bluuebunny/crossref_metadata_2025 huggingface dataset:
Tracking results from the three-dimensional full field processing (FFP) array and the Santa Barbara Channel experiment
Authors: Freese Herbert , Lee Yung , Greene Robert , Mikhalevsky Peter
Published: July 2002

In March of 1998, SAIC under DARPA sponsorship deployed five 170-m, 30-element, vertical line arrays (VLAs) in 200 m of water in the Santa Barbara Channel. The VLAs were arranged in a pentagon with a 200-m horizontal aperture. Each hydrophone package contained tilt sensors for array localization as well as a thermistor for water temperature measurement. Each VLA also contained four compasses that were used in conjunction with the tilt sensors for passive localization of the array elements. In this presentation the acoustic experiments will be described and a number of results including a comparison of acoustic and nonacoustic localization methods, an environmental characterization, and the results of acoustic measurements using a towed source and surface ships of opportunity will be presented. Specific acoustic results to be presented will include measurements of array signal and noise gain, and the results of tracking the towed source and surface ships using adaptive matched field methods.
Highly resolved detection in a waveguide with the D.O.R.T. method
Authors: Prada Claire , Mordant Nicolas , Fink Mathias
Published: July 2002

The D.O.R.T. method (French acronym for decomposition of the time-reversal operator) is a scattering analysis technique that uses an array of transducers in pulse-echo mode. The method was applied to detection and selective focusing on pointlike scatterers through inhomogeneous media [J. Acoust. Soc. Am. 99, 2067–2076 (1996)]. Here, laboratory measurements in a water waveguide are presented. Taking advantage of the multiple reflections at the interfaces of the guide, high resolution is achieved with the D.O.R.T. method without using any a priori knowledge of the guide. The separation of two scatterers and the selective focusing are obtained with a transverse resolution at least nine times better than the predicted free space limit. The detection of a scatterer placed at less than half a wavelength from the water/air interface is also achieved with good sensitivity. Finally, the impulse response functions of each scatterer to the array is computed as a combination of the eigenvectors of the time-reversal operator obtained at each frequency. Using these impulse Green functions, selective focusing with high temporal and spatial compression is performed.
Audiogram, morphological contraints, environmental acoustics, and adaptation
Authors: Plassmann Wolfgang , Glaser Evelyn
Published: July 2002

Phylogenetic contraints are a severe restriction on the morphological diversity of the peripheral auditory system in mammals. In order to investigate problems of sensory adaptation under selective pressure, one has first of all to show in a model analysis the potential variance of the system and physiological consequences resulting from these variances. On the basis of these theoretical considerations, morphological investigations, and physiological measurements, an attempted explanation of given auditory systems can be offered based on additional field observations. Following this line of argument, the subsequent topics will be adressed: (a) size and allometric relations of middle and inner ear structures will be presented; (b) consequences of morphological size and changes of allometric relation for the audiogram will be discussed by means of theoretical models and experimental studies; (c) sound propagation properties of different environments such as desert, grassy plain, and forest will be demonstrated. Based on these data, possible selective pressure on the peripheral auditory system in various habitats will be discussed in relation to audiograms, vocalization, and abiotic sound.
Adaptations for hearing in background noise
Authors: Klump Georg M. , Langemann Ulrike , Hamann Ingo , Nieder Andreas
Published: July 2002

Acoustic communication in the natural environment is greatly affected by the ubiquitous background noise. Therefore, vertebrates have evolved mechanisms to efficiently cope with auditory masking. Spectral filtering of the acoustic input is one possibility to improve signal detection. In addition, communication efficiency can be enhanced by exploiting the amplitude fluctuations in background noise. This improvement of signal detection has been termed ‘‘comodulation masking release’’ (CMR). CMR has two components: (1) release from masking within the spectral filter (i.e., a critical band) centered on the signal and (2) release from masking by comparison of correlated amplitude fluctuations (i.e., comodulation) across separate spectral filters. An excellent animal model for studying CMR is the European starling, a songbird exhibiting a temporal resolution and frequency selectivity in its auditory system that is similar to the human’s processing performance. Psychoacoustic experiments in starlings show a considerable CMR both within and across spectral filters. Furthermore, the range of envelope frequencies is an important determinant of the amount of CMR. Neurophysiological data recorded in the starling’s auditory forebrain reveal that the temporal encoding of the stimulus envelope within auditory filters is a major factor determining the amount of CMR. [Work supported by the DFG, FG Hoerobjekte.]
Echogram noise quantification with application to herring observations
Authors: Korneliussen Rolf J.
Published: July 2002

Pushing the limits of scientific echo sounders involves considerations of noise, which is inherently frequency dependent. Suprisingly, perhaps, there is also a dependence on bottom depth. In this work, noise is quantified by measurement for a standard echo sounder, the EK500, at 18, 38, 120, and 200 kHz. Use of empirical relations of noise as a function of range to reduce echogram noise is described in general, and illustrated in particular for data collected on Norwegian spring-spawning herring (Clupea harengus) when wintering in the Vestfjord system. [Work supported by the Norwegian Research Council through Grant No. 113517/120.]
Preliminary description of swimming activity and estimation of swimming speed of saithe (Pollachius virens) at one location in the North Sea
Authors: Pedersen Jens
Published: July 2002

Individual saithe were tracked with a split-beam echosounder, while the vessel was drifting, in the area around Eigersundbank in the North Sea and their swimming speed estimated. The average swimming speed was approximately 4 and 1 body lengths per second for small saithe (20–30 cm) and saithe >70 cm, respectively, and a significant inverse relationship between length of the saithe and swimming speed was found. There was clear evidence of diurnal variation in swimming speed of small saithe, as the swimming speed was significantly higher during night (18–06 h) than during day (06–18 h). The number of observations on saithe >70 cm was too small to compare day and night swimming speeds. Although the results indicate higher swimming speeds of saithe in the demersal layer compared to pelagic saithe, significant differences were not found. The duration of acoustic observation time was 36–56 min per 4-h sampling interval during the 24-h cycle. A total of 278 series of saithe were selected, which, according to selection criteria, were accepted as representing tracking of single fish over two pings or more. The species identify of the targets tracked acoustically was verified by trawling in the layers investigated.
Acoustics education at the Stuttgart University
Authors: Hübner Gerhard
Published: July 2002

The Stuttgart University is relatively young compared to several other German universities. Nevertheless or fortunately for this reason the developments in Stuttgart’s acoustical teaching went off with some remarkable distinctions in its relevant scientific history and present situation. (i) A new teaching focused on noise generation and radiation caused by practical sound sources was started successfully by K. Gösele already in the late 1940s. (ii) Supported and initiated by the requirements of the important industrial area around Stuttgart, an increasing interest of more and more Engineering Departments to include noise and noise control in lectures and investigations took its origin much earlier than in other places which, in 1993, concluded in the establishment of the ‘‘Work Group of Technical Acoustics and Noise Control at the Stuttgart University’’ (ATALUS) where 13 institutes from 6 different faculties organized their exchange of experiences on a broad basis. Furthermore, in 1994 several acoustical lectures are included in the new curriculum ‘‘environmental engineering and protection.’’ (iii) Several relevant basic lectures are straightened on the general theory, e.g., focused on numerical methods (BEM, the link of DFEM with the physical phenomenon of sound radiation), propagation and sound power of structure-borne sound in general shaped bars and shells using second-order differential equations, and sound intensity measurement techniques.
Standard and nonstandard Galerkin FEM for numerical acoustics
Authors: Ihlenburg Frank
Published: July 2002

Over the last decade, several modern nonstandard modifications of the standard Galerkin finite-element method have been developed for, or applied to, the numerical solution of the Helmholtz equation. Most of these concepts, like Galerkin least-squares or bubble methods, have been first developed and demonstrated in one dimension, where also a rigorous analysis of the so-called preasymptotic error behavior of the standard Galerkin FEM is available. In this talk, a short overview of these topics will be given and the extension of the methods/results to higher dimensions will be discussed, where practical problems are formulated.
Normal modes of vibration in a violin
Authors: Hansen Uwe J. , Rossing Thomas D.
Published: July 2002

The normal modes of vibration of a violin constructed by Carleen Hutchins have been studied using both sinusoidal and impact excitation while the instrument is supported in several different ways. The results from these experiments are compared to those of several previous investigators on the same instrument [see references in Roberts and Rossing, Catgut Acoust. Soc. J. 3(5), 3–9 (1998)]. The observed operating deflection shapes, from which normal modes are determined, are found to vary with means of support and excitation, but by comparing the various data the normal modes can be determined with reasonable certainty.
Nonoccupational noise and hearing capability of children and young adults
Authors: Schulz Detlef , Künzel Karin , Hentschel Lars
Published: July 2002

There has been considerable literature over the past decade ascertaining a worsening of the hearing capability of young persons. It is supposed that this may be caused by some peculiarities of the modern way of living, that is, man-made noise (e.g., noisy hobbies). To get insight in some components of the real noise exposure of young people dosimetric measurements were performed in schools as well as in discotheques and as spectators of sports events (football matches and some indoor sports like basketball). For discotheques equivalent sound pressure levels of 95–105 dB(A) are typical; for indoor sports they are in many cases on the same order of magnitude. Persons frequently attending such events may get an exposure which is comparable with occupational noise exposure above the daily average of 85 dB. Furthermore, audiometric testings of children and young adults at the age of 14–25 years were performed. The high-frequency range (8–16 kHz) was taken into special consideration, since it is assumed to be important for the early detection of noise-induced hearing threshold shifts. Hearing thresholds together with the results of questionnaires are presented showing an increasing threshold shift with age already at young people, especially in the frequency region above 10 kHz.
Characteristics of the Aeolian tone generated from cylinders in conjunction with pantograph noise control
Authors: Fujita Hajime , Sagawa Akio
Published: July 2002

Aerodynamic sound generated from cylindrical objects, such as in pantographs of trains, is a prime noise source in high-speed vehicles. Characteristics of the aerodynamic sound generated from two-dimensional cylinders are studied experimentally in order to contribute to the prediction and control of high-speed vehicle noise. The effect of the angle of inclination to the mean flow of a circular cylinder, and the effect of the angle of attack of a square cylinder, to the Aeolian tone generation are studied quantitatively, and angle ranges for low noise generation are found in the subcritical Reynolds number range. The relative importance of the cylinder surface pressure fluctuation and its spanwise correlation length to the Aeolian tone generation is revealed. Experiments are extended to the super-critical Reynolds number range for a circular cylinder normal to the mean flow. In the beginning of the surface boundary layer transition, the Strouhal number suddenly jumps up to about twice as much and the Aeolian tone level decreases suddenly about 20 dB. It is found that the decrease of the Aeolian tone level is mainly due to the decrease of the cylinder surface pressure fluctuation. Application to recent pantograph development is discussed.
On the noise from turbomachinery and ducted fans—A review of prediction methods
Authors: Roger Michel
Published: July 2002

Noise reduction of turbomachines and ducted fans is a major issue in aeronautics, with regard to both environment of airports and passengers. From the theoretical point of view, it is the final result of a whole strategy, involving physical understanding as a starting point and modeling as a tool for searching low-noise solutions. This is true for generating as well as propagating mechanisms. Within that scope, a complete simulation of the whole acoustic process from source to observer is not necessary. A partial model easy to run and able to predict the right trends is enough. The paper is devoted to a state-of-the-art in prediction methods. A presentation is given of both analytical and numerical techniques understood as complementary tools. Limitations of the former and expected future developments of the latter are discussed, together with hybrid techniques coupling computational fluid dynamics for the definition of sources and analytical acoustics. Examples are taken in the literature, not only in connection to turbofan engines, but also to other rotating blade technologies involving rotors and stators. The specificities of each application are used as a guideline for the best-suited approach.
Scattering at localized absorbers in a lined duct wall
Authors: Mechel Fridolin P.
Published: July 2002

Homogeneous absorbing duct linings usually are tuned for medium and high-frequency attenuation. If they were tuned to low frequencies also, their thickness would become large. A possibility to add low-frequency attenuation to a broadband attenuation at higher frequencies consists in the combination of localized Helmholtz resonators with a broadband lining. The scattering cross section of the resonator orifice is large enough to permit some axial spacing between resonators. A method will be presented to show how the scattering at localized resonator orifices in a lined duct wall can be computed. The sound field in the duct is synthesized with silencer modes. The resonator orifice is introduced into the modal analysis as an effective volume flow source. Its strength is evaluated from the boundary conditions. Numerical examples show the wanted low-frequency attenuation maxima by arrangements with only a little increase of the silencer length and lining thickness.
Acoustic pulse reflectometry for the measurement of tubular structures
Authors: MacGillivray Thomas J. , Greated Clive A.
Published: July 2002

Acoustic pulse reflectometry is used in the reconstruction of bore profile and the calculation of input impedance for tubular objects with varying cross section. The technique, previously employed in the measurement of musical wind instruments, enables differences of the order 0.1 mm in internal radii to be detected. Experimental measurements made on small-scale tubular objects of varying complexity demonstrates the effectiveness of reflectometry for profiling such structures. Certain configurations for which the accuracy of the technique begins to break down are also apparent. Examination of the results allows the effective length and area of an object to be determined from the reconstructed profile. Input impedance curves evaluated for small radius stepped and single-branched tubes exhibit differences compared to the corresponding theoretical expressions. This requires consideration of the object dimensions and also of viscous and thermal effects. A procedure for detecting blockages within the structures is developed through examination of different object profiles. The reconstructed profile for a structure containing an obstruction appears altered in shape compared to the unblocked case. The size of the blockage and the axial position are not, however, distinguishable.
A new acoustic conductor
Authors: Shenoda Fathy , Shenoda Albeer
Published: July 2002

A new acoustic conductor was designed and experimentally studied. This conductor consists of a perfectly rigid pipe of constant cross-section area and is treated at its side wall with a small gradually perforated area. This experimental study aimed to determine its behavior and to outline the design procedures for optimizing its performance. The parameters (reflection coefficient r, normalized acoustic resistance R, and reactance X, at the conductor inlet) characterize the matching performance of the conductor, and were measured for different configurations and at different conditions. The design parameters such as the length, the graduality factor with which the side-perforated area increases, the function with which the side-perforated area is changed and reaches a maximum width Dm at the end of the conductor, as well as the conductor cross-diameter, were separately studied in detail at different conditions. The conductor design is optimized at the conditions: The conductor terminates at a point where almost all the sound energy has been released. Within the effective frequency range, r, R, and X have the following values: r=0.1, R=1, and X=0. The extensive experimental study and the great number of measurements at different conditions lead to an empirical formula for the calculation of cut-off frequency fc and optimum Dm.
Acoustic radiation forces on bubbles in real fluids
Authors: Doinikov Alexander A.
Published: July 2002

Early theoretical studies on the acoustic radiation forces on gas bubbles have treated the surrounding medium as an ideal fluid devoid of viscosity and heat conduction, and for bubble–bubble interactions in addition with no compressibility. However, real fluids have all the above-mentioned properties and this fact may be very essential to the radiation forces. A more rigorous theory, allowing for the nonideality of the surrounding fluid, predicts a host of new exciting effects, such as the change of the sign of the force in a plane-traveling wave at high dissipation, long-range terms in the interbubble force which can cause the two bubbles to form a bound pair with a stable separation distance, etc. The purpose of this paper is to discuss such predictions.
Time-domain Floquet-wave Green’s function for an infinite sequentially excited periodic line array of point radiators
Authors: Felsen Leopold B. , Capolino Filippo
Published: July 2002

A variety of new synthetic materials, especially those exploiting sonic or photonic bandgaps, are based on a background with strict or perturbed periodicity, which may be subjected to rapidly switched external modulation. This has motivated the present prototype study of the time domain Green’s function for an infinite sequentially excited periodic line array of point radiators or scatterers. Such Green’s functions, which parameterize TD Floquet wave (FW) dispersion, are the analytic building blocks for finite planar arrays with physical radiators. Various techniques are employed to develop exact expressions for the dispersive TDFW’s synthesized by Poisson summation over the linearly time-delayed nondispersive impulsive point source radiators. These techniques include Fourier inversion of frequency domain FW’s, TD spectral wavenumber decomposition and recombination, and use of complex space-time analytic signals. Each formulation furnishes different insights into propagating as well as evanescent TDFW spectra, and into TDFW dispersion. Some basic questions concerning causality and TDFW ‘‘physical observability’’ are posed and answered.
Micellar structures in layered oil/water surface systems studied with photoacoustics spectroscopy
Authors: Pogorzelski Stanislaw J. , Szurkowski Janusz , Sliwinski Antoni
Published: July 2002

The model layered system consisting of thin olive oil layers (ranging from 25 to 250 μm in thickness) spread on the water surface was studied using a photoacoustic method. The classical Rosenzweig–Gersho model assuming a one-dimensional heat flow in the sample was adopted as a depth profiling approach to interpret data in terms of the thermal diffusion length and thermal diffusivity. Distinctive changes in the PA signal phase and its amplitude discovered at the air/oil and oil/water interfaces, as well as at the depth of 9−13 μm beneath the oil surface, point out the possibility of formation of organized adsorbed interfacial layers and micellar structures in a bulk oil phase of surfactants, if present in an oil medium at a certain concentration (above critical micellar concentration CMC) and temperature (Kraft point). Such organized structures have different thermodynamic parameters like specific heat or thermal conductivity that affect to a great extent the photoacoustic effect used in the depth profiling sensing.
Localization of distracted speech and noise in reverberant and anechoic environments
Authors: Hartung Klaus , Braasch Jonas
Published: July 2002

Eleven listeners were instructed to localize a test signal in the presence of a distractor sound in an anechoic environment and in a reverberant environment. The experiment was conducted in an auditory virtual environment using individually measured head-related transfer functions. A rectangular room (6×5×3 m) was created using the mirror-image method to simulate the reverberant environment. The distractor (noise burst, 500-ms duration, 20-ms cos2-ramp) was placed at 0, 30, or 90° azimuth. The signals, random noise bursts (200-ms duration, 20-ms cos2-ramp) or the German spoken word ‘‘zuende’’ were presented from 13 directions in the frontal hemisphere of the horizontal plane at different signal-to-noise ratios (S/N). For some S/N, the listeners were unable to discriminate between a number of directions of incidence of the test signal (median test). The distinguishability of locations (DOL) was introduced to measure the percentage of discriminated locations. The DOL is a function of the S/N, decreasing rapidly below a threshold of −12 dB in an anechoic environment. This threshold increased by about 5 dB under reverberant conditions. The reduced DOL was confirmed by a 2A-4IFC experiment. If a distractor were present in the anechoic environment, the perceived directions of the test signal were shifted significantly in the direction opposite to the distractor location for all distractor directions (sign test, α=3D0.01). [Work supported by DFG.]
Application of localization models to noise suppression in hearing aids
Authors: Hohmann Volker , Albani Stephan , Nix Johannes
Published: July 2002

One major problem of hearing-impaired people is the cocktail-party effect, i.e., the reduced ability to understand speech in noisy environments. Normal-hearing listeners use binaural information, i.e., interaural level and phase/time differences, in addition to monaural cues to localize sound sources in the environment. Based on the localization result and on the monaural cues, the listener performs a significant noise reduction by concentrating on one of the sources. Noise reduction algorithms have been proposed that try to imitate the binaural information processing in the hearing system by using two microphones at both ears and a central processor. This setup allows for the use of interaural parameters to estimate the direction of sound sources and to attenuate unwanted sound source directions. However, in noisy environments the binaural parameters become noisy as well, reducing the accuracy of the sound source estimation and therefore reducing the maximum performance of the noise reduction algorithm. In this presentation, a localization algorithm is introduced, which uses a statistical description of the binaural parameters in order to perform a robust estimation of sound source directions in noisy environments. Noise reduction schemes based on this localization algorithm are discussed.
Localization ability in the horizontal plane in adverse auditory environments
Authors: Vause Nancy L. , Grantham D. Wesley
Published: July 2002

The purpose of this investigation was to examine the effects of adverse conditions on localization ability in the horizontal plane. The experiment was designed specifically to assess horizontal plane localization accuracy for a target signal presented in the adverse conditions of noise, headgear, and reverberation for two orientations. Eight normal hearing subjects participated in a source identification task in nine different experimental conditions of headgear, environment, and background noise for the frontal and lateral orientation. The results showed that (a) the effects of noise degraded localization performance in both the frontal and lateral orientations to a greater extent than reverberation or the combination of reverberation + noise; (b) subject performance was not significantly disrupted by reverberation in a quiet environment, but did slightly decrease when noise and reverberation were combined; however, performance in the lateral orientation improved in both quiet and noise conditions; and (c) localization accuracy declined when subjects wore protective headgear in all conditions for both orientations. [Work supported by NICDC and U.S. Army.]
Frequency discrimination following linear frequency glide
Authors: Kayahara Takuro , Sato Takao
Published: July 2002

Difference limen for frequency (DLF) following linear frequency glide was measured using 2IFC procedure to investigate mechanisms underlying detection of frequency change in successive tones. In the experiment, a pair of 100-ms tone, one of which had fixed frequency of 1000 Hz, was presented. The interval between the two pulses was 200 ms and the order of pulses in each trial was randomized. An adapting tone preceded each pair by 200 ms and was either 200-ms linear frequency glide with frequency extent of 100 Hz (950–1050 Hz) or 200-ms stationary tone of 1000 Hz. In the result, when the glide preceded the pair of pulses, DLFs at 1000 Hz were significantly elevated irrespective of direction of glide compared to DLFs with no preceding tones. The stationary sine wave, however, had no effect on DLFs as several previous studies have reported. In sum, unidirectional continuous change of frequency disrupted a process of sampling and comparing frequencies from successive tones. These results indicate that the process of detecting continuous and successive change of frequency are at least partially mediated by a common mechanism.
Simulations of spectral masking and the asymmetry of masking
Authors: Verhey Jesko L. , Dau Torsten
Published: July 2002

Measurements and simulations of (i) notched-noise experiments and (ii) masking experiments where the signal and the masker have the same center frequency but different bandwidths, are presented. The simulations were performed with a model originally developed to describe effects of temporal masking and modulation detection [Dau et al., J. Acoust. Soc. Am. 102, 2892–2905 (1997)]. The model accounts well for notched-noise data which are mainly a consequence of the gammatone-filter parameters used in the peripheral filter stage of the model. The model also accounts for effects in notched-noise experiments with an asymmetric notch (‘‘off-frequency listening’’), which results from the decision stage that integrates signal information across frequency. A simple modulation-lowpass filter (energy integrator) would be sufficient to describe these experimental results. In contrast, the processing by a modulation filterbank is necessary to account for the conditions of ‘‘asymmetry of masking’’ in the second experiment.
Near-field control of complex sources
Authors: Tichy Jiri
Published: July 2002

The near-field of complex sources that are characterized by standing waves on their surface is modeled by baffled point sources. The near field is visualized by showing the spatial distributions of sound pressure, sound intensity, and energy density. The field maps provide essential information on the local energy flows. Several sound control schemes to minimize the total radiated sound power by placing small sources in the vicinity of the primary sources are analyzed and tested.
Experiments on the active increase of sound transmission loss using distributed actuators
Authors: Henrioulle Kris , Sas Paul
Published: July 2002

A series of experiments have been performed to improve actively the sound transmission loss through a single-panel partition in the low-frequency range. Two types of flat acoustic actuators, designed in the framework of the European project DAFNOR (distributed active foils for noise reduction), are tested. The first acoustic actuator is based on PVDF piezoelectric material, which induces a bending motion in the actuator. The second acoustic actuator is based on the electrostatic principle. The experimental setup consists of a single steel panel of size 300×400×1 mm which is subject to a plane-wave acoustic excitation. The control system consists of the distributed acoustic actuator, which is placed near the steel panel. Various control strategies, which were previously simulated, are tested. The control actuator is driven by a controller which aims at reducing the acoustic energy at the radiating side. [The work reported herein was related to the EC Brite/Euram Research Project ‘‘DAFNOR’’ (under contract BRPR-CT96-0154). The project is supported by the Directorate-General for Science, Research and Development of the CEC.]
Estimating the nonacoustic fluctuating velocity amplitude from Cranen and Boves’ vowel production data
Authors: Barney Anna
Published: July 2002

In a dynamic mechanical model of the larynx and vocal tract [Barney et al., J. Acoust. Soc. Am. (in press)], the fluctuating velocity field was shown to have contributions from an acoustic disturbance and also from nonacoustic flow associated with vortex transport along the model tract. Nonacoustic velocity fluctuations, interacting with changes in the tract area function, can constitute significant sources of sound; in the vocal tract these could occur at the lips, teeth, etc. However, such additional sources of sound are not generally included in models of voiced speech production. Cranen and Boves [J. Acoust. Soc. Am. 77, 1543–1551 (1985)] obtained in-tract and radiated sound-pressure time histories from phonating subjects, from which they computed in-tract velocities. In this study the relative amplitudes of the acoustic and nonacoustic velocity fluctuations have also been estimated from their data. Though vocal tract area functions are not available, simple models of sound transmission and radiation allow an order-of-magnitude prediction of the contribution to the radiated sound associated with each part of the in-tract velocity field. Although these are preliminary investigations using a restricted data set, the evidence suggests that there is nonacoustic fluctuating flow within the vocal tract, capable of significant sound generation.
The relevance of the shoulder bounce to HRTFs considering head movements
Authors: Pellegrini Renato
Published: July 2002

When measuring HRTF data for use in virtual auditory environments, individual HRTFs of subjects are commonly measured keeping the head in the front position vertical to the upper body. This measured data set is then used to simulate the direction of incidence of virtual sound sources. In real world situations, subjects normally tend to turn their heads only, instead of rotating with the whole upper body when focusing on a sound source. In virtual auditory environments this relative head-to-upper-body rotation is not accurately simulated. This error could theoretically deteriorate the perception of virtual sound sources. HRTF data sets were measured for different shoulder to head positions other than front, and psychoacoustical tests were made to give evidence to the importance of shoulder to head positioning. Timbral aspects and localization accuracy in static and dynamic situations were investigated by auditory testing. Results are discussed.
Role of two ears in upper hemisphere localization
Authors: Morimoto Masayuki
Published: July 2002

It is well known that the elevation of a sound source is determined by spectral cues in the median plane, where the input signals to both ears are regarded as being identical. It is, however, still a problem how the elevation of a sound source is determined in any plane apart from the median plane, because the input signals to both ears are not identical. This paper investigates how the input signals to the left and right ears contribute to determine the elevation of a sound source at any position in the upper hemisphere. The localization tests regarding the elevation in five planes parallel to the median plane were performed. In the localization tests, pinna cavities of both or one of two ears were occluded. The results indicate that both ears play a role in determining the elevation of a sound source in any plane. Furthermore, they infer that the summation of spectral features of input signals to the left and right ears contributes to determine the elevation.
Generation of binaural information using cross ambiguity functions
Authors: Uchiyama Michiaki , Tohyama Mikio
Published: July 2002

This paper describes a method to generate binaural information. First of all, the accuracy of a conventional linear interpolation was examined by using signal-to-deviation ratio (SDR) and spectrum distortion. It was found that the accuracy of left and right direction HRTFs (60−120°) did not deteriorate despite the presence of a wide directional interval (SDR ≂ 23 dB). However, the accuracy of front and back direction HRTFs (0−−60° and 120−180°) deteriorated as the interval broadened. For the shade-side ear’s HRTFs, particularly, it was confirmed that the accuracy of linear interpolation deteriorated dramatically (SDR≂8 dB). Therefore, a method was investigated for generating a shade-side ear’s HRTF by using cross ambiguity functions (CAFs) whose model is related to dichotic listening. A shade-side HRTF could be generated with about a 10-dB SDR from a sunny-side HRTF, while the SDR becomes about 8 dB for a conventional linear interpolation estimation method. In this paper, the independence of HRTFs is also discussed, using the singular value decomposition from the point of view that it is possible to generate a new vector from a linear combination of vectors. In this article the accuracy of a linear interpolation from a base of HRTFs by SDR and spectrum distortion is examined.
Modeling scattering from objects near boundaries using infinite elements
Authors: Burnett David , Holford Richard
Published: July 2002

There has been a rapidly growing interest in using infinite elements for modeling acoustic scattering and radiation from finite objects in unbounded regions. The paper will briefly review current infinite element technology, which is based on radial expansions in curvilinear coordinate systems. Although the radial expansions are derived for free-space conditions (4 pi steradians), the elements are also valid for half-spaces, using local hard, soft, or finite impedance conditions on the boundary. In this approach, wave propagation is modeled in only one half-space, with the boundary conditions on the interface approximating the effects of the other half-space, thereby enabling these types of infinite elements to model scattering from objects near, and on the illuminated side of, the boundary of a half-space. Theory and numerical results will be presented. A more general approach, which models wave propagation in both half-spaces, uses infinite elements in both half-spaces, with appropriate continuity conditions at the interface. This is the approach currently under investigation by the authors; their findings will conclude the paper.
Experimental analysis of two-dimensional vibration of a piano string measured with an optical device
Authors: Tanaka Hideyuki , Nagai Keinosuke , Mizutani Koichi
Published: July 2002

It is well known that a piano string vibrates in the direction not only perpendicular to the soundboard but also parallel to it. The sound decays rapidly at first and then it decays slowly. A change in the predominant direction of vibration of the string from perpendicular to parallel contributes to the compound decay rate [G. Weinreich, Sci. Am. 240, 118–127 (1979)]. In this paper, two-dimensional vibration of a piano string is measured with handmade equipment consisting of two phototransistors. Just after the string was struck with a hammer, it vibrates perpendicularly to the soundboard. Then it vibrates gradually in the direction parallel to the soundboard. As a result, a point of the string moves on an elliptic trajectory. It was found that the fundamental resonant frequency of the perpendicular vibration of the E1 string, for example, which is about 40 Hz, is 0.07 Hz lower than that of parallel vibration. This causes the change of rotating direction of the string every 7 s. The discrepancy of the resonant frequencies is also found in the vibration of the D♯1 string.
European helicopter rotor noise research—Highlights of recent projects
Authors: Splettstoesser Wolf R.
Published: July 2002

Significant advances in understanding, modeling, and controlling helicopter rotor noise have been made in the course of several major European rotor aeroacoustic research projects over the past years. The availability of the German–Dutch Wind Tunnel (DNW), in particular, has enabled research on relatively large main rotor models (up to typically 40% full scale). Here, the simultaneous application and use of blade-mounted absolute pressure sensors to study the very aeroacoustic source mechanisms, of advanced acoustic-data acquisition systems for the radiated noise, of non-intrusive flow visualization as well as flow and blade deflection measurement techniques to investigate the rotor wake and the blade dynamic response has provided much insight into the physics of rotor aeroacoustics, particularly into the rotor blade-vortex interaction (BVI) impulsive noise phenomenon. In efforts towards controlling BVI impulsive noise, in some research projects, active blade-root control techniques have been investigated to favorably affect the BVI geometry and thus minimize the impulsive noise radiation. Much of the experimental data acquired served as benchmark information to enhance the mathematical modeling of the rotor wake, the rotor airloads, and the noise radiation. In this lecture, several major European helicopter aeroacoustic research projects are discussed.
Searching for a time window for timbre: Dynamic spectral profiles
Authors: Schouten M. E. H. , Peeters W. J. M.
Published: July 2002

Subjects were asked to match resonance glides, of various durations and sweep widths around 1000 Hz, with two connected steady-state resonances whose frequencies could be controlled independently. The expectation was that short resonance glides (below 100 ms), would give rise to single, apparently stationary, perceptual events, so that the two matching steady-state patterns would not rise or fall. Long glides, on the other hand, were expected to be perceived as rising or falling and to be matched accordingly. It was hoped that this would provide a basis for an explanation of diphthongs as unitary percepts. The resonances formed part of dynamic spectral profiles and consisted of single formantlike peaks swept along the top of a spectral base, consisting of 25 harmonically or logarithmically spaced components from 200 to 5000 Hz. There was an effect of duration, although it interacted with sweep width. At durations shorter than about 120 ms, subjects placed the two steady profiles that they had to match with the dynamic profile, much closer together than with durations over 120 ms; however, this only occurred if sweep width was more than 600 Hz.
Multidimensional acoustical and psychoacoustical aspects of inharmonic plate and bar tones of real and resynthesized instruments
Authors: Carterette Edward C. , Kendall Roger A.
Published: July 2002

Following a brief review of the mathematics of inharmonic plates and bars, the acoustics and comparative psychoacoustics of the tones of Indonesian bars and plates are considered. Tones were derived from real instruments of the UCLA Department of Ethnomusicology’s Javanese and Balinese gamelans, and from resynthesis of spectral analyses of instrumental tones. The comparative psychoacoustics of the tones was assessed by a number of methods, including JNDs, ratings by MDS similarity-scaling, and verbal-attribute magnitude estimation. Several methods were used to converge on the general perceptual and cognitive properties of the tones. Illustrative findings are: It was shown (1) that Indonesian gamelan makers can and do tune their metallophones in stretched octaves. Their stretchings (1.02 to 1.05 times the normal octave) were fit by a mathematical model which was derived from the data of many gamelans. (2) That the perceptual dimensions of Indonesian and Balinese gongs were interpreted as pitch, ombak (beating), and volume. These dimensions can be mapped to frequency, amplitude modulation, and spectral envelope. It was concluded that gong makers can modify the plates to achieve desired physical properties and correlated perceptual dimensions. One important finding is that the relative positions of two close axially symmetrical modes control the ombak rate.
Cross-modal integration: Synchronization of auditory and visual components in simple and complex media
Authors: Lipscomb Scott D.
Published: July 2002

Recently, there has been a significant amount of interest in the perceptual interaction between auditory and visual (A–V) systems in multimodal contexts. Both psychologists and musicians are beginning to investigate the manner in which a stimulus perceived in one sensory modality may affect the cognitive processing of a stimulus in a separate modality. Beginning in the 1950s, a series of psychophysical investigations revealed cross-modal influences using extremely simple stimuli. Most studies incorporating more complex stimuli have focused on the referential aspect of musical sound, i.e., the ‘‘cognitive congruency’’ of the music and the visual images. The present study will specifically investigate the alignment of accents (i.e., salient events) in the auditory domain with those in the visual domain, and the effect of this alignment on subject perception of the A–V composite. The author will report on three experiments utilizing varying levels of stimulus complexity: single-object animations, animations by Norman McLaren, and motion picture excerpts. A–V alignment was manipulated as the independent variable with subject ratings of effectiveness as the dependent variable. A revised model of film music perception will be presented, proposing a dynamic relationship between stimulus complexity and significance of AV synchronization in the determination of subject ratings.
Music serves as a vehicle in multimedia contexts
Authors: Cohen Annabel J.
Published: July 2002

The vast amount of information in multimedia presentations ought to place inordinate demands on perceptual and cognitive systems. Yet there seems to be a superfluity of resources for processing music in the midst of processing information from visual and verbal sources. Research of the author and others reveals the remarkable ability of listeners to accomplish three kinds of musical processing in multimedia contexts: picking up cross-modal structural similarities, generating meanings and associations, and establishing cross-modal linkages in memory. Emphasizing principles of structural congruence and semantic association [S. E. Marshall and A. J. Cohen, Music Perc. 6, 95–112 (1988)], a framework is described for examining these musical processes in the context of simultaneous visual and verbal input streams [A. J. Cohen, Proc. 5th International Conf. on Music Perc. and Cog. (1998), pp. 13–20]. The present paper highlights how music transports different types of information (e.g., acoustic, temporal-structural, semantic). These different types of information carried by the music can be selectively exploited to accomplish such functions as masking, focusing attention, disambiguation, reminiscence, suspension of disbelief, and creation of aesthetic experience. [Work supported by SSHRC.]
Identification and qualification of instrumental sound sources: Elaboration
Authors: Sémidor Catherine , Couthon Léonie , Barlet Aline
Published: July 2002

From multitrack anechoic recordings of orchestral, or other musical instruments, a quantitative analysis of musical instruments’ power and directivity was conducted [Semidor and Couthon, ICA98]. A qualitative study is now proposed to complete the knowledge about the perceived sound of these same instruments by listeners. The following elements are examined: (1) the ‘‘recognizing threshold’’ of instrumental sources, from a directivity point of view. Particularly, studying the differentiation difference between sources of the same family, or sources with a pronounced directivity and others. (2) The subjective ‘‘qualification’’ of sources in function of directivity, that is the influence of the quantity of information on directivity, on the qualification of the source. (3) Correlation of those results with the quantitative study conducted before. Are the main objective parameters noticed in a subjective point of view? This study is conducted in a first approach in laboratory conditions. Its main applications are sound recording, sound reinforcement, measurement, and simulation in room acoustics. It is well known that taking into account sound sources’ characteristics permits the improvement of a room’s design in order to increase the audience comfort.
Amplification of the difference tones (f2–f1)
Authors: Dutilleux Pierre
Published: July 2002

Difference tones have been known by composers and performers for a long time. Nevertheless, these tones are still difficult to use in a musical context because their appearance is fairly unpredictable. A cello player, Michael Bach, has developed a sophisticated playing technique to produce these tones. Unfortunately the public was not aware of it because it could not here them as well as the player did. Evidence of the presence of the difference tones in the acoustic signal has been found in some cases. In most cases these tones develop in the head of the listeners only at fairly high sound-pressure levels. In order to control the amount of difference tones in a musical sound, a processing method has been developed that artificially produces the difference tones. The squared and filtered instantaneous envelope of the acoustic signal delivers them. These tones can then be mixed to the original sounds as needed and the public can perceive the difference tones even at moderate sound-pressure levels. The system was implemented and performed well during concerts. With a celletto (cello without any sounding box) it is possible to hear only difference tones.