bluuebunny/crossref_metadata_2025 huggingface dataset:
Hammerson Hall, The Living Arts Centre in Mississauga, Ontario
Authors: Wolff Robert , Doria Damian
Published: July 2002

This paper describes how adjustable acoustics features, effective micro and macro shaping, low background noise levels, and appropriate seat count contribute to a high-quality flexible performance environment. Diagrams, graphs, and images explain the techniques. Hammerson Hall is a concert theater designed to provide suitable acoustic environments for such varied performance media as symphony orchestra, chamber orchestra, recital ensembles, opera, musical theatre, ballet, dance, popular music, speech plays, comedy, lectures, and meetings. The room accommodates 1233 patrons for events when an orchestra is in the pit or on a concert platform, but it can quickly accommodate 1315 patrons for events where seats are located on seating wagons on the orchestra pit lift. Hammerson Hall is basically rectangular in shape; however, architectural elements make it appear to be curvilinear. The room also has flexible acoustic absorption and volume. Horizontal balconies and shelves act as reflectors, lateralizing sound energy for acoustic music events.
Sopukka, the new experimental stage of the Turku City Theatre
Authors: Ruusuvuori Anssi
Published: July 2002

Sopukka, the new experimental stage of the City Theatre of Turku, Finland, was built inside the frame of the old small stage. The new stage was designed as a ‘‘black box,’’ with movable seating areas for 80 to 150 spectators. The main design goal was flexibility, which would allow as many different staging arrangements as possible. Because of the lack of traditional stage area, the main acoustical design goal was to have an even distribution of sound from any part of the room to any possible seating area. As the ceiling was too high for useful reflections, it had to be absorbed, and the reflecting surfaces were designed as separate elements. The first idea was to hang movable reflecting panels from the ceiling, but the idea was dropped because their moving and adjustment for different settings would have been too cumbersome. Instead, the walls were equipped with fixed diffusers, formed by curved surfaces inside rectangular boxes. These diffusers reflect sound evenly across the room and also back to the performers. The building was completed in March 1997. It has been used for musicals, childrens’ plays, and experimental theatre. The acoustics has been considered very favorable.
Is it worth applying sound particle scattering methods to sound propagation in factory halls?—Results and comparisons of some simulations
Authors: Stephenson Uwe M.
Published: July 2002

Sound particle simulation, a version of ray tracing, is meanwhile a well-established method in room acoustics. In usual closed rooms, sound propagation is dominated by multiple reflections. In factory halls, the problem arises that a lot of inner sound scattering obstacles (machines, etc.) may practically not be registered in detail. A principal deficiency of ray tracing is the impossibility of taking diffraction into account. Often only a mean level decay per distance is of interest. Therefore, it is a well-founded and -tested semi-analytical approach of using only a mean scattering surface area per volume q=S/(4V) and combining that with a simple mirror image source method for rectangular rooms (German guideline VDI 3760). With the sound particle method, this multiple scattering is directly simulated in a Monte Carlo manner. This has several advantages: rooms of arbitrary form, diffuse reflecting surfaces, and inhomogeneous distributions of scattering bodies may be simulated. Meanwhile, on modern PCs, computation times lie in the range of only a few minutes. A disadvantage is the much higher effort of defining the (often uncertain!) input data. Some recent results of simulations of typical setups in factory halls (not computable by common methods) are presented and compared.
Modeling of sound fields in enclosed spaces with absorbent room surfaces
Authors: Dance Stephen M. , Shield Bridget M.
Published: July 2002

Commercial computer models are now generally available for the prediction of sound levels in industrial workshops. Four models have been used in this investigation; including raynoise v3.0, ramsete v1.3, rayscad+v3.3, and fame v1.0. These models were validated in rooms with highly absorbent room surfaces using standard and Millington-based absorption coefficients. The rooms included hypothetical, performance, experimental, and industrial spaces. Measurement data were available for the six central octave bands. The results are presented for sound levels and reverberation time, in addition to which the execution time for each model is detailed. It was found that Millington-based absorption coefficients provided consistently more accurate predictions for each type of space than standard absorption coefficients.
PlantNoise system for the prediction, visualization, and auralization of industrial noise during computer walk-through
Authors: Hodgson Murray , Heerema Nelson , Halingten Kurtis
Published: July 2002

PlantNoise, a novel hardware and software system for predicting, visualizing, and auralizing noise in typical industrial workrooms, is described and demonstrated. It is simple, rapid, accurate, and user-friendly, responding to the needs of industrial practitioners. Input data describing a workroom and its noise sources are input. The workroom floor-plan is visualized, along with the noise sources, a 1-m receiver grid, and a receiver icon. Octave-band and total noise levels, and octave-band reverberation times, are predicted at the receiver location, using new empirical models, and are displayed. The noise is also auralized, using a sound card. The receiver can be ‘‘walked-through’’ the workroom arbitrarily under user control. Visualized and auralized data are updated in real time. Noise contours can be plotted. At any time, the workroom can be modified—for example, to introduce noise-control measures—and the modified noise visualized and auralized.
Acoustic modeling and measurements of hearing protectors for impulsive noise
Authors: Gerges Samir N. Y. , Miranda Erasmo F. V.
Published: July 2002

REAL is the world-wide common method used for the hearing protector attenuation measurement in accordance with ISO 4869 and ANSI S12.6. This method is not suitable to determine the attenuation of the peak level of impulsive noise, because it measures the attenuation subjectively at low levels of a steady-state sound near the threshold of hearing. Alternative low cost and quick methods are required for assessing hearing protector attenuation for impulsive noise. In this paper a numerical model based on the finite-element method (FEM) for the impulsive noise attenuation prediction of the hearing protector is presented. The insertion loss of the peak level in the time domain is calculated and compared with laboratory measurements.
Calculation of aircraft sound propagation on airports via determination of ground impedance
Authors: Blumrich Reinhard , Altmann Jürgen
Published: July 2002

In the framework of acoustic and seismic monitoring of airports for verifying disarmament or peace-keeping agreements, the propagation of sound emitted by aircraft was investigated. This propagation is strongly affected by the porous ground. A determination of the ground surface impedance allows one to calculate the attenuation and phase shift with distance and height. In the course of aircraft signal measurements at the military airport at Jever, Germany, the surface impedance of the grass-covered ground was determined from c. 60 Hz to 2.2 kHz by measurements of complex transfer functions using (quasi-)white noise and a numerical inversion of the spherical wave reflection factor. A three-parameter impedance model was fitted to the measured impedance. By means of the ground impedance propagation calculations are carried out. In the case of the transfer function measurement signals, the calculated spectra agree very well with the actually measured ones. In the case of the jet aircraft, the simulated spectra fit to those of the measured signals only in general appearances of amplitude, and not in phase. This is probably due to an inaccurate localization of the sound source, its extension and directivity. Wind and temperature gradients were not considered.
Moving point source—A simple model for predicting statistical sound levels in the neighborhood
Authors: Schaffert Edelbert , Becker Stefan
Published: July 2002

When predicting the noise produced by moving sound sources (e.g., vehicles, airplanes, etc.) sound-level statistics can be helpful to describe the situation. However, the common sound models only yield the average sound level, and under special circumstances, also the maximum sound level. The calculation model moving sound source also gives sound level and the level-versus-time diagram. For the calculation, first the paths of the sound sources have to be digitalized and divided into parts. Each part will be represented as a point sound source which is active during a certain time slot, which again is based upon the velocities of the moving objects. Sound propagation from all point sources to the point of emission is calculated by using ISO or equivalent standards. This leads to the time versus level diagram and (when the steps from one to another point source are small enough) to the desired statistical levels. The described model, named the moving sound source, has no limitations. It could be easily integrated in commercial sound propagation calculation programs. It was successfully used for the approval of a new racetrack in Germany. Also, the new DIN 45684 (noise of small aircraft) is based on it.
Sound radiation from flow-excited, elastically supported rectangular panels
Authors: Mongeau L. , Fazio J. , Vlahopoulos N.
Published: July 2002

The objective of this study was to evaluate the influence of the viscoelastic properties of glass run weather seals on the interior noise radiated by vehicle window panel vibrations. The vibrational response of a flow-excited rectangular aluminum panel supported on all four edges by a fixture hosting a glass run rubber sealing system was measured in a low-speed wind tunnel. The panel was attached to an acoustically treated enclosure located underneath the wind tunnel test section. The resonance frequencies, damping coefficients, and mode shapes of the panel were measured following established modal analysis procedures. Two different sealing systems were investigated to study the influence of seal compliance and damping factor on the sound transmission loss of the elastically supported panel. The experimental results were compared to predictions obtained using finite-element and boundary element numerical methods. The sensitivity of the radiated sound level to variable seal properties was evaluated. The possible applications of these results to quiet vehicle seal design were considered.
Boundary-layer noise from high-speed vehicles: Theory and measurement
Authors: King W. F. , Barsikow B. , Klemenz M.
Published: July 2002

Wayside noise emitted by the turbulent boundary layer (TBL) on a moving vehicle represents its minimal acoustic radiation. When wayside noise levels for aerodynamically clean vehicles (particularly maglev ones) begin to approach within 3 dB of the TBL noise level, further abatement measures become ineffectual. Knowledge of this lower bound to wayside noise is important to avoid expenditures for useless attempts at further lowering sound levels. The acoustic energy of TBL noise per unit area is low, but its cumulative contribution on a long vehicle could be significant. The present paper comprises two parts. One is a semi-empirical theory of TBL noise. Although Lighthill’s acoustic analogy correctly describes TBL noise, this equation cannot be solved without knowledge of the (as yet) unknown source terms. Furthermore, the fluctuating density, the desired solution, appears on both sides of the equation. Hence, a model approach becomes necessary. The present offering is based upon extrapolated parameters describing the burst-sweep cycle within a flat-plate TBL. The second part of the paper summarizes results of array measurements made above the smooth roof of the maglev vehicle Transrapid 07 at speeds up to 400 km/h. Predictions show reasonable agreement with measurements.
Introductory remarks
Authors: Marsh Alan H. , Finke Hans-Otto
Published: July 2002

The new International Standard (IEC 61672, sound level meters) is being developed by Working Group 4 for IEC/Technical Committee 29, Electroacoustics. The standard will be published by the International Electrotechnical Commission (IEC). It amalgamates and extends the specifications and test procedures that were given in IEC 60651:1979 for conventional instruments that display sound levels with exponential time weighting and IEC 60804:1985 for integrating-averaging instruments that display time-average sound levels. A sound-level meter conforming to the specifications of IEC 61672 will provide the user with higher-quality measurements than previously available, especially for transient and short-duration sounds. This standard represents the culmination of years of effort by a dedicated team of experts from several countries. This special session will describe the acoustical and electrical characteristics that are required. All presenters of papers in this session are members of Working Group 4.
Summary
Authors: Marsh Alan H.
Published: July 2002

The proposed International Standard, IEC 61672, for sound-level meters, differs significantly from the existing standards for such instruments. Existing standards include IEC 60651:1979 for conventional instruments that measure time-weighted and frequency-weighted sound levels and IEC 60804:1985 for integrating-averaging and integrating sound-level meters that measure time-average sound levels and sound exposure levels, respectively. The principal differences and improvements will be highlighted. The anticipated schedule leading to eventual publication by IEC will be indicated.
Sound-speed determination in a fluid-filled elastic waveguide
Authors: McConnell James A. , Bastyr Kevin J. , Lauchle Gerald C.
Published: July 2002

Theory and test methods are presented for determining the longitudinal sound speed inside a fluid-filled elastic waveguide. The waveguide is a vertically oriented open-ended column of water equipped with a moving coil driver. The elastic properties of the waveguide reduce the sound speed to the point where a passive anechoic termination can be used effectively. This facilitates a compact device that exhibits plane progressive waves for sensor calibration at low frequencies. Once the sound speed is determined, then the size of the device and usable bandwidth are known. The sound speed is determined through measurement of the longitudinal resonance frequencies associated with the free vibration case of a duct having ideal rigid and pressure-release boundary conditions. In practice, however, the moving coil driver presents an impedance boundary condition (e.g., not ideally rigid) that inadvertently couples the mechanical system of the driver to the acoustic system of the waveguide. To circumvent this issue, frequency response measurements are made between a neutrally buoyant velocity sensor suspended in the fluid and an accelerometer mounted to the active face of the driver. In effect, the measurement decouples the two systems and provides the standing wave resonance frequencies of the duct having ideal boundary conditions.
Interaction of driven delay-coupled bubbles
Authors: Luther Stefan , Allen John S.
Published: July 2002

A rigorous derivation of the governing equation of two spherical, coupled, compressible bubbles indicates a variable time delay coupling term [Fujikawa and Takahira, Acustica 61, 188–199 (1986)]. However, the effects of this time delay were examined solely for free oscillations in that work. For acoustically driven bubbles, these time delay effects have been neglected in subsequent studies. This may be in part due to the computational complications involved in solving variable delay differential equations. In this study, delay effects are rigorously explored for acoustically forced bubbles. An equation for the overall energy balance of the system is derived and used to investigate novel Bjerknes Force effects. A direct comparison with nondelay formulations reveals important differences for certain parameter regimes. [Work supported by ‘‘Graduiertenkolleg Strömungsinstabilität und Turbulenz.’’]
Effects of shape oscillations on bubble dynamics and coalescence
Authors: Feng Zaichun
Published: July 2002

When a bubble is excited by an acoustic field, in addition to the volume oscillation, the bubble may deviate from the spherical shape and undergo shape oscillations. The shape oscillations can be directly caused by the anisotropic pressure or they can be caused by the energy transfer from the volume mode through nonlinear modal couplings. The shape oscillations influence the bubble dynamics in many ways. First of all, shape oscillations can limit the volume responses of the bubble since the onset of shape oscillations can lead to the development of localized large curvature and hence the breakup of the bubble. Second, the resonance coupling between the volume mode and a shape mode can give rise to the occurrence of chaos even when the acoustic forcings are so weak that the bubble oscillates close to its equilibrium radius. Third, the coupling between two or more shape oscillation modes can lead to translational motions of the bubble centroid; this can indirectly affect the bubble coalescence in an acoustic field. Finally, the shape oscillations may influence the coalescence through modifying the short-range attractive forces between two neighboring bubbles.
Propagation features of evanescent acoustic waves
Authors: Alippi Adriano , Bettucci Andrea , Farrelly Francis A. , Germano Massimo
Published: July 2002

Evanescent waves are not true propagating waves, so questions regarding their traversal time or propagation direction are ill posed. However, physical phenomena related to the time delay of pulsed evanescent perturbations, or refraction effects through a discontinuity, can be easily put into evidence in acoustics by proper choice of acoustic modes and geometrical configuration. In this paper, lateral displacement and time delay through forbidden barriers of Lamb mode waves with contradirected phase and group velocities are studied. Low noise detection achieved by cutting off spurious mode generation and a common objection made toward experiments of this type is avoided. This objection states that the highest frequency components of the pulse travel faster and are less attenuated than the main carrier frequency of the packet, distorting the original pulse to such an extent as to render measurements unrelated to barrier traversal times. Elastic waves generated with piezoelectric transducers are injected into aluminum plates through wedge-shaped Lucite waveguides so as to generate the appropriate Lamb mode which traverses a forbidden region and is picked up after it with a similar setup of transducer and waveguide. Various configurations of wave barriers have been measured and possible theoretical interpretation is given.
Binaural hearing by listeners with hearing impairments
Authors: Hawley Monica L. , Litovsky Ruth Y. , Colburn H. Steven
Published: July 2002

The ability of listeners with hearing impairments to understand speech in the presence of competing speech sources is often compromised; however, the extent to which poor binaural hearing contributes to this compromise is not known. These studies are aimed at relating listeners’ abilities to utilize binaural information in speech-based and non-speech-based tasks. In one set of measurements, speech intelligibility and localization of speech material were tested in the presence of one to three competing sources, placed at various spatial configurations. A second set of measurements tested the sensitivity to binaural information (i.e., MLD, ITD, and ILD discrimination). Listeners who obtained maximal binaural benefit in the speech intelligibility tasks generally showed good performance in all other tasks. Listeners who showed normal ITD/ILD discrimination could not always exploit that ability in more complicated tasks (MLD and speech-based tasks). Data were compared with the prediction of a model [Zurek, in Acoustical Factors Affecting Hearing Aid Performance (Allyn and Bacon, Boston, 1993)] which separates benefit of the ‘‘better monaural ear’’ and binaural interaction. Prediction was best when individuals MLDs were used to predict the binaural interaction term. Knowledge of the individual’s binaural abilities may help clinicians choose an appropriate rehabilitative strategy. [Work supported by NIH.]
The ‘‘cocktail-party effect’’ viewed through the lens of psychophysics
Authors: Divenyi Pierre L.
Published: July 2002

In an attempt to understand the psychophysical processes underlying the ‘‘cocktail-party effect’’—a phenomenon whereby one speech signal is extracted from the ensemble of other speech signals distributed in space—it has been reduced to the psychophysical task of segregating two concurrent signals X1(t) and X2(t) differing with respect to various attributes. These attributes represent three orthogonal ‘‘canonica’’ dimensions—spatial location, temporal (low-frequency envelope AM) structure, and pitch, although the pitch dimension is split into spectral and virtual pitch components. In any segregation experiment, the two signals differ along at least two attributes; segregation is assumed when each signal is associated with the correct values on each dimension. At segregation threshold, there is a trade-off over a certain range, where an increased difference along one dimension leads to a reduced difference along another. Along each dimension, there is also a ‘‘critical’’ difference at which the trade-off no longer exists. When converted to speech parameter coordinates, these critical differences—for azimuthal, spectral, fundamental frequency, and modulation frequency separation—can provide psychophysical predictions for conditions under which a speech signal could be successfully recovered in a real-life cocktail-party situation. [Work supported by the National Institute on Aging and the VA.]
Localization of distracted sound sources: Determining the role of binaural cues using unilaterally attenuated and interaurally uncorrelated signals
Authors: Braasch Jonas , Hartung Klaus
Published: July 2002

Six listeners were instructed to localize manipulated random noise (200-ms duration, 20-ms cos2-ramp, 12 directions in the horizontal plane) in an auditory virtual environment using individual head-related transfer functions (HRTF). In the first condition, the left headphone channel was attenuated by 20 dB. In the second condition, an interaurally uncorrelated noise was used as the test signal by filtering two uncorrelated noises with the left and right HRTF. A second random noise was used as a distractor (500-ms duration, 20-ms cos2-ramp, 0° azimuth). The distractor was manipulated in the same way as the test signal. Four different signal-to-noise ratios (S/N): 0, −5, −7, and −10 dB (plus an unmasked condition) were tested. In condition 1, the additional level difference of 20 dB caused a lateral displacement of the auditory event in four listeners. This displacement decreased when the distractor was added. In condition 2 (uncorrelated noise), the listeners were able to estimate the left-right direction, but frequently showed front-back confusions. With decreasing S/N the perceived directions clustered at −90°, 0°, and 90° azimuth.
Direction discrimination for formant-frequency glides and modulation
Authors: Lyzenga Johannes , Carlyon Robert P.
Published: July 2002

The vowel properties that determine their classification are the positions of peaks in their frequency spectra: the formants. For normal-hearing subjects, the sensitivity of detecting linear glides in the positions of these formants and the ability to discriminate between the two possible directions of these glides as functions of their sizes, were measured. A frequency rove was used to prevent subjects from using the start and end points of the glides as cues. The same quantities were measured for a 5-Hz formant-frequency modulation. A three-interval, three-alternative forced-choice paradigm was used in all measurements. The stimuli were single-formant ‘‘vowels’’ generated for a number of different stimulus parameters, including fundamental, spectral slope, frequency region, and position of the formant relative to the harmonic spectrum. For most stimuli, sensitivity was approximately equal for detection (of glides and modulation) on the one hand, and direction discrimination on the other. Exceptions were found for a few ‘‘formants’’ that consisted of a single harmonic that only showed an amplitude change over the stimulus length. The detection of formant-frequency modulation can be disrupted by off-frequency modulated formants, and new data on this phenomenon for modulation-direction discrimination will be reported.
Unexpected dependency of psychoacoustical sharpness on the bandwidth of noise
Authors: Haupt Ulrich , Weber Reinhard
Published: July 2002

Based on semantic differential and scaling experiments, von Bismarck [Acustica 30, 146–159 (1974)] proposed a model for psychoacoustical sharpness calculation, which later on was modified by Aures [Acustica 58, 282–290 (1985)]. Categorical sharpness evaluation of sounds exhibited systematic deviations from calculated sharpness after Aures under special conditions. Sharpness ratings of bandpass noises with the same calculated sharpness decreased if the bandwidth ΔB of the noises increased [E. Kabot and R. Weber, DAGA ’94, 1029–1032 (1994)]. The intent of the present study was to investigate this effect in detail. The stimuli to be assessed consisted of two sets of sounds containing tones and broadband noises (ΔB=0.5, 1, 2, and 5 Bark). All sounds in a set had equal calculated sharpness and were presented over headphones at the same loudness. Using a five-category scale, subjects evaluated each sound in two sessions, six times, in a random order. The results will be discussed against the background of new sharpness model calculations.
What can we learn from human processes of spoken language comprehension?
Authors: Fujisaki Hiroya
Published: July 2002

Conventional approaches for automatic speech recognition and understanding are essentially bottom-up. Namely, they are based on the assumption that speech can be decomposed reliably into the smallest units, can be recognized by identifying these units first, and then by clustering them to find larger units and their interrelationships. It is thus taken for granted that speech recognition is the prerequisite to speech understanding. The human processes of spoken language comprehension, on the contrary, are rather holistic and do not necessarily require identification of every constituent unit. This is possible since the listener has the capability of constructing the expected message while listening, and of matching it against the input message on the basis of global features, without paying much attention to the details. In the author’s view, the unit of human speech processing is considered to be not fixed, but variable in size depending on the context, situation, and knowledge. Likewise, the human processes of lexical access and parsing are considered to be quite different from those used in conventional systems for natural language processing. These issues will be discussed on the basis of studies by the author and his co-workers as well as by others.
Recognition of speech from temporal patterns
Authors: Sharma Sangita , Hermansky Hynek
Published: July 2002

Spectrum of short segments of speech signal is accepted as one of the prime acoustic correlates of phonetic quality and it forms the basis of most current feature representations of speech. However, the short-term spectral vectors are affected by coarticulation. The coarticulation can span time intervals of several phonemes. In spite of that, human speech perception is able to derive the identity of the individual underlying phoneme. This process does not seem to require successful decoding of neigboring phonemes [Allen, IEEE Trans. AU-2(4), 567–577 (1994)], but it may require information from a relatively large temporal span of the speech signal [Kozhevnikov and Chistovich, ‘‘Speech: Articulation and Perception,’’ Dept. of Commerce (1967)]. To allow for effective use of a syllable-level temporal information, a short-term spectrum-based vector is replaced as input to the classifier by rather long (about 1 s) vectors representing the time trajectory of critical-band spectral energy. Each temporal vector then spans many more than one phoneme. This technique forms the basis of a complete ASR system which is competitive with the current state-of-the-art speech recognition technology. [Work supported by the Department of Defense and by the National Science Foundation.]
Vocal microtremor assessed by complex demodulation
Authors: Gallasch Eugen
Published: July 2002

Vocal microtremor refers to small involuntary fluctuations of fundamental frequency (f0) and sound intensity during attempted constant phonation. Physically such an utterance may be considered as a FM/AM process x(t) with f0 as carrier frequency (after filtering the acoustic signal around f0 to eliminate its harmonics): x(t)=a(t)*cos(f0t+φ(t)) and f(t)=d[φ(t)]/dt. To study the instantaneous changes of f(t) and a(t) as well as their mutual relationships, complex demodulation was applied. First, the complex analytic signal y(t) was formed using the Hilbert operator H: y(t)=x(t)+jH[x(t)]=a(t)*ej[f0t+φ(t)]. Then demodulation was performed by multiplying y(t) with the negative complex carrier, thus shifting the narrow-band spectral content around f0 to zero: z(t)=y(t)*e−jf0t=a(t)*ejφ(t). At least the instantaneous quantities a(t) and f(t) could be recovered from z(t). The analysis was implemented with matlab functions including procedures to estimate modulation indices and the cross-correlation coefficient of the demodulated signals. Both waveforms showed periodic as well as irregular patterns and were positively correlated (about 0.5), indicating physiological or mechanical coupling in the sound generating mechanisms. This analysis might be further useful to study musical vibrato as well as laryngeal and neurological disorders.
Audition and the sense of presence in virtual environments
Authors: Gilkey R. H. , Simpson B. D. , Isabelle S. K. , Kordik A. J. , Weisenberger J. M.
Published: July 2002

Historical reports from suddenly deafened adults describing a sense of detachment from a world devoid of auditory input suggest that auditory cues may be crucial for achieving a sense of presence in virtual environments [R. H. Gilkey and J. M. Weisenberger, Presence 4, 357–363 (1995)]. However, factors other than acoustic fidelity can affect the perceived quality of a virtual auditory display. For example, Simpson et al. [J. Acoust. Soc. Am. 100, 2633(A) (1996)] showed that the sense of presence in virtual auditory environments was driven by characteristics of the real listening environment in which the virtual audio was heard. Specifically, when the virtual auditory environment matches the real listening environment, the listening experience is particularly compelling. However, it was not clear whether auditory or visual experience with the real room mediated this effect. This paper describes attempts to separate the impact of auditory and visual information on presence by systematically manipulating these cues in synthesized environments. Implications for design of virtual environments and spatialized auditory displays are considered. [Work supported by AFOSR and Ohio Board of Regents.]
The importance of binaural auralization for automotive industry
Authors: Genuit Klaus , Poggenburg Jens
Published: July 2002

In the automotive industry reliable information about the acoustical situation of a vehicle is necessary in an early stage of the development process. This enables the involved parties to work effectively on sound design and sound quality tasks. In this context, the so-called ‘‘Binaural Hybrid Model’’ has been developed. It is based on the determination of binaural transfer paths at vehicles using corresponding measurements. They are combined with airborne and structure-borne noise measurements at test rigs. Applied to vehicle interior noise caused by the engine, the influence of modified transfer paths, airborne radiation and/or input accelerations on the acoustical situation at the passenger’s position can be simulated binaurally. This includes the consideration of the head orientation and the aurally equivalent reproduction of the interior sound. It enables the engineer to judge the resulting sound quality before a prototype is realized. The presented model considers airborne shares up to 12 kHz and structure-borne ones up to 2 kHz. An application for different vehicles with the same engine clarifies in which way relevant causes for customer complaints at one vehicle may be determined. The variation of single transfer paths and the resulting interior noise will be included.
Measurements of acoustic scattering from partially and completely buried thin-shelled targets
Authors: Fox Warren L. J. , Maguer Alain , Tesei Alessandra , Schmidt Henrik
Published: July 2002

Conventional high-frequency imaging sonars become less effective in detecting and classifying bottom-laid targets when the targets become buried into seabottom sediments. Some recent work at the NATO SACLANT Undersea Research Centre has concentrated on investigating the use of lower-frequency sonars (2–16 kHz) in order to better exploit scattering features of buried targets that can aid in their detection and classification. Part of the recent GOATS’98 experiment performed near Elba Island, Italy, involved carefully controlled monostatic measurements of scattering by thin-shelled objects (spheres and cylinders) that were partially and completely buried in sand, as well as in free-field. Preliminary results are presented which show promising comparisons of the data to existing models. In addition to examining peak levels of the returns for detection purposes, special attention is paid to resonance effects (examined in the time and frequency domain), and the effect of burial on this part of the target response. Scholte–Stoneley and Lamb-type waves can be clearly identified in the backscattered signals. The issue of inverse scattering for classification purposes is also addressed. [Work supported in part by EC MAST Programme and ONR.]
Sea experimental detection of an array of hollow spheres
Authors: Carbó R. , Molero A.
Published: July 2002

A study of acoustic backscattering produced by a system of scatter arrays is reported. A simplified theoretical model has been developed to study the interference phenomenon due to the echo formation. This model establishes the relation between system geometry (target dimensions, scatter distance and depth) and sonar performances (directivity, pulse length, pulse repetition rate, wave length, and ship speed) getting a better system response. Target strength of a single scatter was defined at the laboratory, at an underwater tank with a dual-frequency side scan sonar working at 100 kHz. Different array geometries were placed in order to characterize minimum distance between scatters needed to discriminate them at the echogram. Experimental test has been realized at the sea with the same side scan sonar at 100- and 500-kHz frequency. Scatter arrays were placed at 15-m depth, were clearly located, and identified from a distance of 50 m. To identify the system two different arrays were placed: a short one with a reduced number of scatters, and a second one with a large amount of same scatters, with a total length of 48 m.
Sound diffraction by an underwater topographical ridge
Authors: Zhou Ji-Xun , Zhang Xue-Zhen , Chase Yun S.
Published: July 2002

Scale model experiments have been conducted to study the diffraction of sound by an underwater convex surface with a very soft boundary (closed-cell foam material). The diffraction boundary can be well described as a locally reacting surface based on measured acoustic parameters (very low speed and density, and very high attenuation). The diffraction acoustic pressure (insertion loss) behind the ridge and along the diffraction surface were measured. Measured insertion losses were compared with the theory of matched asymptotic expansion (MAE), developed by Fock for electromagnetic waves and Pierce for acoustic waves. The experimental results are in good agreement with predictions from the MAE theory. This technique might be used to infer acoustic parameters for materials. It was found that the model of Delany and Bazley, which is widely used in architecture acoustics and outdoor sound propagation, may be inappropriate for the description of an underwater locally reacting impedance. [Work supported by ONR.]
A linear constrained design method for minimizing the interference effect in linear and planar array
Authors: Song Joon-il , Sung Koeng-Mo , Lim Jun-Seok
Published: July 2002

In beamformer design, acoustic interaction effects experienced among the elements of an array will degrade its electroacoustic performance and change the beam patterns. An architectural modification has already been proposed as an effective measure for reducing this effect. Further reduction for interaction effects can be achieved by optimization techniques. This paper provides a new design method including the above constraints for linear or planar array. It is based on the solution of the linearly constrained least-squares problem and can be used in the arrays having arbitrary element locations. Results of computer simulations illustrating the performance of the beamformer are presented. [Work supported by UARC.]
New sonar technologies using lower sound pressure levels
Authors: Wicker Kai , Arens Egidius
Published: July 2002

Depending on the conditions, modern active sonar equipment is capable of detecting targets up to distances of about 100 km. To reach this goal, high sound-pressure levels up to 220 dB or more are usually necessary. Simultaneously, this sound-pressure level dramatically increases the probability of being detected, which represents a tactical disadvantage. A second problem arose in the last few years. Sometimes it is assumed that there could be some influences on the marine environment due to the high sound intensities. A new active sonar method and its signal processing is proposed. It could also be applied for acoustical oceanography applications, e.g., seafloor classification. By maintaining very wide frequency ranges during very long time periods, the high sound-pressure level can be reduced drastically. This results in a low probability of being intercepted and also in a reduced influence on the environment. For a realization of this method, four main topics have to be investigated: (1) improved broadband transducers, (2) special designed signal types, (3) a new signal processing concept, and (4) detailed calculations of sound propagation taking into account the local sea and seafloor properties.
Broadband geo-acoustic inversion in shallow water
Authors: Hermand Jean-Pierre
Published: July 2002

The paper discusses an inversion method that allows the rapid determination of in situ geo-acoustic properties of the ocean bottom without resorting to large acoustic receiving apertures, synthetic or real. The method is based on broadband, waterborne measurements and modeling of the waveguide impulse response between a controlled source and a single hydrophone. Results from 1994–1997 experiments in Mediterranean shallow waters with single elements of vertical array and advanced drifting acoustic buoys are reviewed. Inversion of the bottom parameters is performed with an objective function that includes the processing gain of a model-based matched filter (MBMF) receiver relative to the conventional matched filter. The MBMF reference signals incorporate waveguide Green’s functions for the known geometry and water column acoustic model, and hypothesized bottom geo-acoustic models. The inversion results demonstrated that—even for the fairly complex environmental conditions of the experiments—a single transmission of a broadband coded signal received at a single depth and a few hundred of forward modeling runs were sufficient to resolve the bottom features. These included the sound-speed profile, attenuation, density, and thickness of the top sediment layer, and sound speed and attenuation of the bottom. Exhaustive parameter search proved unequivocally the low-ambiguity and high-resolution properties of the MBMF-derived objective. The single-hydrophone bottom inverses closely agree with the ones obtained under the exact same conditions from matched-field processing of multitone pressure fields sampled on a vertical array. Both of these results agree with expectations from geophysical ground thrust.
Model of acoustic monitoring of heat and mass transport in the Fram Strait
Authors: Naugolnykh Konstantin A. , Johannessen Ola M. , Esipov Igor B. , Ovchinnikov Oleg B. , Tuzhilkin Yuri I.
Published: July 2002

A computation model of acoustical monitoring of heat and mass transport in the Fram Strait is discussed. The method relies on the advections of small-scale inhomogeneities across the acoustic path and travel-time variations in the process of signal crossing of the Strait on a number of paths to infer the intervening of fine-scale variability and transverse current. Inhomogeneities of the current produce perturbations in the travel time of the sound, and the current can be sensed by generating a time-lagged cross correlation of the full acoustic field. Both the linear four-element transmission array and four-element receiving array with equally spaced elements were used for calculations. By combining the signals from each transmitter–receiver pair in different ways, a number of different path positions were probed and a profile of transverse current U(r) along the propagation path r was retrieved, whereas the value of the travel-time variation is related with T—temperature change. The possibility of estimation of heat Q∼∫U(r)T(r)dr and mass M∼∫U(r)dr transport in the cross section is regarded. [Work supported by Environmental and Climate Programme of EC, Project PL970338.]
Application of the Tonpilz design for optimizing parameters of rod transducers
Authors: Aronov Boris
Published: July 2002

The optimum design of a rod transducer under various loading conditions and operating requirements is examined. It is often possible that the operational requirements can be met by using only a part of the rod volume with the electromechanical or piezoelectric element. The remaining sections of the rod can be replaced by passive materials or caps (i.e., a ‘‘head’’ and a ‘‘tail’’) while still maintaining the same input impedances and resonant frequency. Such a composite rod transducer made of active and passive materials having different length and cross-sectional areas is of the ‘‘Tonpilz’’ type. The proper choice of dimensions and materials for the caps will result in optimum transformation of mechanical (acoustical) loads and transducer performance. Both criteria of optimization, the rod transducer parameters and the procedure for determining properties of the caps that provides optimal loads, are discussed.
The development of a doubly curved acoustic array using injection molded 1-3 piezocomposite
Authors: Benjamin Kim C. , Oliver Stephen F. , Arteiro Jose C. , Petrucci Russell S. , Serwatka William J.
Published: July 2002

The design, fabrication, and acoustic calibration of a doubly curved acoustic array for UUV oceanographic applications is presented. The sensor employs a unique piezoceramic thermoplastic composite which is conformable and capable of deep ocean operation. The composite consists of several small diameter (1.0 mm) PZT-5H rods regularly distributed in a two-dimensional lattice and aligned perpendicular to their diameter. The PZT-5H rods are encapsulated within a thermoplastic viscoelastic moldable epoxy matrix to form an active acoustic composite panel. The thermoplastic epoxy phase of the composite allows the panel to be shaped to a specific vehicle geometry. Low-profile (<1.0 mm) coaxial cable embedded within the composite panel provides addressing between the acoustic array elements and the processing electronics. [This work was supported by Code 80 of the Naval Undersea Warfare Center, Division Newport.]
The influence of inharmonicity on piano tuning
Authors: Mori Taro , Bork Ingolf
Published: July 2002

It is well known that octaves of piano sounds are stretched to more than a 2 : 1 ratio. One reason is that partials of piano tones are not exactly harmonic because of the flexural rigidity of the string. This inharmonicity is characterized by the coefficient B, which depends on the tension, length, Young’s modulus, the area of cross section, and the radius of gyration of the string. Piano technicians are trained to tune the octave as ‘‘beautifully’’ as possible, but the inharmonicity of the string hinders perfect tuning without beat. For example, when the second partial of the lower sound and the first partial of the upper sound are tuned without beat, all other partial pairs show beats. On the other hand, since the timbre structure of partials is unique for each instrument, the tuned octaves of two different instruments may differ. Some piano sounds were simulated by additive synthesis with different octave, inharmonicity, and timbre in order to perform subjective hearing tests with some piano tuners. It could be shown that the tuned octave width depends not only on the coefficient B but also on the timbre. The influence of the sound board is also discussed.
Auditory processing of coherent and incoherent frequency modulation
Authors: Carlyon Robert P.
Published: July 2002

When two elements of a visual array move coherently, there is a strong tendency to group them perceptually. This has prompted several authors to suggest that acoustic frequency components which undergo coherent FM should be perceptually fused, and that those which are modulated incoherently should be segregated. This paper will review the mounting body of evidence demonstrating that, once a number of co-varying cues are controlled for, the auditory system is insensitive to the difference between coherent and incoherent FM. Such cues include the greater inharmonicity, smaller swings in average frequency, and more salient distortion products which can occur with incoherent compared to coherent modulation. In addition, new evidence will be presented showing that listeners can be more sensitive to one form of incoherent FM than to another: Opposite-direction quasi-linear frequency sweeps imposed on two carriers are more detectable when the frequency of the lower component decreases and that of the higher component increases than vice versa. This occurs even though the frequency sweeps imposed on the individual components are equally detectable when those components are presented in isolation, and even though within-channel interactions are masked by noise.