bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustical aspects of chamber music for strings
Authors: Meyer Juergen
Published: July 2002

Playing string quartet music under different room acoustics conditions demands adaptation of the playing technique, particularly of the articulation. If not damped by the bow, violin tones have a 60-dB decay time of about 0.5 s in the highest part of the range, increasing to about 1 s for the lowest tones; in the lower stringed instruments, decay times are as much as 3 s. When damped by the bow, the decay times can be shortened to about 0.1 s. Attack times of staccato violin tones are of the order of 30–50 ms; smoothed attacks may be prolonged up to 200 ms. The temporal structure of a fast staccato tone sequence (14 tones per second) is mainly characterized by breaks in the overtones of the order of 25 ms, whereas the fundamentals seem to be blended together or slurred. These values correspond with typical acoustical criteria of historical and contemporary chamber music halls. To determine the reaction of musicians to the boundary conditions as well as the listeners’ impressions, a test was performed by playing the same programs of short and different music examples during only 3 h in three halls having different room acoustic properties.
A new approach to low-frequency room noise criteria
Authors: Hellweg Robert D. , Pei Hsien-sheng (Jason)
Published: July 2002

There are three different and conflicting room criteria methods that are widely used by consultants today: room criterion (RC), balanced noise criterion (NCB), and noise criteria (NC). The RC procedure is documented in ANSI S12.2-1995 ‘‘Criteria for evaluating room noise’’ and the ‘‘1995 ASHRAE Applications Handbook;’’ the NCB procedure in ANSI S12.2; and the NC ‘‘tangency’’ method in the ‘‘1991 ASHRAE Applications Handbook.’’ There are significant differences in the three criteria, especially for low frequencies, which can impact the costs both of new designs and of remedial actions to existing facilities. In order to promote a resolution to these differences, NOISE-CON96 and NOISE-CON97 held special technical sessions on room noise criteria. Based on our analysis of published research and on experiences dealing with other noise sources, the authors suggest an approach to resolve the low-frequency differences: (1) establish basic room criterion curves for noises without abnormal temporal or spectral characteristics, and (2) apply correction factors to the measured data to account for abnormal features before comparison to the basic criterion. This paper presents details of the suggested approach, the status of revisions to ANSI S12.2 and to the ASHRAE handbook, and other ongoing research.
Effects of the acoustical characteristics of road pavements on long-range sound propagation
Authors: Bérengier Michel C. , Hamet Jean-François , Pichaud Yves
Published: July 2002

For many people in industrial countries, traffic noise is considered as one of the most important elements of discomfort. In the past 15 years, the only protections for the neighboring populations against road traffic noise were road barriers. For a short time, the modification of the road pavement can partly contribute to decrease this nuisance. Recent research on the acoustical performances of road pavements have been carried out, and, a first ranking in the close proximity of the road has been established using a standardized method. From these first results, it is possible to predict the road traffic noise far from the road taking into account the acoustical characteristics of various pavements. This prediction procedure requires the theoretical knowledge of all the physical phenomena occurring along the propagation path. This is the reason why the model used for calculation integrates ground effects over mixed surfaces and atmospherical parameters such as vertical sound-speed gradients. The pavement effect is introduced in the model through the sound power spectrum of the tire/road noise source located a few centimeters above the road surface. The predicted results are advantageously compared to in-situ measurements carried out over some hundred meters.
What are we still doing wrong in assessing occupational noise exposure?
Authors: Johnson Daniel L.
Published: July 2002

Sound level meters/dosimeters and measurement/prediction procedures have improved greatly over the last 20 years. Instrumentation nowadays can be expected to assess the energy of impulsive noise properly, provide a time history of the noise exposure, and have a sufficient dynamic range. Nevertheless, some improvements are still needed. These improvements are to (1) rework the SLM standards so the measurement envelope for all instruments, regardless of type, can be specified to be the same; (2) provide better methodology for determining when infrasound and ultrasound need to be measured; (3) provide a mechanism for properly assessing intermittent exposures, including the inclusion of effective quiet; and (4) finally, as a visionary dream, provide long-term measurement of occupational/nonoccupational noise exposure coupled with environmental factors as well as health factors such as blood pressure and hearing threshold levels. The need for and the details of each of these improvements will be discussed.
Hearing hazard evaluation and hearing conservation in a very reverberant indoor firing range
Authors: Sachs Felix Z.
Published: July 2002

OSHA and Army regulations differentiate steady from impulse noise exposure, but the boundary is fuzzy. A 300-m research indoor firing range produced 10-s reverberation times resulting in noise exposures straddling this boundary. Worker exposure was assessed as both steady and impulsive under both OSHA and Army criteria. The highest peak level was 164.9 dBP, the maximum ‘‘slow’’ level was 133 dBA. Hearing protector calculations for steady noise used the NIOSH ‘‘assumed protection values’’ method. Army noise criteria use 3-dB exchange between level and time. Any parallel filter analyzer could measure the time-averaged octave band levels. OSHA uses 5-dB exchange rate, a rare feature on parallel filter analyzers. Administrative controls were most stringent for the Army steady noise method and least restrictive for the MIL-STD-1474 impulse noise method. For OSHA steady noise, allowable rounds per day depended on the number of shots fired per burst due to the 5-dB exchange rate. [The opinions or assertions contained herein are the views of the author and are not to be construed as official or as reflecting the views of the Department of the Army or the Department of Defense.]
Sonochemical reactor optimization using computational acoustics techniques
Authors: Migeot Jean-Louis Y.
Published: July 2002

The designer of a sonochemical reactor is confronted by a complex problem characterized by a large number of design parameters: reactor shape and dimensions, number, positions and power of ultrasound sources, frequency, solvant, temperature, etc.? A wide parametric study cannot be conducted experimentally due to time and cost constraints and a numerical model seems the only viable approach. Such a model must reflect the inherent complexity of the system and at the same time be simple enough to provide relevant results in a reasonable amount of time. This paper proposes an iterative modeling methodology based on the following steps: (1) a finite-element model of the fluid in the reactor, (2) a relationship between local acoustic pressure and gas fraction released by cavitation, and (3) equivalent acoustic properties for the cavitating medium. After a few iterations, this model leads to two main results: (1) the pressure field in the reactor and (2) the cavitation zone and the distribution of gas fraction in this cavitation zone. One can further postprocess the pressure field to predict the streaming forces on the fluid and therefore predict the flow pattern using classical CFD techniques.
Slow nonlinear dynamics in rock
Authors: Guyer R. A. , McCall K. R. , Van Den Abeele K. E-A.
Published: July 2002

Recent experiments on the elastic response of Berea sandstone show that nonlinear excitation of a resonant bar is accompanied by a slow dynamics, a dynamics involving time scales many orders of magnitude longer than the excitation period, 2π/ω. A lumped-element description of a resonant bar is developed. A complementary phenomenological theory of slow nonlinear dynamics is postulated. This phenomenological theory is used in the lumped-element description to explain the experiments. The results illustrate the possibility of experimental exploration of slow nonlinear dynamics using the acoustic analogue of a NMR experiment. [Work partially supported by OBES, Engineering and Geosciences Grant No. W7405-ENG-36, the National Science Foundation Grant No. EAR-9528965, and the Institute of Geophysics and Planetary Physics at Los Alamos National Laboratory.]
Shared attention to simultaneous stimulation
Authors: Hafter Ervin R. , Bonnel Anne-Marie , Gallun Erick
Published: July 2002

Shared attention is studied using a dual task, where observers respond independently to two simultaneously presented signals. Cost is defined by a loss in performance relative to one signal tested alone. Standard stimuli are followed by signals that can be either incremental or decremental changes in level, frequency, or spatial location, relative to the standard. In a ‘‘detection’’ paradigm, the subject says ‘‘change’’ for signals of either polarity, but in an ‘‘identification’’ paradigm where all trials present signals, the subject responds with the polarities. Comparing results with fixed standards to those with a standard that roves from trial to trial shows that a major determinant of the cost of shared attention is the type of memory used in the task. Fixed standards, which show reliance on absolute memory, yield relatively high performance, but there is a cost of shared attention. Conversely, with roved standards, where absolute memory cannot be used, performance is relatively low and there is no cost. Thus a dissociation is found in which the additional process used in the context-coding mode is costly for attention but not performance. [Research supported by the Natl. Inst. of Health (NIDCD 07787), USA, Univ. of California, USA, and CNRS, France.]
Electrophysiological studies of auditory spatial attention
Authors: Hillyard Steven A. , Teder-Salejarvi Wolfgang A.
Published: July 2002

Recordings of event-related potentials (ERPs) from the scalp can track the processing of auditory signals through the afferent pathways from brainstem to cortex. ERP components elicited in auditory cortex during the time range 50–200 ms are highly sensitive to selective attention, with attended sounds eliciting greater amplitudes than unattended sounds. Using combined ERP and behavioral measures, the allocation of auditory attention was studied in a free-field array of seven or eight speakers. Brief noise bursts were presented in a random, fast-paced sequence from all the speakers, and subjects were required to attend to the sounds at one location while ignoring the others. Both behavioral detection responses and ERP amplitudes were highest for stimuli at the attended location, with a progressive falloff for responses to increasingly distant sound sources. These data revealed the shape of spatial tuning curves for auditory attention in free-field and support gradient models of attentional allocation. An analysis of ERP waveforms indicated that spatial selectivity is established by consecutive stages of progressive fine-tuning over the time range 50–350 ms poststimulus. [Work supported by Grants from NIMH and ONR.]
On the use of assumed spatial distributions for the vibration analysis of built-up plate structures
Authors: Fulford Ross A. , Petersson Bjorn A. T.
Published: July 2002

For the vibratory analysis of built-up structures, traditional pointlike connections cannot be applied where the interface is large and the wavelength small. Instead, the spatially distributed wave field has to be accounted for whereby the interface has to be considered as either a surface or, for a one-dimensional contact, a line. To analyze such, the traditional approach is to rely upon numerical methods such as the finite element method. These approaches are often cumbersome, however, in that they reveal little about the physics of the systems. As an alternative, this paper considers the development of a method whereby the spatially distributed properties of a line contact can, with the salient physics retained, be included in simplified mathematical models. The basis of the method is to describe the spatial properties with respect to Fourier components and hypothesize that the first order, i.e., uniform, component is most influential. Via an assumed uniform force distribution, the approach is applied, with some promise, to the problem of predicting the vibratory power transmission between a box and a plate.
A multifrequency inversion method for geoacoustic parameters
Authors: Zhang Renhe , Li Fenghua , Luo Wenyu
Published: July 2002

The effects of the bottom parameters on the sound field are discussed theoretically and two characteristic angles relevant with geoacoustic parameters are put forward. Based on the theoretical analysis of the characteristic angles, a multifrequency inversion scheme is given as follows: High-frequency and far-field data are used to invert the upper bottom parameters first, and then low-frequency and near-field data are used to invert the lower bottom parameters. Simulated geoacoustic inversions based on simulated annealing method and the beam-displacement ray-mode (BDRM) sound propagation model are applied to the test cases from the 1997 Geoacoustic Inversion Workshop. The simulated inversions show that the inversion results of multi-frequency inversions are more accurate and reliable than those of the single-frequency inversions. The multi-frequency inversion scheme is also applied to the experimental data from ’96 China–US Far Yellow Sea Experiment. [Work supported by NSF of China and ONR of USA.]
Pitch-synchronous decomposition of mixed-source speech signals
Authors: Jackson Philip J. B. , Shadle Christine H.
Published: July 2002

As part of a study of turbulence-noise sources in speech production, a method has been developed for decomposing an acoustic signal into harmonic (voiced) and anharmonic (unvoiced) components, based on a hoarseness metric [Muta et al., J. Acoust. Soc. Am. 84, 1292–1301 (1988)]. Their pitch-synchronous harmonic filter (PSHF) has been extended (to EPSHF) to yield time histories of both harmonic and anharmonic components. These corpus include many examples of turbulence noise, including aspiration, voiced and unvoiced fricatives, and a variety of voice qualities (e.g., breathy, whispered). The EPSHF algorithm plausibly decomposed breathy vowels, but the harmonic component of voiced fricatives still contained significant noise, similar in shape to (though weaker than) the ensemble-averaged anharmonic spectrum. In general, the algorithm performed best on sustained sounds. Tracking errors at rapid transitions due to jitter and shimmer were spuriously attributed to the anharmonic component. However, the extracted anharmonic component clearly exhibited modulation in voiced fricatives. While such modulation has been previously reported (and also in hoarse voice), it was verified by tests on synthetic signals, where constant and modulated noise signals were extracted successfully. The results suggest that the EPSHF will continue to enable exploration of the interaction of phonation and turbulence noise.
A safe method of truncating an echogram calculated by a sound ray technique
Authors: Embrechts Jean J.
Published: July 2002

A recent international test of room acoustics computer simulations (presented by Prof. Vorlander at the 1995 ICA Congress in Trondheim) has shown the deviations that can occur between several programs when they are calculating the same acoustical parameters. Beyond data evaluation, the method of deriving these parameters from a computed echogram may be of significant importance. Therefore, the different steps in the calculation of the echogram by a sound ray technique are analyzed. In particular is shown the influence of the time discretization step (dt) and of the late part of this diagram (the tail) on the parameters T30 and EDT. An algorithm is proposed to reduce their influence. It consists in the truncation of the echogram at a well-chosen instant, independent of dt. Indeed, the tail of the echogram is greatly affected by statistical errors, and it is replaced by a linear decay, with a precisely defined slope, which is also independent of dt. The acoustical parameters are finally derived from this modified echogram. This algorithm is compared with other crude truncation procedures. It is shown that it significantly enhances the stability of the calculated results. Its performance is limited by the definition of T30 and EDT themselves.
Measuring the traffic noise in Valencia
Authors: Gaja E. , Manglano J. L. , Reig A. , Sancho S.
Published: July 2002

The City of Valencia has approximately 700 000 inhabitants. Due to this large population, the city is subjected to heavy traffic, and, as a consequence, high acoustic pollution. The Townhall of Valencia is very worried about this acoustic pollution and has commissioned the Laboratory of Acoustics in the Universidad Politécnica de Valencia to work on various projects within the city. Due to the work the department has carried out on behalf of the Townhall of Valencia, it has gained experience measuring with the sonometer. An efficient way of measuring noise within the city that avoids wasting long periods of time has been developed. Continuous 24-hour measurements have been taken by sonometers located in different stations spread out around the city and controlled by the Townhall. With these measurements a method that indicates when to measure, in the morn- ing, afternoon, and at night, as well as the exposure time, has been designed. Other parameters, such as characteristics of the traffic, types of streets, etc., have been considered in order to obtain the time and the exposure time of measuring that represents the level of the noise.
A mathematical model for the evaluation and prediction of the mean energy level of traffic noise in Caracas
Authors: Montbrun Nila , Rastelli Victor , Bouza Alexis , Montbrun Di Filippo Jenny , Sánchez Yamilet
Published: July 2002

The increase of the population in capital cities has provoked a serious increase in the acoustical pollution, where the real cause is the traffic noise. Over ten years, sound levels have been measured in several points in the city of Caracas. The main objective was to establish a mathematical model for the mean energy level of traffic noise. This model is a modification and improvement of the model that was used in Germany and Italy. Parameters of the fundamental equations were obtained by the use of large experimental data. In the particular case of Venezuela, an innovation was introduced by the consideration of the motorcycle noise. The obtained model can be used to predict noise pollution in cities and contour maps can be made as well.
Response curves of bubbles
Authors: Lauterborn Werner H. , Mettin Robert
Published: July 2002

A spherical bubble in a liquid is a nonlinear oscillator that can be set into radial (and surface) oscillations by a sound field. The nonlinear response depends on a set of parameters, mainly the bubble radius, the driving amplitude, and the driving frequency. The response is involved. It consists of an intertwined set of nonlinear resonances, that besides turning over, comprises period doubling sequences to chaos when going up and down in bubble radius, driving frequency, or driving amplitude. A specific feature is the giant resonance appearing for small bubbles and being the host for single bubble sonoluminescence. A set of response curves will be given to acquaint the listener with some understanding of how bubbles behave in a sound field.
Diffraction of a sound wave on an open end of a half-infinite waveguide with impedance walls and impedance flanges
Authors: Shenderov Evgeni L.
Published: July 2002

Diffraction of a plane sound wave on an open end of a waveguide with impedance walls, connected with impedance flanges, is considered. A plane wave is incident upon the waveguide from open space. As a result of diffraction, part of the sound energy is scattered in the half-space and the other part of the energy penetrates into the waveguide. Results of the solution of this problem can be used for calculation of the amplitude of a sound wave, scattered by a hole, in particular, in the backward direction. Besides, one can calculate the sound energy penetrating into the waveguide. Absorption of the energy in the waveguide can be taken into account by introducing impedances with nonzero real parts. The distortion of the directivity pattern of a system of receivers located in the waveguide can also be obtained. The solution of the problem has been carried out by the integral equation method using Green’s function of the half-space with an impedance boundary. The equation was solved by expanding an unknown function in the eigenfunctions of the waveguide. It reduces to an infinite set of linear algebraic equations. Results of computation of bistatic diagrams of scattering on the hole and diagrams of backward scattering are presented.
Formulation of a boundary value problem by a new principle of diffraction
Authors: Ueda M.
Published: July 2002

A new diffraction principle, that is formulated from a viewpoint centered at an observation point by considering wave propagation in a space seen by the observer virtually, has been proposed [M. Ueda, J. Acoust. Soc. Am. 95, 2354–2361 (1994)]. This new approach allows us to grasp a simple understandable concept of diffraction and to formulate the first rigorous and unique representation of diffracted waves, that is, it satisfies both the wave equation and hard or soft boundary conditions for 3D objects. Thus the diffraction problem for scalar waves is solved by this approach. The presentation is expressed in a form of integral equation. Thus it is another problem to formulate the algorithm to find the solution. In this paper the numerical procedures to formulate the algorithm are described. The boundary value problem is formulated by a new principle of diffraction. The structure of the response function, that shows mutual dependence of potentials at the surface of the object, is made clear by this new formulation and it reveals a special role of the edge point in the diffraction process. The usefulness of the response function is made clear by the analysis of diffraction arisen from a semi-infinite plane.
Short-term auditory memory interference: The effect of speech pitch salience
Authors: Ueda Kazuo , Seo Naoko
Published: July 2002

The interference effect of speech pitch salience on tone pitch recognition and digit recall was observed. For the pitch recognition task, 11 subjects had to recognize pitches of two harmonic complex tones separated by a 5-s retention interval. During the interval, no sounds or six randomly chosen intervening sounds, either complex tones from a semitone scale or spoken digits, were presented. The speech included natural stimuli and synthesized stimuli with the original and with flattened pitch contours. For the digit recall task, the subjects had to recall serially the intervening digits. They had to perform both tasks in the dual task conditions. The results were: (1) the synthesized speech was equally intelligible as natural speech, (2) both pitch recognition errors and digit recall errors were significantly increased in the dual task conditions compared to the single task conditions, and (3) the synthesized speech with the flattened pitch contours as well as the complex tones had the most deteriorating effect on tone pitch recognition. Thus it became obvious that the weak interference effect of natural speech on tone pitch recognition was due to their pitch ambiguity. [Work supported by Grant-in Aid for Scientific Research and by Nagai Research Fund.]
FEM aided structural intensity measurement method for thick body
Authors: Koike Yoshikazu , Aramaki Chie , Nakamura Kentaro , Ueha Sadayuki
Published: July 2002

Many studies about structural intensity(SI) measurement, which is useful to know transmission paths of the vibration energy in a body, have been reported. Most of the works, however, have been done only for thin bodies, such as beams or plates because of the difficulty in measuring interior vibration distribution in thick structures. In this paper, a new SI measurement method applicable to thick bodies is proposed. The entire vibration distribution in the structure is calculated by dynamic analysis of the finite element method (FEM) from the partial surface vibration data if the elastic parameters, the sound velocity, sizes of a body under study, and the locations of excitation and absorption are given a priori. The entire SI distribution can be estimated using the results. Based on this method, several SI measurements for thick vibrating bodies have been successfully carried. The merits of this method are as follows. (1) Only surface vibration data of the structure is needed to estimate the entire SI distribution. (2) It is applicable for arbitrary 3-D structures if the conditions mentioned above are satisfied a priori. The principle and the experimental results are described in the report.
Resilient mounting of engines
Authors: Nilsson Anders , Kari Leif , Haettel Romain
Published: July 2002

In any type of vehicle a number of mechanical sources generate structure-borne noise. The energy is transmitted from excitation points to surrounding structures. These radiate noise externally and internally. The most commonly used method to reduce the radiated noise is to mount the sources resiliently to the supporting foundation. The insertion loss due to the resilient mounting can in many cases be extremely poor. This could be due to the break down of the stiffness of the foundation in the audible frequency range. At the same time the stiffness of the mounts could be increased drastically as compared to the static stiffness. A test rig and measurement procedures for determining the dynamic properties of mounts are described. Measured and predicted results are compared for rubber mounts with simple geometries. The insertion loss of a resilient mounting very much depends on the location of the engine feet on a foundation. Case studies relating to passenger vessels and catamarans built with aluminum and sandwiched are presented. Reduction of both noise and weight is discussed.
Using a personal computer platform to develop an information-rich learning environment for instruction in acoustics
Authors: Celmer Robert
Published: July 2002

Teaching students the subject of acoustics involves large amounts of technical information, including equations, derivations, and example problems. It also requires the explanation of a multitude of concepts, many of which are often difficult for students to grasp and equally frustrating for the instructor to disseminate. Part of the difficulty can be traced to the use of text (a visually based medium) to teach about sound (an aurally based medium). This presentation will describe the development of multimedia techniques for in-class presentation of acoustics instruction. Some of the materials were developed using certain authoring applications, drawing and animation programs, sound manipulation software, as well as 3D CAD and spectral analysis applets. In addition to the classroom materials, self-paced exercises have been developed to review the material in a student-centered learning environment, along with scheduled interaction time with the instructor for questions, clarifications, and test administration. Demonstrations of the materials for the instruction of acoustical concepts, as well as case studies, will be presented.
FishMASS: ADCP technology adapted to split-beam fisheries echo sounding
Authors: Gordon R. Lee , Zedel Len
Published: July 2002

The objective of FishMASS, a NSF-funded research project, is to adapt elements of ADCP technology into split-beam echo sounding. Improvements come from wide bandwidths (50% bandwidth at 300 kHz), incorporation of velocity (Doppler) data, and addition of autonomous operation (battery and recording capacity to allow operation for up to a year). Wide bandwidths can reduce the uncertainty of single-target echoes from 3–6 dB (characteristic of narrowband signals) down to a fraction of a dB. It enables better differentiation of single-target from multiple-target echoes (when targets are separated either radially or at different ranges). The 50% bandwidth gives access to at least part of the target echo spectrum. These three advantages combine to improve both target discrimination and target classification. Split-beam sonars can sometimes track individual targets through their beams to obtain swimming speed. Doppler processing (identical to the processing of broadband ADCPs) will be added to obtain single-ping swimming speeds, both of individuals and groups. Mean velocities allow observation of migration and flux. Statistical products such as velocity standard deviation enable observation of feeding or predator-avoidance behavior. Correlations of velocity with, for example, TS may further aid species classification.
On the use of elements of piano notes to improve the identification of polyphonic piano sounds
Authors: Rossi Lucile , Girolami Gerard
Published: July 2002

An original algorithm for identifying polyphonic piano sounds signals, based only on the use of frequential positions of note partials, has been developed by the authors [Rossi and Girolami, J. Acoust. Soc. Am. 100, 2842(A) (1996)]. This paper presents a study of the evolution of the amplitude of piano notes’ partials and their energy, carried out to see if these elements can be used to improve the identification of notes in cases like short notes, long notes, repeated notes, and polyphonic sounds. In the case of monophonic sounds, Doval [Ph.D. thesis, University of Paris VI (1994)] has shown that the evolution of the amplitude of piano notes’ partials along with their frequency variations can be used to identify the fundamental frequency of signals. The case of polyphonic piano sounds (several notes played simultaneously on the same piano) is more difficult, but if a particular behavior of partials belonging to a given note could be detected, for example, synchronous variations or typical amplitude evolution, the sorting of partials would be more reliable, the identification of notes more efficient, and the detection of notes’ onsets more precise.
Musical instrument synthesis by nonregenerative nonlinear processing
Authors: Beauchamp James W.
Published: July 2002

When a sinusoid is distorted by a nonlinear function, a complex tone is produced. An nth degree polynomial nonlinear function produces an output waveform containing n harmonics. Moreover, when the amplitude of the input sine wave changes, the harmonic structure of the output changes in a way which can be predicted from the polynomial coefficients. Also, the coefficients can be set to provide any arbitrary magnitude spectrum when the sine wave has a particular amplitude, say unity. Generally, the spectral centroid, a measure of the spectral bandwidth, increases monotonically as the input amplitude increases. This possibility is enhanced by placing a high-pass filter at the output. An acoustic instrument sound can be imitated by (1) performing time-variant harmonic analysis on it, (2) selecting a representative spectrum from it to determine the polynomial coefficients, (3) measuring the sound’s time-varying spectral centroid and rms amplitude, and (4) coercing the sine wave amplitude to change so that the spectral centroid of the output matches that of the original sound. The amplitude of the original sound is matched by a multiplier at the nonlinear output. This method has been successfully used to synthesize instruments such as the trumpet, saxophone, clarinet, oboe, and piano.
Time-domain modeling and numerical simulation of timpani
Authors: Rhaouti Leïla , Joly Patrick , Chaigne Antoine
Published: July 2002

Timpani are made of a circular elastic membrane stretched over a enclosed air cavity and set into vibrations by the impact of a mallet. The motion of the membrane is coupled with both the external and internal sound pressure field. A time-domain model of this instrument has been developed in order to investigate the influence of the main geometrical and physical quantities on the resulting sound. This model consists of a set of partial differential equations which govern the displacement of the membrane and the acoustic pressure inside and outside the cavity, respectively. These equations are coupled with a nonlinear differential equation which governs the excitation by the mallet. A numerical scheme has been derived from this model using three-dimensional finite element methods. Absorbing conditions have been implemented to simulate the free space. The validity of the model is illustrated by successive snapshots showing both the pressure field and membrane displacement. In addition, time histories of energetic quantities are presented for a better understanding of energy balance between membrane, cavity, and external space in real instruments. Sound examples are obtained by simulating the sound pressure at specific positions corresponding to the player’s ears.
Low-flying military aircraft noise—Operational flying
Authors: Flindell Ian H. , Weston Ralph J.
Published: July 2002

The distribution of military low-flying training across Great Britain is essentially random within the permitted low-flying areas, although there are certain areas where above-average concentrations of flights occur. However, there are very little data available on actual flight tracks as flown, and training operations are normally conducted so as to encourage as much flexibility as possible. This means that calculations of noise exposure in different areas have to be based to a considerable extent on assumptions about typical flight track distributions, using aircraft noise source reference levels extrapolated from data collected under specially arranged measurement trials. This paper reports the results of a number of recent field surveys carried out in representative overflown areas which show generally good agreement between the assumptions made about typical flight distributions, the special trials data, and actual operational flying.
Predictions for upscaling sonoluminescence
Authors: Hilgenfeldt Sascha , Lohse Detlef
Published: July 2002

Detailed comparison between the hydrodynamical/chemical approach towards single bubble sonoluminescence (SBSL) and recent experimental data is offered. In particular, the water temperature dependence is focused on. Moreover, detailed predictions of how to upscale SBSL are presented.
Sonoluminescence and its neighbor cavitation bubble luminescence
Authors: Lauterborn Werner H. , Kurz Thomas
Published: July 2002

The field of sonoluminescence, the making of light from sound, and the related field of cavitation bubble luminescence are reviewed. The single-bubble acoustic trap invented by Crum has started great activities through the enhanced possibilities of observing the dynamics of a single bubble in great detail. The question of spherical versus aspherical collapse in light emission and problems of enhancing the light output are addressed. For sonochemistry the case of many bubbles versus a single bubble is important. Some results on the dynamics of multi-bubble systems and their light emission are given. One neighbor of sonoluminescence, i.e., cavitation bubble luminescence from laser-induced bubbles, may be of great help for clarifying some aspects of light emission from collapsing bubbles.
Apparent effects of the use of digital hearing aid ‘‘CLAIDHA’’ on the several hearing functions of the impaired listeners
Authors: Hidaka Hiroshi , Sasaki Naoko , Kawase Tetsuaki , Takasaka Tomonori , Ozawa Kenji , Suzuki Yôiti , Sone Toshio
Published: July 2002

The portable digital hearing aid system CLAIDHA (compensate loudness by analyzing input signal digital hearing aid), which employs frequency-dependent amplitude compression based on narrow-band compensation, has been developed [Asano et al. (1991)]. The patients with hearing loss using hearing aids experience a new sound world as amplified by the device. It seems to be important to know the apparent change of hearing function by the use of a hearing aid. In the present study, the apparent effects of CLAIDHA amplification especially on the intensity discriminability and the masking function were examined in the patients with hearing loss. The results obtained indicate that the CLAIDHA amplification often improves not only the apparent thresholds but also the apparent masking relation between two tones as compared to simple linear amplification; i.e., in the CLAIDHA amplification, more signals could be heard under the same masking condition. On the other hand, from the viewpoint of intensity discriminability, when the compression ratios of the amplification were larger than two, the IDL (intensity difference limen) rapidly increased more than expected. The advantage of the compressive amplification of the CLAIDHA system will be discussed.
The effect of frequency-modulation systems on the communication skills of deaf–blind children
Authors: Franklin Barbara
Published: July 2002

Classrooms and other public places provide a poor acoustical environment for individuals who have a hearing loss, and it is often difficult for them to discriminate between the speaker and background noise with a hearing aid. The use of a frequency-modulation (FM) system can often reduce this problem by providing a constant sound pressure level of the speaker’s voice. The effect of FM systems on communication skills of deaf–blind children was investigated using a single-subject alternating treatment design. There was a marked improvement for a number of the subjects in the FM condition. The results will be discussed along with individual case studies. FM technology has exploded. A relatively new type of FM receiver is now available which combines the hearing aid and FM in a single behind-the-ear unit (BTE/FM). The most recent development is a ‘‘boot’’ attachment which changes a standard over-the-ear hearing aid into a FM system. An overview of these FM systems will be presented. [Work supported by OSERS.]
The effect of rib resonances on the vibration and wave scattering of a ribbed cylindrical shell
Authors: Marcus Martin H.
Published: July 2002

Wave phenomena have previously been observed for cylindrical shells with periodically spaced ribs. The understanding of these waves, such as Bragg and Bloch waves, requires knowledge of the periodicity of the ribs and how they interact with each other. In this paper, the understanding of the individual rib resonances gives explanation to the resonance bands that are seen in both experiment and numerical simulations. For these resonances, periodicity becomes less important in computing the frequencies of resonances on the shell and in the far field. However, lack of periodicity in rib spacing and size affects the width and strength of these resonance bands. [Work supported by ONR.]
Reference-beam detection for scanning laser acoustic microscopy
Authors: Cywiak Moisés , Solano Cristina , Wade Glen , Isakson S.
Published: July 2002

Reference-beam detection is inherently superior to the knife-edge detection currently in use in scanning laser acoustic microscopy. This new detector makes use of a reference beam, retarded 90 degrees, which is mixed coherently in a photodiode with the acoustically obtained image-modulated beam. The new detector has an isotropic transfer function which is circularly symmetrical around its highest value, namely the zero-frequency point in the spatial spectrum. This property makes it possible to detect spatial frequencies in all directions simultaneously and with equal sensitivity and simplifies the associated electronics. It also makes possible the employment of acoustic evanescent-wave detection so that ultrasound of low temporal frequency can be used and at the same time high spatial frequencies can be detected for obtaining high resolution. Oblique insonification, required for best operation in the knife-edge detector, is thus not preferred in the reference-beam detector and the resultant Doppler shift in the detected frequency of the transmitted zero-order acoustic waves is avoided.
Observing middle and inner ear mechanics with novel intracochlear pressure sensors
Authors: Olson Elizabeth S.
Published: July 2002

Intracochlear pressure was measured in vivo in the base of the gerbil cochlea. The measurements were made over a wide range of frequencies simultaneously in scalae vestibuli and tympani. Pressure was measured just adjacent to the stapes in scala vestibuli and at a number of positions spaced by tens of micrometers, including a position within several micrometers of the basilar membrane, in scala tympani. Two findings emerged from the basic results. First, the spatial variation in scala tympani pressure indicated that the pressure is composed of two modes, which can be identified with fast and slow waves. Second, at frequencies between 2 and 46 kHz (the upper frequency limit of the measurements) the scala vestibuli pressure adjacent to the stapes had a gain of approximately 30 dB with respect to the pressure in the ear canal, and a phase which decreased linearly with frequency. Thus, over these frequencies the middle ear and its termination in the cochlea operate as a frequency independent transmission line. A subset of the data was analyzed further to derive the velocity of the basilar membrane, the pressure difference across the organ of Corti complex (defined to include the tectorial and basilar membranes) and the specific acoustic impedance of the organ of Corti complex. The impedance was found to be tuned in frequency.
Cancellation model of pitch perception
Authors: de Cheveigné Alain
Published: July 2002

A model of pitch perception is presented involving an array of delay lines and inhibitory gating neurons. In response to a periodic sound, a minimum appears in the pattern of outputs of the inhibitory neurons at a lag equal to the period of the sound. The position of this minimum is the cue to pitch. The model is similar to the autocorrelation model of pitch, multiplication being replaced by an operation similar to subtraction, and maxima by minima. The two models account for a wide class of pitch phenomena in very much the same way. The principal goal of this paper is to demonstrate this fact. Several features of the cancellation model may be to its advantage: it is closely related to the operation of harmonic cancellation that can account for segregation of concurrent harmonic stimuli, it can be generalized to explain the perception of multiple pitches, and it shows a greater degree of sensitivity to phase than autocorrelation, which may allow it to explain certain phenomena that autocorrelation cannot account for.
Integral formulations for the vibroacoustic characterization of a cello
Authors: Langrenne C. , Garcia A.
Published: July 2002

An isoparametric element formulation is used for implementing the Helmholtz integral formulation associated with arbitrary shaped three-dimensional bodies [M. Bonnet, Eyrolles editions, 1997]. The inverse problem is solved, i.e., the normal velocity distribution is determined on the surface of the instrument. In this way, the direct problem can be solved: the acoustic field can be calculated anywhere (near field, far field). Because of difficulties in solving the inverse problem (ill-posed problem, i.e., of which the solution is very sensitive to the errors of measurements), a regularization method is used to find the nearest solution of the real one [C. Langrenne and A. Garcia, Inter-Noise 97 Proc., Hungary, August 25–27 (1997)]. Experimental vibroacoustic characterization is done from holography measurements, without any contact with the stucture of the instrument, with the help of a robot describing a cylindrical surface.
Chat space models
Authors: Cohen Michael
Published: July 2002

Shared virtual environments, especially those supporting spatial sound, require generalized control of user-dependent media streams. Nonimmersive perspectives in virtual environments enable fluid paradigms of perception, especially in the context of frames-of-reference for conferencing and musical audition. Traditional mixing idioms for enabling and disabling various sources employ mute and solo functions, which, along with cue, selectively disable or focus on respective channels. Exocentric interfaces which explicitly model not only sources, but also location, orientation, directivity, and multiplicity of sinks, motivate the generalization of mute/solo & cue to exclude and include, manifested for sinks as deafen/confide & harken, a narrowing of stimuli by explicitly blocking out and/or concentrating on selected entities. Such functions can be applied not only to other users’ sinks for privacy, but also to one’s own sinks for selective presence. Figurative representations of these functions are introduced, virtual hands to be clapped over avatars’ ears and mouths. An audibility protocol is described, comprising revoke, renounce, grant, and claim methods, invocable by these narrowcasting commands to control superposition of soundscapes, and a taxonomy of modal narrowcasting functions is proposed. Applications include groupware for collaboration and teaching, teleconferencing and chat spaces, and authoring and manipulation of distributed virtual environments.
Improving the performance of ultrasonic range meters
Authors: Alonso-Sanudo Monica , Gracia Cesar , Olmos Pedro
Published: July 2002

Range meters based on the time of flight of ultrasonic pulses are instruments widely used in a large number of daily operations. However, their performance may be seriously degraded when the environment has plenty of spurious obstacles masking the true target, a situation not rarely encountered in practice. Here some methods to reduce the influence of such echoes on monofrequency, wide-aperture range meters are analyzed. The discussion is focused on the use of that information contained in the pulse that allows one to find most of the target features. Relevant parameters to play with are the ratio between transmitted and received pulse lengths and energies, as well as the time evolution of the echo pattern. These techniques, suited for low-cost and small-size equipment, have been simulated with the help of a simple geometric ray model, showing the significant improvement achieved in the rejection of those spurious echoes that would lead to erroneous range determination if only time counting were implemented. In addition, experimental results obtained with a uP-based prototype employing 40-kHz ultrasound pulses and apertures of 30 deg are presented and discussed.
Adaptation of underwater acoustic propagation models to infrasonic atmospheric propagation
Authors: Norton Guy V. , Kinney Wayne A. , Ramsdale Dan J. , Whitaker Rodney W.
Published: July 2002

For decades, the U.S. Navy has been engaged in the development of underwater acoustic propagation models to aid in the design and effective use of sonar systems. Much of the expertise created in this development is applicable (directly or by modification) to challenges associated with the use of infrasonic arrays to monitor CTBT compliance. The results of the modification and adaptation of some existing underwater acoustic propagation models to predict and analyze acoustic-gravity wave signals in both a range-independent and a range-dependent atmosphere will be reported. Examples are given that include the effects of range-dependent winds on received signal amplitude and arrival direction for the case of an airborne explosive source and a ground-based infrasonic receiver. Frequencies of interest are in the band from 0.02 to 10 Hz, and ranges of interest are 1000 km or less. [Work sponsored by the U.S. Department of Energy and the Los Alamos National Laboratory.]