bluuebunny/crossref_metadata_2025 huggingface dataset:
Sound power of musical instruments measured with sound intensity in a practice room
Authors: Fidecki Tadeusz
Published: July 2002

The data on sound power of musical instruments published by various authors demonstrate considerable spread. In most studies, sound pressure was used for the determination of sound power. In the present study sound power was determined from sound intensity. The discrete point method with 10 measurement points on a hemisphere and 80 to 300 points on rectangular surfaces was used for the evaluation of sound power. The measurements were carried out for three instruments: the bassoon, the upright piano, and the snare drum. The instruments were played by professional musicians in a typical practice room. The music samples played on the bassoon and on the piano were scales and excerpts of music. The samples played on the snare drum were rhythmic sequences and tremolo sounds. The results demonstrate that sound intensity is a reliable method for the estimation of sound power level of musical instruments played in a small practice room. The paper discusses the influence of the measurement method, the test room acoustics, the playing technique, the choice of music sample, and the player’s stability on the uncertainty of measurement. [This work was supported by Grant No. 7-T07B-043-10 from the State Committee for Scientific Research.]
A tale of three violins: Comparison of radiation mechanisms measured with near-field acoustic holography
Authors: Wang Lily M. , Burroughs Courtney B.
Published: July 2002

Complex pressure measurements have been made on multiple planes surrounding three violins, while bowed with a mechanical bowing apparatus: a Scherl & Roth student instrument, Hutchins’ SUS295, and a Hutchins’ mezzo-violin. The data are used with a multiplanar near-field acoustic holography (NAH) algorithm to reconstruct the surrounding radiated sound fields of each instrument, including pressure and intensity from the surface extending out to the far field. Areas of significant radiation on each violin are identified, compared, and related to structural differences between the instruments. [Work supported by NSF Graduate Research Fellowship, Lucent Technologies/Bell Laboratories GRPW Grant, and AAUW Selected Professions Dissertation Fellowship.]
The acoustic analysis of reinforced harmonics
Authors: Edgerton Michael E. , Bless Diane , Thibeault Susan , Fagerholm Margaret , Story Brad
Published: July 2002

The relationship between articulation and harmonic reinforcement among five overtone singers was mapped from analyses of acoustic signals and radiographic data, with much variation found. A dominant feature of reinforced harmonics (R.H.) is the isolation of one harmonic while attenuating its neighboring harmonics. This is achieved by laying regions of high pressure (formants) over the framework of the harmonic series. These formants are manipulated to reduce the bandwidth which boosts the selected harmonic and dampens adjacent harmonics. LPC Fourier analyses contrasting normal speech to R.H. production demonstrated that bandwidth narrowed significantly (at 3 dB down) from 150 to 20 Hz, while center bandwidth (and selected harmonic) during R.H. increased amplitude by 34 dB, and damping of adjacent harmonics decreased amplitude by 30 dB. Next, bandwidth was measured by applying an artificial larynx to two subjects who showed bandwidth narrowing from 258 to 18 Hz; center bandwidth frequency increased by 12 dB and damping of adjacent harmonics by 14 dB. Last, R.H. production was modeled using a resynthesis program to allow manipulation of formant frequencies, amplitudes, and bandwidths of normal phonation to achieve R.H., while R.H. production was manipulated to attain normallike phonation.
Jazz drummers’ swing ratio in relation to tempo
Authors: Friberg Anders , Sundstroem Andreas
Published: July 2002

It is common in jazz to perform consecutive eighth notes as long–short patterns. Drummers’ performances on the ride cymbal of these long–short patterns were measured on excerpts taken from commercial recordings. It was found that the ratio between the long and the short note (swing ratio) varied substantially with tempo. At slow tempi the swing ratio was as high as 3.5:1, while at fast tempo it reached 1:1. The absolute duration of the short note, however, was found to be constant (about 100 ms) for medium to fast tempi. This suggests the existence of a limit on tone duration and also that the short note may function as an up-beat to the following note. A similar relation between swing ratio and tempo was also observed in a production experiment where listeners adjusted the swing ratio of a computer-generated performance.
Effect of timing variations on the categorical perception of musical rhythm patterns
Authors: Papadelis George , Papanikolaou George
Published: July 2002

Musical rhythm patterns are mainly characterized by durational relationships between their different time levels. As written rhythmic values imply, these relationships are graphically represented by simple integer ratios and that fact reflects a basic criterion for their perceptual encoding into discrete perceptual categories. Temporal analysis of any musical performance reveals that deviations from accurate metronomic timing is a very common practice among musicians and these deviations are perceived either as musical expressivity, or just as inaccurate performance of rhythm. The degree of deviation also affects the categorization of a certain rhythmic pattern. In the present study, a mapping of the perceptual space between adjacent rhythm categories was attempted, through a systematic timing variation of simple rhythm patterns, as well as the investigation of the perceptual magnet effect [P. Iverson and P. Kuhl, J. Acoust. Soc. Am. 97, 553–562 (1995)] at various musical tempi. Sixteen musicians participated in identification, goodness rating, and discrimination tasks. The relative strength of the prototype within each category was determined, as well as the change in the position of category boundaries in different tempi. Discrimination graphs were finally plotted and comparisons were performed between the positions of discriminability peaks and the related category boundaries.
Simulation of vacuum-tube amplifiers
Authors: Sapp Markus , Becker Jörg , Brouër Claas
Published: July 2002

Vacuum tube amplifiers are said to add ‘‘warmth’’ to an audio signal. They are used for subtle signal modifications as well as soft sounding distortions. But since most signal processing for musical recordings is performed in the digital domain, algorithms for a digital simulation of the desired effects produced by tube amplifiers would be useful. In the work presented here two approaches are followed. The first is based on the simulation of the characteristic transfer curves (waveshaping) combined with a frequency response correction. In the second, more complex one, the complete electronic circuit is simulated using an analog behavioral model of the vacuum tube. The output signals of the simulations are compared to the output of a real vacuum tube preamplifier analytically as well as within listening tests. The computational cost for a real-time implementation of the different algorithms is set into relation to the simulation results.
Limiting technology for combustion turbine power plants—how quiet can we make them?
Authors: Brittain Frank , Kirschfink Norbert , Parzych David
Published: July 2002

There is a demand for ever quieter combustion turbine power plants. As the community noise limit goes down, the cost and impact of the noise controls on the design increase. At some point, the noise controls significantly affect the size of the plant and the cooling device (cooling tower or air-cooled condenser). This is caused, in part, by silencer self-noise and because the octave bands controlling the A-weighted noise level decrease to include the 31.5-Hz band. While meeting almost any noise limit is considered technically feasible, the cost and impact on the design can become excessive. Several low-noise limits that have been met by several existing plants are reviewed. A specific combined-cycle power plant is used to investigate the impact of lowering noise limits. An indoor plant is used, because adequately controlling the multitude of sources for an outdoor plant is not considered feasible. The noise controls that are expected to be needed to meet various A-weighted noise limits are identified. Limits at which costs for these become excessive or start affecting the size of the plant and cooling device are estimated. While no cost estimates are provided, several definitions of excessive are made.
Measurement methodology for sound power levels of industrial noise sources
Authors: Bruyninckx Willy , Kelders Luc , Lauriks Walter , Thoen Jan
Published: July 2002

For the purpose of noise impact evaluation of industrial noise sources, one often needs reliable values of the sound power level LRW of an entire plant or large parts of it, where one quite often has to do with a large number of distributed noise sources. A methodology will be presented which is based on sound-pressure level measurements at a sufficiently large number of points distributed around a plant, resulting in noise contour maps allowing the calculation of LRW. The method has been compared with other approaches in a well-controlled free field setting of a simple combination of noise sources.
A spatial analysis of topographic and vegetation factors affecting the propagation of highway traffic noise
Authors: Bennett John C.
Published: July 2002

This paper presents the results of a spatial analysis of measured noise levels with environmental factors derived from a geographic information system (GIS). The more traditional methods used by the Federal Highway Administration (FHWA, 1978 and 1982) are range limited and not suitable for large-scale topographic variation near a freeway noise source. Over 30 field measurements of noise levels were collected along a freeway in San Diego County that included a heterogeneous landscape of mesas and valleys. A digital elevation model and a vegetation map of the study area have been incorporated into a GIS to evaluate a suite of environmental factors. Topographic features limit the line-of-sight analysis traditionally used in the past and new variables of landscape ‘‘ensonification’’ and slope configuration are shown to bridge part of that gap in this study.
Prediction of traffic noise control by forests using a three-dimensional GFPE model
Authors: Barriere Nicolas , Gabillet Yannick
Published: July 2002

To predict outdoor sound propagation through forests, multiple natural effects have to be taken into account such as ground effects and meteorology that vary along the path of propagation as well as horizontal trunk scattering effect. It is of great interest, for engineering applications, to know the weight of the different effects. A previous work with a two-dimensional model has shown the interest of an infinite forest band [N. Barriere and Y. Gabillet, Proceedings of Euro-noise 98, Munich]. The prediction method is based on a theory for fast calculation of the parabolic equation [K. E. Gilbert, J. Acoust. Soc. Am. 94, 2343–2352 (1993)], and has recently been developed at the CSTB for a three-dimensional sound propagation. This method allows prediction of noise attenuation due to a forest band of finite length with its corresponding, ground, meteorological, and scattering effects. Results show the great interest of forests on traffic noise control.
Comment on “Enhancement of the transient-evoked otoacoustic emission produced by the addition of a pure tone in the guinea pig” [J. Acoust. Soc. Am. 104, 344–349 (1998)]
Authors: Ren Tianying , Nuttall Alfred L.
Published: July 2002

Recently, [J. Acoust. Soc. Am. 104, 344–349 (1998)] reported a pure tone acoustic enhancement of the transient-evoked otoacoustic emission (TEOAE). They state that their data conflict with the existing theory that the TEOAE represents a one-to-one frequency response to stimulus component frequencies. Therefore, they interpreted the observed enhancement as a result of a complex interaction on the basilar membrane of hypothetical intermodulation distortion products. This Letter reviews recent data and proposes an alternative interpretation that the enhancement results from an impedance discontinuity at the pure tone characteristic frequency (CF) location. TEOAEs at frequencies below this tone are reflected from its CF location rather than from their own CF locations.
The acoustical properties of large studios
Authors: Ivancevic Bojan , Domitrovic Hrvoje , Fajt Sinisa
Published: July 2002

New studios for Croatian Radio-Television were planned with the starting user conditions. The background noise level must fulfill N15 or N20 criteria. Demanded reverberation time for a 400 m2 radio studio is Tr=0.6 s, for TV studios of the same area, Tr=0.7 s, and for the 1000 m2 TV studio, reverberation time is 1 s. Since all three studios with built-in control rooms were built in an urban area with excessive noise, it was necessary to obtain very good protection against noise, and at the same time, using the same elements, achieve the desired reverberation time. Special attention was given to flutter echoes and achievement of diffuse sound fields. The inner shell construction was built up using special acoustical building blocks. Besides using double walls, the broad analysis was done in order to avoid any possible problem in sound insulation. The achieved coincident frequency for walls and floors is 7.6 Hz, and this is quite satisfactory. The computer simulation was used during the design of internal sound-absorbing and reflecting surfaces. Described solutions take care of all user demands, as well as the usage particularities of such spaces. The measurements of relevant acoustical parameters were done after the building was finished.
Factors in the performance of systems for the production of virtual acoustic environments
Authors: Nelson P. A. , Kirkeby O. , Kahana Y. , Hamada H.
Published: July 2002

This paper will describe recent progress in the development of systems designed to present, at the ears of a listener, the signals necessary to produce the illusion in the listener of the existence of a ‘‘virtual’’ source of sound. Attention will be restricted to systems which use a relatively small number of loudspeakers whose input signals are determined by optimal processing of the signal to be associated with the virtual source. It will be demonstrated that the form of the sound field produced is crucial to the success of such systems. Emphasis will be given to a system that uses two very closely spaced loudspeakers to transmit a particular form of sound field which most easily produces the iteraural time delay associated with a given virtual source position. Results will be presented of subjective experiments and of numerical simulations of the sound field. Factors determining the system performance will be discussed, particularly with regard to front–back confusions and individual differences in head related transfer functions. It will also be shown that systems using four louspeakers can significantly reduce the degree of front–back confusion even when assuming a relatively simple model of the listener’s HRTF.
Comparison of virtual sound source positioning with amplitude panning and Ambisonic reproduction
Authors: Strauss Holger , Buchholz Jörg
Published: July 2002

Different kinds of panning algorithms can be applied to render virtual sound sources in pantophonic or periphonic loudspeaker setups. One popular method is constant energy amplitude panning, where only the loudspeakers nearest to the virtual sound source are used for sound reproduction. Another well established reproduction method is Ambisonics, which uses the spherical harmonics theory in combination with knowledge from psychoacoustics to render virtual sound sources by simultaneously using all loudspeakers in a setup. Technical details and background information on both algorithms are given, and technical implementation issues (e.g., required computational effort) are discussed. A series of psycho- acoustic listening tests was performed to evaluate virtual sound source localization with both reproduction methods for a six-loudspeaker pantophonic and an eight-loudspeaker periphonic setup. Localization accuracy and the subjects’ reaction time were measured. The results show that the localization blur does not depend on the virtual sound source position for Ambisonics, while amplitude panning causes a high localization blue for virtual sources between the loudspeakers and a low localization blur for images close to the loudspeakers. Similar results were found in tests carried out to ascertain how dependent the reaction time is on the virtual sound source position.
New numerical simulation tool for the design of electromagnetic transducers
Authors: Schinnerl Michael , Landes Hermann , Kaltenbacher Manfred , Lerch Reinhard , Schöberl Joachim
Published: July 2002

Due to the high geometric complexity and the interaction of different physical field types, like mechanical displacement, acoustic pressure, and magnetic induction, the design of electromagnetic transducers is a challenge for the developers. Since the fabrication of prototypes and experimental-based design is a lengthy and costly process, needs for appropriate numerical simulation tools arise. In this paper, a simulation program is presented which is especially adapted to multifield problems like the one described above. First, the description of the underlying physical fields (acoustic, mechanical, and magnetic) with partial differential equations (PDE) and their coupling is reported. The solution of these PDEs is based on either finite-element methods (FEM), boundary-element methods (BEM), or coupled FE-BE methods, depending on the considered problem type. However, these numerical techniques yield to long computer time, especially in the 3-D case. Therefore, a multigrid solver has been developed which enables considerably faster solutions of the presented numerical tasks. Finally, two application examples, the calculation of an EMAT (electromechanical acoustic transducer) in transmitting and receiving mode and the prediction of the sound field of a vibrating machine part, show the reliability of the simulations.
A comparison of wind instrument time-variant spectra
Authors: Beauchamp James W.
Published: July 2002

Musical instruments can be compared in terms of the sound spectra that they produce. First, they may be compared in terms of spectral envelope, which allows an observation of change of amplitude with respect to frequency. Moreover, the spectra change with respect to time, usually with a burst of low-amplitude, broadband spectrum initiating the attack phase, followed by a gradual increase of spectral centroid. The so-called ‘‘steady state’’ may actually include some interesting changes, depending on performance expressivity. The final decay phase is perhaps the least manipulable by the performer, and is generally characterized by a gradual lowering of the centroid as the amplitude decreases. A centroid-indexed family of spectral envelopes based on data for several tones with different fundamental frequencies gives a good average picture of the spectra for a particular instrument. However, this representation fails to take into account special details such as pitch-dependent and harmonic-dependent spectral features. While such special details are least important for the brass, they are very important for the woodwinds, and several features of the clarinet, oboe, saxophone, and flute will be illustrated and compared.
Differences between cylindrical and conical reed instruments
Authors: Dalmont Jean-Pierre
Published: July 2002

Most investigations on reed instruments have been carried out on clarinet probably because this is the most common reed instrument. However, the extrapolation of the results to conical instruments is risky. The fact that internal clarinet spectrum does not contain even harmonics is not the only difference. For example the transfer function between input and output pressure is very different: this makes a saxophone, for lower notes, much louder than a clarinet. On the other hand, application of linear analysis near the threshold of oscillation to conical reed instruments is limited: because of the inverse bifurcation, small oscillations do not exist for conical reed instruments. This is confirmed by the observation that it is clearly much easier to play pianissimo with a clarinet than with a saxophone or an oboe. The saturation mechanism seems also to be different for conical and cylindrical instruments. On a saxophone the saturation occurs at a lower mouth pressure because of the existence of more than one regime at the fundamental frequency. All these aspects will be discussed with reference to simplified physical models and illustrated by experiments using artificial blowing. [This work has been done in collaboration with C. J. Nederveen, J. Gilbert, and J. Kergomard.]
Blowing and breathing in the woodwinds: Generation and control of input parameters
Authors: Fuks Leonardo
Published: July 2002

Wind instruments, being more than an extension to the performer’s body, are strongly connected to the respiratory system. Such a relationship permits a multiple and fine control of input parameters, enabling wide ranges of tone variation and expressiveness, but also imposes constraints concerning muscular demands for aerodynamic energy supply, breathhold times, and lung capacity, among others. The present investigation combined physiological and acoustical approaches, focusing on reed woodwinds under naturalistic playing conditions. Airflow and blowing pressure used by professional players were shown to be systematic in rehearsed tasks and to depend on pitch and dynamic level, instrument type, and reed characteristics, suggesting that musical training involves the incorporation and adjustment of the mechanical properties of the system to the musician’s technique. Blowing pressure sensation was assessed by a psychophysical method, showing a quasi-linear behavior. Respiratory movements, decomposed into rib cage and abdominal contributions, have been recorded and analyzed during performance by an inductive plethysmographic method, providing insights to the musicians’ use of the respiratory system. The ever-changing pulmonary air composition was shown to affect fundamental frequency to some extent. [The research work presented in this paper was carried out at KTH, Stockholm, and was sponsored by a Brazilian CAPES scholarship.]
Modeling of traffic noise pollution with neural networks
Authors: El Mallawany A. I. , Abdallah M. I. , Abd El Gawad M.
Published: July 2002

A neural-network-based model for evaluating and predicting the mean energy and the noise pollution level of the traffic noise in the city of Cairo is developed. A comparison between the different prediction models was carried out, and the neural model was shown to be associated with minimum error. In conclusion, the neural model can be used to make systematic predictions of noise level, and provide planners with some suitable means for formulating legal regulations for urban noise pollution.
Low noise road surfaces in the Federal Republic of Germany. State of the art
Authors: Ullrich Siegfried
Published: July 2002

The texture of road surfaces influences the rolling noise of cars and trucks. High texture depths cause high noises by the incitation of tire tread vibrations. Low depths favor the generation of aerodynamic noise sources in the tire tread, for example, air pumping. This type of noise can be suppressed by open surfaces on wearing courses with high void contents. As coatings with low diameter gritting material on worn concrete showed, air pumping noises can also be reduced on closed road surfaces. But trials on gussasphalt turned out to be not very successful thus far. The greater part of the rolling noise on such surfaces is apparently caused by a waviness with wavelengths between 3 and 6 cm, which could be caused by the embedment of chippings. A similar reason seems to be responsible for the limited noise reduction on concrete surfaces with burlap finishings. Measurements at a 5.5-m-diam internal drum with sandpaper surfaces and different grittings on asphalt concrete allows the hope that the rolling noise of car tires can be reduced by several decibels, if road surfaces are gritted by small diameter stones from a small fraction without oversizes.
Analytic expressions for the far-field transition distance for localized waves
Authors: Chambers David , Gautesen Arthur
Published: July 2002

Expressions estimating the distance to the near-field/far-field transition point are obtained for the problem of a localized wave emitted from a circular aperture. These expressions are based on bounding the error be-tween the exact diffraction integral and its far-field approximation. The resulting formulas involve integrals over the normal velocity distribution within the aperture. They can be evaluated for both narrow- and broadband pulses. For the special case of a single frequency with uniform spatial weighting, the predicted transition distance is the standard Rayleigh range. The expressions are evaluated for a number of localized waves and compared with numerical simulations of the full fields on axis.
Universal local state equation for description of exponential and resonant relaxation at sound-wave propagation
Authors: Maksimov German A. , Larichev Vladimir A.
Published: July 2002

The generalization of the thermodynamic theory of internal parameters [L. I. Mandelshtam and M. A. Leontovich, Sov. Phys JETP 7, 438–444 (1937)], which leads to a description of medium response in terms of exponential relaxations, is presented in the report for the case of resonant relaxations. The basis of this generalization is an alternative formulation of Onsager’s principle of kinetic coefficient symmetry for values of different symmetry at time inversion. Using this fact, the generalized local linear state equation is derived for medium holding in the vicinity of thermodynamic equilibrium. It is shown that the developed state equation is an exhaustive one for the description of linear response of medium in the framework of local models. The variety of dispersive and dissipative properties, which is described by the generalized state equation, is demonstrated for the case of the single process of resonant relaxation. It is shown that all the existing local models of linear response of medium are the particular cases of the generalized state equation. [Work supported by ISTC in Moscow.]
Current progress and problems in atmospheric acoustics
Authors: Kallistratova Margarita A.
Published: July 2002

The problems of basic and applied research in the area of atmospheric acoustics are elucidated. Main results are reviewed from the theoretical and experimental investigations of audible and infrasonic wave propagation in the atmosphere which were carried out during the last few decades. The role of the acoustic techniques in studying the structure and dynamics of meteorological fields is discussed. The data are given on a practical use of echo sounders (SODARs) and the systems of radio acoustic sounding (RASS) in monitoring the atmospheric boundary layer parameters. Some other practical applications of the acoustic methods in the atmosphere are considered. A number of problems are formulated, whose solution is necessary for a further advancement of atmospheric acoustics and for the use of the acoustic techniques in studying the lower and middle atmosphere. [Work supported by INTAS through Grant No. 96-1869 and RFBR through Grant No. 96-05-65741.]
Investigation of thin film shear modulus from spatially resolved measurement of Love wave velocity
Authors: Behme G. , Chilla E. , Froehlich H.-J.
Published: July 2002

The investigation of transversely polarized surface acoustic waves (SAWs) is strongly hindered by their missing out-of-plane oscillation component because commonly applied measurement techniques are based on access to that component. Nevertheless, these waves are very interesting for accurate extraction of thin film elastic properties as well as for SAW devices and sensor applications. In this contribution we report a new development of spatially resolved SAW analysis. The scanning acoustic force microscope (SAFM) has been extended in its sensitivity to in-plane polarized waves. A nonlinear mechanism could be found that couples the in-plane surface oscillations to torsional cantilever vibrations. This mechanism is used to transfer the rf SAW signals to kHz frequencies and to measure the SAW phase velocity. The new technique has been applied to study the Love wave propagation in the layered system fused silica on ST-cut quartz. The phase velocity could be derived from measurements of the phase shift along a propagation distance in the micron range. The obtained velocities show a clear spatial variation at elastic changes within the investigated surface. This is used to investigate the shear modulus of the layer material by numerically solving the inverse problem of SAW propagation.
Vibration response of perforated thin plates
Authors: Allemon Pascal , Hazebrouck Robert , Breaban Florin , Deffontaine Andre
Published: July 2002

The vibration behavior of perforated metal thin plates is studied here. In order to highlight specific phenomena, a rather simple experiment has been researched. While covering a large range of frequencies, a sinusoidally varying point force has been applied to a free boundary circular plate at its midpoint. In that case, the plate vibrations obtained are of an axial symmetry type; for that geometry the first symmetrical modes only are excited. The driving-point mechanical impedance is measured. For a plane stress problem, the perforated plate vibration response is evaluated by assimilating perforated plates to equivalent solid thin plates whose effective elastic constants are determined with the help of Meijers’ theory [Ph.D. thesis, University of Delft, Holland, 1967]. First of all, the numerical simulation is validated by comparing theoretical impedance results to those obtained experimentally. Then, the use of double exposure holographic laser interferometry permits one to verify that modes observed on photographs correspond in general to theoretical ones. Initially developed for the study of perforated plate static behavior, that theory can certainly be used for the dynamic study of various geometry perforated plates.
An acoustic wave’s transmission through an elastic cylindrical shell and investigation of its parameters
Authors: Oboznenko Igor L. , Oboznenko Elena I. , Samonenko Victor
Published: July 2002

The method of elastic parameter’s estimation of cylindrical shells and surroundings is proposed. The method is based on measurement of the transmitted wave by the calibrated ultrasonic hydrophone inside of the elastic shell. Metallic, plastic, rubber, and blood shells are discussed. It has been demonstrated that acoustical pressure inside the shell, for frequencies lower than the frequency of mechanical resonance, does not depend on the frequency, but rather depends on mechanical parameters like Young’s modulus, Poisson coefficient, density, and geometry (median radius and thickness) of the shell. Irradiation in various wave reference angles in relation to normal is applied for determination of the shell and those resonances also are different (10–1000 times in frequency). Transmitted waves have been calculated for both the elastic plate model and elastic shell models of different geometry. The pass wave field significantly depends upon shell material. Its resonances also are different (10–1000 times in frequency). It has been demonstrated that up to 60% higher accuracy in calculations of the pressure of transmitted waves can be achieved when the exact solution is used in comparison with the approximate solution for the elastic plate model. There is a good correlation between computational and experimental results.
The normalized correlation: Accounting for NoSπ thresholds with Gaussian and ‘‘low-noise’’ masking noise
Authors: Bernstein Leslie R. , van de Par Steven , Trahiotis Constantine
Published: July 2002

Recently, both Eddins and Barber [J. Acoust. Soc. Am. 103, 2578–2589 (1998)] and Hall et al. [J. Acoust. Soc. Am. 103, 2573–2577 (1998)] demonstrated that ‘‘low-noise’’ noise produced more masking of antipha- sic tones than did Gaussian noise. Eddins and Barber could not account for this finding by considering changes in the normalized interaural correlation produced by adding antiphasic signals to the diotic maskers. They calculated the normalized correlation subsequent to half-wave, square-law rectification and low-pass filtering. That approach was used successfully by Bernstein and Trahiotis [J. Acoust. Soc. Am. 100, 3774–3784 (1996)] to account for binaural detection performance as a function of frequency. In this presentation, it will be shown that one can account quantitatively for thresholds obtained with ‘‘low-noise’’ noise and with Gaussian noise by incorporating a stage of compression prior to rectification and low-pass filtering. The compressive function employed produces effects similar to those measured along the basilar membrane. Adding compression allows one to account quantitatively for the low-noise-noise and Gaussian-noise thresholds while maintaining a highly successful quantitative account of the Bernstein and Trahiotis data. Thus the normalized correlation remains a very useful metric that accounts for a wide variety of binaural detection data. [Work supported by NIH-DC-02103 and NIH-DC-00234.]
Interaction in the perceived lateralization of two sounds having different interaural time differences
Authors: Kashino Makio
Published: July 2002

Perceived lateralization of a 0.25-s, 500-Hz test tone was measured in the presence of a 1-s, 1/3-oct band noise centered at 500 Hz. The interaural time difference (ITD) of the noise was fixed at zero, and that of the test tone was varied trial by trial in the range of −500 to 500 μs (negative sign meaning that the left channel signal leads). The test tone started at various temporal positions between 1.25 s before the onset of the noise and 1.25 s after the offset of the noise. Subjects indicated perceived lateral position of each test tone on a left–right axis. It was found that perceived lateralization of test tones presented before the noise was approximately a linear function of ITD. On the other hand, that of test tones presented concurrently with the noise was shifted away from the noise, depending on the ITD relationship between the tone and noise. This contrast effect remained even 1.25 s after the offset of the noise, replicating the previous finding of localization aftereffect [M. Kashino and S. Nishida, J. Acoust. Soc. Am. 103, 3597–3604 (1998)]. These results suggest rapid gain control of ITD-selective units, probably due to inhibitory mutual interactions.
Audiometry for binaural hearing
Authors: Stettbacher Juerg M.
Published: July 2002

As traditional audiometry is based on very artificial conditions, the results in many cases do not coincide with practical experience in everyday life. The accuracy of the so-called useful hearing loss in legal cases has been questioned many times. A new procedure is therefore proposed. It is based on real acoustic free-field situations reproduced over a loudspeaker array. The sound field is controlled by using digital signal processing. The kinds of measurements and stimuli have been derived from a model of cognitive hearing. Different measurement types consider different aspects of natural hearing. An assessment model helps to combine the results from various measurements to a single index. It directly describes the useful hearing loss with respect to everyday hearing. It is not intended to serve as a diagnostic tool, but as a useful and accurate measure in legal or insurance cases. Results from first experiments will be presented.
Reflection and transmission at an attachment half-space
Authors: McCoy John J.
Published: July 2002

The reflection and transmission at the connection of an attachment half-space to a core half-space that contains a source of mechanical energy and one or more scatterers is considered. Both the attachment and core half-spaces have property measures that are invariant with distance from the connection plane, but may vary with position in planes parallel to the connection plane. At issue is the construction of a reflection operator that both applies at the connection plane and is expressed in terms of operators that separately describe the responses of the half-spaces at the connection plane, i.e., of impedance operators that can be written for the half-spaces in isolation. A successful construction is achieved for a connection that spans the entire connection plane. For this scenario, the construction achieved is easily identified as a generalization of a ‘‘nonreflecting boundary condition,’’ in that the reflection operator obtains no reflection for any incident field, if the impedance operators for the two half-spaces are the same. The achieved construction is next demonstrated to not apply for a connection that spans only a finite portion of the connection plane, i.e., an aperture. For this second scenario, pressure release conditions apply for both half-spaces at points on the connection plane that are outside the aperture. The reason for the failure is the impedance operators that apply in the context of a full-plane connection are not uniquely described for a connection that applies to a finite aperture. That is, mobility operators that can be identified with the DtN map obtained on factoring a two-way description of the fields in the half-spaces into separated one-way descriptions, are uniquely defined regardless of the connection region. However, the inverse of this mobility operator requires connection across the entire plane. The implication of this demonstration for more complex experimental scenarios in both propagation modeling and structural acoustics is discussed.
Control of oral closure in stop consonants with different closure durations
Authors: Lofqvist Anders
Published: July 2002

This study examined lip and tongue kinematics in the production of stop consonants that differ in the duration of the oral closure. The linguistic material consisted of Swedish words that contained either a short vowel followed by a long stop, or a long vowel followed by a short stop. Articulatory movements were recorded using a magnetometer system. The average closure duration for the long stops was 100–150 ms, about twice the closure duration for the short stops. Preliminary results for one subject suggest that the lip closure for the long stop was produced with an upward movement trajectory of the lower lip that reached a higher position during the closure than the trajectory for the stop with a short oral closure. As a result, there was more mechanical interaction between the upper and lower lips, with the lower lip pushing the upper lip upward. For stops articulated with the tongue, the tongue moved along a longer path during the stop closure for the long than for the short stops. The average velocity of the tongue movement during the stop closure was lower for the long than for the short stops. [Work supported by NIH.]
No role for syllables in English speech production
Authors: Schiller Niels O.
Published: July 2002

Four experiments investigating the role of the syllable in English speech production are reported. In experiment 1, pictures had to be named. Pictures were preceded by visually masked primes that corresponded to the first syllable (CV or CVC) of the picture name or were one segment shorter or longer than its first syllable. Naming latencies were shortest in the CVC priming condition independent of the syllabic structure of the target. Experiment 2 replicated this effect with words. In experiment 3, all words could be grouped into pairs and overlapped in the first three letters while being different in syllable structure (e.g., se.cret–sec.tion). The same result as in the previous two experiments was obtained. Experiment 4 was added to show that the size of the priming effect increased with longer prime exposure duration. The results contradict the syllable priming hypothesis but are in line with the segmental overlap hypothesis.
Two kinematic models of the human tongue from MRI data
Authors: Davis Edward , Stone Maureen , Douglas Andrew
Published: July 2002

Two models of tongue motion were developed and applied to tagged cine MRI data of the human tongue during movement from the consonants /k/ and /s/ to the vowels /i/, /a/, and /u/, in three sagittal slices (L, M, R). The first model, a global kinematic model, utilized surface edges of the tongue perimeter observed within the images. The model focused on decomposing tongue surface movements into the following mechanical deformations: rigid-body motions (translation and rotation) and homogeneous stretch and shear. The global model indicated that coarticulatory effects occurred in both vowels and consonants. Deformation was greatest from /k/ to /a/, and was composed primarily of horizontal expansion, vertical compression, and vertical lowering. Asymmetries were observed in the deformation patterns. The second model, a local kinematic model based on the motion of approximately 40 tag intersection points within the tongue, utilized the same low-order polynomial basis functions that are applied in finite-element analysis, within a finite number of small discreet regions. Specifically, the local model quantified the principal strains within discrete regions inside the tongue during consonant–vowel deformations. The local model was able to illuminate detail of local muscle activity within an individual syllable.
Challenges facing 3-D audio display design for multimedia
Authors: Begault Durand
Published: July 2002

The challenges facing successful multimedia presentation depend largely on the expectations of the designer and end user for a given application. Perceptual limitations for distance, elevation, and azimuth sound source simulation differ significantly between headphone and cross-talk cancellation loudspeaker listening and therefore must be considered. Simulation of an environmental context is desirable but the quality depends on processing resources and control over interaction with the host acoustical environment. While techniques such as data reduction of head-related transfer functions have been used widely to improve simulation fidelity, another approach involves determining thresholds for environmental acoustic events. Psychoacoustic studies relevant to this approach are reviewed in consideration of multimedia applications. [Work supported by San Jose State University–NASA Ames Research Center Cooperative Grant NC 2-327.]
Interactive acoustic modeling of complex environments
Authors: Funkhouser Thomas , Carlbom Ingrid , Elko Gary , Pingali Gopal , Sondhi Mohan , West James
Published: July 2002

A primary challenge in acoustic modeling is computation of reverberation paths from sound sources fast enough for real-time auralization. This talk describes a system that uses precomputed spatial subdivision and ‘‘beam tree’’ data structures to enable real-time acoustic modeling and auralization in interactive virtual environments. The spatial subdivision is a partition of 3-D space into convex polyhedral regions (cells) represented as a cell adjacency graph. A beam-tracing algorithm recursively traces pyramidal beams through the spatial subdivision to construct a beam tree data structure representing the regions of space reachable by each potential sequence of transmission and specular reflection events at cell boundaries. From these precomputed data structures, high-order specular reflection and transmission paths can be computed at interactive rates to spatialize fixed sound sources in real-time as the user moves through a virtual environment. Unlike previous acoustic modeling work, this beam-tracing method: (1) supports evaluation of reverberation paths at interactive rates, (2) scales to compute high-order reflections and large environments, and (3) extends naturally to compute paths of diffraction and diffuse reflection efficiently. A system based on this method is being used to develop interactive applications in which a user experiences a virtual environment immersively via simultaneous auralization and visualization.
Echo cancellation for multimedia applications
Authors: Sondhi M. Mohan
Published: July 2002

In any attempt at creating natural audio-visual communication between two remote locations, acoustic echoes are generated due to the coupling between the loudspeakers and microphones in each conference room. If ignored, these echoes can completely disrupt audio communication. Teleconferencing systems today use single-channel (monophonic) speech communication. In order to recreate a natural ‘‘presence’’ in the remote location, multichannel (stereo) communication is desirable, and will no doubt be the preferred future mode of teleconferencing between two spatially separated groups of participants. A third type of conference is a desktop conference between several locations. In such applications the remote participants are displayed on a screen, and synthesized stereo may be used to simulate sound coming from the direction of each of them. The talk will discuss the characteristics of echoes in all such situations, and the solutions currently available to deal with them. The solutions can also be useful for eliminating disturbances other than echoes, e.g., the sound from a car’s stereo system picked up by the transmitter of a hands-free telephone.
Automatic classification of wideband acoustic signals
Authors: Dufaux Alain , Besacier Laurent , Ansorge Michael , Pellandini Fausto
Published: July 2002

Until now, very few contributions were published in the field of wideband acoustic signal recognition, especially for handling impulsive noise signals such as glass breaking, detonations, or door slams, as encountered in security applications, where the signals are highly nonstationary and composed of higher frequency components. This paper shows how the audio alarm recognition problem can efficiently be tackled using either pattern recognition methods relying on Bayes classifiers, or on artificial neural networks (ANN). After extraction of filterbank coefficients in the acoustic analysis module, typical feature vectors are achieved from the concatenation of k consecutive signal frames, in order to exploit dynamic temporal information. The redundancy induced by this dynamic modeling requires a reduction of the feature space dimension, performed with a conventional principal component analysis. The performance of both systems is evaluated experimentally (7000 tests) for the classification of three types of noise events (glass breaking, door slams, and stationary noises). The score of correct classification reaches 99% for the statistical approach, and 98% for the ANN-based system. It is further noticed that errors do not occur on the same signal files for both methods. Thus robustness enhancement can be reasonably expected, when using hybrid methods or fusion strategies.