, 3029(A) (1998)]. Neglecting horizontal refraction, the resulting spectra can be mapped onto a corresponding spatial grid. Several modal maps are presented for various source/receiver configurations and frequencies with emphasis placed on the variablility of the wave numbers in space. These modal maps can then be used as the basis for inferring the local geoacoustic properties of spatially varying shallow-water waveguides. [Work supported by ONR.]
Effects of the time-varying ocean on broadband propagation in shallow water
Authors: Nielsen Peter L. , Bini-Verona Francesco , Jensen Finn B.
Published: July 2002

Prediction of underwater sound propagation is generally performed by assuming a time-invariant ocean. In May 1997 SACLANTCEN conducted an experiment south of Elba in the Mediterranean Sea with the aim of investigating the time-dependent effects of the ocean on sound propagation in shallow water. Broadband LFM signals (300 Hz to 7.5 kHz) were transmitted every 1 min for 36 h over a fixed 15-km propagation path. Extensive measurements of the environmental parameters were done during the transmissions in order to perform a meaningful correlation between the time-varying environmental and acoustic data. Previous analysis of the low-frequency part of the acoustic data [Jensen et al., J. Acoust. Soc. Am. 103, 2856(A) (1998)] show high stability in the first 6 h of propagation, but at longer transmitting time and higher frequencies a strong variability in the received acoustic signals is seen. The main features in the time-varying environmental parameters causing fluctuations in the acoustic signals are identified. These features are then used in a numerical propagation model in order to assess to what degree it is possible to predict the time dependence of the received acoustic data in this particular shallow-water area. [This work was supported by the EC under the MAST-III project PROSIM, Contract No. MAS3-CT95-0008.]
Modeling huge sound sources in a room acoustical calculation program
Authors: Christensen Claus L.
Published: July 2002

A room acoustical model capable of modeling point sources, line sources, and surface sources is presented. Line and surface sources are modeled using a special ray-tracing algorithm detecting the radiation pattern of the surfaces of the room. Point sources are modeled using a hybrid calculation method combining this ray-tracing method with image source modeling. With these three source types it is possible to model huge and complex sound sources in industrial environments. Compared to a calculation with only point sources, the use of extended sound sources is shown to improve the agreement with measured data.
Insect sounds in agriculture
Authors: Hickling Robert
Published: July 2002

Insects make sounds, either in the course of normal activity (for example, eating or moving about), or to communicate. To investigate activity sounds, it is generally necessary to have a quiet environment, which occurs naturally in grain or soil, and can be created artificially in a soundproof box or anechoic chamber. These sounds can be used to detect pests in agricultural commodities, such as rice weevils in grain, grubs in soil, fruit-fly larvae in fruit, and bollworm in cottom bolls. Examples can be heard on the internet [www.olemiss.edu/∼lee/]. A brief discussion is presented of sound transmission in grain and soil. Sound does not appear to be used for communication by larvae. However some grubs in soil may use sound to warn off other grubs from their feeding areas. Acoustic communication appears to be important to ants. Results are presented of sounds made by the black imported fire ant Solenopsis richteri. Examples can be heard on the internet [www.olemiss.edu/∼hickling/]. It is shown how ants can make use of the near-field properties of a simple sound source in air. [Work supported by the USDA.]
A comparison of surface transverse wave propagation on quartz and gallium orthophosphate
Authors: Bigler Emmanuel , Briot Jean-Bernard
Published: July 2002

Gallium orthophosphate (GaPO4) is a piezoelectric material (32-class) with the same symmetry as quartz, but with a higher electromechanical coupling factor and richer possibilities of temperature-compensated acoustic modes in a wide range of temperatures, mainly at high temperatures above 300°C. Surface transverse waves (STWs) can be generated on both quartz and GaPO4 crystals on Y-rotated plates, so it is interesting to compare both materials on this respect. It is shown that a temperature-compensated orientation exists on GaPO4 for STWs, but with a lower velocity (2500 m/s) than for quartz (5100 m/s). Propagation of STWs on GaPO4 under an array of metal strips or grooves is studied and dispersion curves for grooves, aluminum, and gold strips are presented. It is shown that energy trapping near the surface will occur in all cases, allowing STW resonators to be built on GaPO4. The sensitivity of STW devices on GaPO4 to metal thickness and geometrical parameters of electrodes will be investigated. Possible applications to high-temperature devices will be discussed. [This work is supported by CEC, Thomson Microsonics (France) and CNRS (France).]
Detection of foreign accent by L2 English listeners
Authors: Tsukada Kimiko
Published: July 2002

This study investigates the question of whether non-native listeners are capable of differentiating native from non-native English talkers when presented with a specific foreign accent. Seven Japanese listeners (six female, one male), five of whom have lived in English-speaking countries for a minimum of nine years, were asked to identify the native talker in a forced-choice paired comparison task. The stimuli used were monosyllabic English words (in /CVt/ and /CVd/) produced by Australian and Japanese talkers in which vowels were matched on their F1, F2 at the temporal midpoint and duration. Non-native listeners performed the task more poorly than the native listeners (60% vs 79%). It was observed that some adult non-native listeners have difficulty in recognizing even their own L2 speech production [H. Goto, Neuropsychologia 9, 317–323 (1971)]. Perception errors were spread quite evenly across 11 vowel types. However, vowels such as /ε/, /æ/, /ʌ/, and /ɔ/ induced more errors. This may be related to the acoustic vowel space of their L1 in which, unlike English, the only low vowel /a/ is not cramped by the neighboring vowels. The results revealed large interlistener variability which suggests a complex nature of L2 perception.
Active volume sound absorber
Authors: Mazanikov Alexander A.
Published: July 2002

The absorbtion cross section (ACS) of a passive volume sound absorber asymptotically equals its geometric cross section. On the other hand, the ACS of an active absorber principally could be arbitrarily large. However, the increase of the ACS results in a rapid growth of the near acoustic fields. This puts restrictions on the practically achievable values of the ACS. The talk presents the results of a simulation of sound field levels generated by active absorbers of spherical shape and different wave sizes in response to incident plane waves. It is shown that near the absorber the level of the sound field increases as compared to the passive absorber of the same size. In the far zone the field of the active absorber corresponds to the field of the passive absorber with larger geometric scale. It is demonstrated that sound energy is absorbed rather on the back side of the absorber then on its front face (with respect to the direction of incident wave).
The influence of millimeter-scale seafloor roughness on measured and predicted high-frequency acoustic backscattering strength
Authors: Briggs Kevin B. , Jackson Darrell R. , Stanic Steve
Published: July 2002

Seafloor microtopography, as measured by underwater stereo photogrammetry, is systematically characterized concurrently with other sediment properties as model parameters to be used for predicting high-frequency acoustic backscattering from the seafloor as part of field experiments. Typically, seafloor roughness is digitized at 2–5 mm intervals from 35- or 70-mm film transparencies with a stereocomparator capable of 1-mm-scale lateral precision and sub-millimeter-scale vertical accuracy. Relative seafloor height profiles are used to estimate roughness power spectra, which are parameterized as slopes and intercepts of the line fit to the roughness variance regressed on the spatial frequency. According to the composite roughness model developed by APL-UW, the decay of the roughness spectra with increasing spatial frequency (slope) and the spectral strength (intercept) can be used to determine the contribution of the seafloor roughness to the overall backscattering strength. Data from field experiments are presented in order to demonstrate the relationship between seafloor roughness at sub-centimeter scales and high-frequency acoustic scattering. Experimental acoustic data and model predictions are presented from a coarse, shelly sea floor off Jacksonville, Florida, a rippled fine sand in the Quinault Range off Washington, and a very fine sand with subtle seafloor roughness off Tirrenia, Italy. [Work supported by ONR.]
The influence of small-scale seafloor surface morphology on acoustic backscattering
Authors: Pouliquen Eric , Zerr Benoit , Lyons Anthony P.
Published: July 2002

Sediment transport induced by current flows and surface gravity waves often causes the formation of nonisotropic features of the seafloor interface. Depending on environmental conditions (e.g., current, sediment type, etc.), these directional ripples may display non-Gaussian or sharply peaked forms. Contrary to RADAR sea surface scattering, these two aspects of the morphology of the seabed interface have not been deeply considered in the study of scattering of acoustic waves from the seabed. Their influence on backscattering is performed using two-dimensional and three-dimensional models including a finite-difference time-domain model (FDTD), a model based on the integral equation method, and a model based on the Kirchhoff approximation. All these models compute the time response as well as an averaged scattering strength from realistic randomly generated surfaces and profiles. The results of the simulations and acoustic data at high frequency show a sensitivity of the backscattered signals and of the backscattering strengths to both the anisotropy and the sharp nature of the seafloor interface edges.
Individual noise sensitivity and performance on psychoacoustic tasks
Authors: Ellermeier Wolfgang , Eigenstetter Monika , Zimmer Karin
Published: July 2002

In environmental noise surveys, self-reported noise sensitivity turned out to be a major predictor of noise-annoyance reactions. Its relationship to basic measures of auditory functioning, however, has not been systematically explored yet. Therefore, in the present investigation, a sample of 61 unselected listeners was subjected to a battery of psychoacoustic laboratory procedures ranging from threshold determinations to loudness scaling tasks. No significant differences in absolute thresholds, intensity discrimination, simple auditory reaction time, or power-function exponents for loudness emerged, when the sample was split along the median into two groups of ‘‘low’’ versus ‘‘high’’ noise sensitivity on the basis of scores obtained from a psychometrically evaluated questionnaire [K. Zimmer and W. Ellermeier, Diagnostica 44, 11–20 (1998)]. Small but systematic differences were found in verbal loudness estimates, and in ratings of the unpleasantness of natural sounds, thus suggesting that self-reported noise sensitivity captures evaluative rather than sensory aspects of auditory processing.
Analysis of composite resonators for ultrasonic micromanipulation and separation
Authors: Gröschl Martin , Trampler Felix , Benes Ewald , Nowotny Helmut
Published: July 2002

Piezoelectric composite resonators for the separation or manipulation of particles dispersed in a liquid by means of acoustic forces are investigated theoretically and experimentally. Such devices typically comprise at least four material layers: a piezoelectric transducer, a carrier (e.g., glass), the liquid (suspension), and an acoustic reflector. A mathematical matrix model for multilayer structures [M. Gröschl, Acust. Acta Acust. 84, 432–447 (1998)] is used to calculate the electric admittance spectrum of the resonator and results are compared to measurements obtained with a specialized computer-controlled measuring system. Additionally, the model allows the calculation of the spatial course of electric and acoustic field quantities (e.g., displacement amplitude, energy flow, energy density) along the multilayer structure. To characterize resonators for particle separation, the resonator performance number is introduced, which is a measure for the stored acoustic energy in the liquid (suspension) with respect to the total electric energy supplied by the driving source. It is shown that, for a given setup, the performance number depends strongly on frequency. The frequency range of maximum performance numbers defines the optimal operating range of the particle separator. [Work supported in part by the European Commission, Project ♯ FMRX-CT97-0156.]
Monaural azimuth sound localization with linguistic and nonlinguistic stimuli
Authors: Muller Barbara S. , Bovet Pierre
Published: July 2002

Many studies have been conducted to measure monaural azimuth sound localization performances with different sounds varying in frequency and complexity, but few are using language. This experimental design included phonologically and semantically different sentences, that is, they either contained fricative consonants or not, and all sentences were presented right way out or reversed. Analysis of response correctness of 30 subjects showed that localization performance is better with the left ear than with the right ear, but that neither the phonological nor the semantic aspect of the stimuli have any influence on localization accuracy. Analysis of response times of the subjects revealed an effect of the stimuli’s nature. Reversed sentences as well as those containing fricative consonants need more time to be localized than other stimuli. These results suggest that monaural sound localization is independent of phonological or semantic attributes of the signal, but that linguistic stimuli are processed more quickly than nonlinguistic ones.