bluuebunny/crossref_metadata_2025 huggingface dataset:
Benchmarking virtual backscatter in parabolic evolution
Authors: Smith Eric
Published: July 2002

A perturbative method has recently been presented for improving parabolic equations to account for intermediate backscattering induced by slow range dependence of the acoustic index of refraction. In realistic cases, it yields solutions to the full wave equation from a truly one-way algorithm, and generalizes a number of special results of this kind which have been known for many years. Numerically validating the method, though, involves certain unique difficulties, because the physical corrections it introduces to local evolution are accompanied by medium dependence of the projection operator that defines the input and output wave fields. A method for benchmarking is presented here, which permits numerical isolation of range-local effects from the endpoint projections, and also from particular details of the choice of one-way model used. To illustrate that the physical content of the new terms can be sensibly validated in this manner, an exact solution is considered, where they are shown to correct the “idealized” one way equation on which current parabolic algorithms are based. This solution is used to illustrate the relation of energy conservation to one-way evolution in the presence of slow range dependence. In particular, it demonstrates that energy conservation alone is not a sufficient requirement to produce correct parabolic evolution, even when it pertains.
Does air absorption modify ground effect?
Authors: Makarewicz Rufin
Published: July 2002

Sound propagation in the atmosphere involves many wave phenomena. Some of them, e.g., scattering on turbulence, and ground effect, interact with each other. Using the equation of state for relaxation in a binary mixture of gases, such as oxygen and water vapor, it is shown that the interaction between the ground effect and air absorption can be neglected.
What should be the goals of cochlear modeling?
Authors: Chadwick R. S.
Published: July 2002

The recent thought-provoking paper by de Boer, “Classical and nonclassical models of the cochlea,” not only deserves comment on its own right, but more generally it should cause cochlea modelers to reflect on their ultimate goals. While his claim that a “classical” model can be found that has the same response as a “nonclassical” model is mathematically correct for linear, single degree-of-freedom models, this paper argues that his algorithm produces a “nonrealizable” and hence less useful model. Also, his assertion of uniqueness needs to be qualified.
Response to “Comments on ‘The attenuation and dispersion of sound in water containing multiply interacting air bubbles’ ” [J. Acoust Soc. Am. 102, 1239–1241 (1997)]
Authors: Feuillade C.
Published: July 2002

A response to the comments made by Zhen Ye in a Letter to the Editor about the paper “The attenuation and dispersion of sound in water containing multiply interacting air bubbles [J. Acoust. Soc. Am. 99, 3412–3430 (1996)]” is presented. It shows that, while Ye has made some useful observations, his comments fail to bring the theoretical assumptions of the original paper into question.
Temporal backward planar projection of acoustic transients
Authors: Clement G. T. , Liu R. , Letcher S. V. , Stepanishen P. R.
Published: July 2002

Backward projection schemes use data in front of a transmitter to reconstruct a field at closer points. Existing techniques have concentrated propagating radial and temporal information along constant planar cross sections in front of a planar source. This approach requires a careful consideration of evanescent waves, as the transfer function used to backproject in space causes evanescent wave solutions to increase exponentially with the projected distance. Erroneous signals may result from exponentially increasing noise, experimental error or roundoff error. A method is presented that is designed to work with imaging methods that record three-dimensional spatial data at constant times. Several widely used optical methods are of this type. Our algorithm projects the field backward in time via linear wave theory. The approach is similar to previously reported methods but is designed to work with time-constant data and avoids problems associated with evanescent waves.
Scattering from elastic sea beds: First-order theory
Authors: Jackson Darrell R. , Ivakin Anatoliy N.
Published: July 2002

A perturbation model for high-frequency sound scattering from an irregular elastic sea bed is considered. The sea bed is assumed homogeneous on the average and two kinds of irregularities are assumed to cause scattering: roughness of the water–sea bed interface and volume inhomogeneities of the sediment mass density and the speeds of compressional and shear waves. The first-order small perturbation approximation is used to obtain expressions for the scattering amplitude and bistatic scattering strength. The angular dependence of the scattering strength is calculated for sedimentary rock and the influence of shear elasticity is examined by comparison with the case of a fluid bottom. Shear effects are shown to be strong and complicated.
A stochastic model for wave localization in one-dimensional disordered structures
Authors: Yang Yang , Photiadis Douglas M.
Published: July 2002

This paper presents a probabilistic study of the effects of structural irregularity on wave propagation along an infinite 1-D chain. A general integral equation method based on Markov chain theory is used to determine the phase probability density function (pdf) at the scatterers distributed irregularly along the chain. The scatterers could be atoms in a one-dimensional crystal, or ribs on a flat plate or membrane. The integral equation derived for the phase pdf is simplified considerably when the scatterers are distributed completely randomly or quasi-periodically. In these cases, the integral equations may be asymptotically solved for the phase density functions in the limit of weak or strong scattering; the localization factors are then obtained. The present approach is quite general and is directly applicable to any disordered one-dimensional system consisting of identical scatterers that are arranged according to a probability distribution function. The validity of the present asymptotic solutions is examined and verified by comparing against the existing analytical solutions for simple atomic or mechanical disordered systems.
Experimental evaluation of simplified models for predicting noise levels in industrial workrooms
Authors: Hodgson Murray
Published: July 2002

Simplified models for predicting noise levels in industrial workrooms have been developed by Friberg, Thompson et al., Wilson, Embleton and Russell, Kuttruff (“diffuse” and “specular” models applicable to fitted rooms only), Zetterling, Sergeyev et al. (applicable only to untreated workrooms), and Hodgson. They predict octave-band or A-weighted steady-state sound-pressure level as a function of source/receiver distance. These models have been programmed and evaluated by comparing predicted sound-propagation curves with those measured in 30 empty and fitted industrial workrooms with and without absorptive ceiling treatments. In empty workrooms the Sergeyev et al., Thompson, and Hodgson models worked quite well. The Zetterling model performed moderately well. The other models were inaccurate. Models underestimated levels in most cases. With the addition of absorbent treatments the accuracy of the Friberg, Wilson, Zetterling, and Embleton and Russell models improved; that of the Thompson and Hodgson models worsened. In fitted workrooms the Hodgson and Kuttruff (diffuse) models were accurate. The Friberg and Zetterling models were moderately accurate. The other models were inaccurate. The Thompson and Kuttruff (specular) models generally overestimated levels; the other models tended to underestimate levels. With absorbent treatment the accuracy of the Embleton and Russell model improved.
Locus of generation for the 2 f1−f2 vs 2 f2−f1 distortion-product otoacoustic emissions in normal-hearing humans revealed by suppression tuning, onset latencies, and amplitude correlations
Authors: Martin Glen K. , Jassir David , Stagner Barden B. , Whitehead Martin L. , Lonsbury-Martin Brenda L.
Published: July 2002

The present study used distortion-product otoacoustic emission (DPOAE) suppression tuning curves (STCs), DPOAE onset latencies (OLs), and DPOAE amplitude correlations to investigate the locus of generation of the 2 f1−f2 DPOAE versus the 2 f2−f1 DPOAE in humans. The results of the tuning study revealed that, for the 2 f1−f2 DPOAE, the tips of the STCs tuned consistently below the geometric-mean (GM) frequency of the primary tones. In contrast, for the 2 f2−f1 DPOAE, STCs tuned above the GM of the primaries, with 50% of the tip frequencies at, or above, the 2 f2−f1 frequency place. When the average ratio of the 2 f2−f1 to the 2 f1−f2 tip frequencies was computed, a factor of 1.44 provided an estimate of the frequency shift needed to align the two DPOAE generation sites. Other results showed that OLs for the 2 f2−f1 DPOAE were uniformly shorter than those for the 2 f1−f2, with differences at the low frequencies amounting to as much as 6–7 ms. Further, for both DPOAEs, curves describing latency decreases as a function of increasing GM frequencies were best fit by power functions. Shifting the GM frequency producing the 2 f2−f1 DPOAE by a factor of 1.6 caused the latency distributions for both DPOAEs to overlap thus resulting in a single function that described cochlear delay as a function of GM frequency. Finally, for each GM frequency in the DP-gram, sliding correlations from 108 normal ears were performed on both DPOAEs by holding the primaries producing the 2 f1−f2 DPOAE constant, while all 2 f2−f1 DPOAE amplitudes were successively correlated with the 2 f1−f2 amplitudes. This procedure demonstrated that, for a given GM frequency producing the 2 f1−f2, the correlations between the two DPOAEs peaked when the primaries of the 2 f2−f1 were at a GM frequency that positioned the 2 f2−f1 frequency place near the GM of the primaries that produced the 2 f1−f2 DPOAE. As a whole, the above findings strongly suggest that the 2 f2−f1 DPOAE in humans is generated basal to the primary-tone place on the basilar membrane.
A new method of sequential information analysis for consonant perception
Authors: Tan Chin Tuan , Tong Yit Chow , Chang Joseph S.
Published: July 2002

A new method of information analysis which considers all entries in the confusion matrix is proposed for identifying important articulatory features for consonant perception. For a system of N features, the important features are selected sequentially in a series of iterations. In the first iteration, the feature whose transmitted information is maximum is selected. In the second iteration, the first feature is combined with each of the remaining N−1 features to form N−1 sets of two joint features. The second feature is selected by choosing the set of joint features which has the maximum transmitted information. The transmitted information for a set is determined by summing its unconditional transmitted information and a correction factor to account for redundancy. The third and subsequent important feature are selected in the same manner. The selection process terminates when the transmitted information for the selected set of joint features in the last iteration is equal to the maximum possible transmitted information for the confusion matrix. The proposed method of analysis was tested with several confusion matrices published in past research papers, and the results showed that it was able to select a common set of important features consistently across the confusion matrices.
A training system to improve the usefulness of hearing aids
Authors: Ikeda Takashi , Watanabe Akira , Hino Mitsuaki , Ueda Yuichi
Published: July 2002

An audiovisual system was proposed previously. It can make the hearing-impaired notice phonemic cues by using vision together with hearing. In the preliminary experiments by a hearing impaired person with a prototype hearing aid, it was noted that the information of visualized speech could contribute to combine sound images and their understanding as phonemes. First, to generalize this method, a new type of training system that consists of a personal computer, audio equipment, training programs and data is presented. The system provides output sounds of hearing aid and visualized speech together automatically. Next, the results of experiments, which are conducted with the groups of hearing-impaired persons by sound of the single resonant analysis type hearing aid, are presented. The contribution of visual information in the training will be examined. Finally, it will be discussed that the design to realize a practical single resonant analysis type hearing aid is small enough to use in the field test.
An efficient Korean speech recognition for name and digit on a communication network
Authors: Lee Hyeon-Gu , Hong Kwang-Seok
Published: July 2002

The necessity of service using speech recognition goes on increasing day by day, but for the difficulty of speech recognition, the efficient application service is not activated yet. For the security of users and the convenient use of a disabled person, a desire for using speech about the user’s name, ID number, or an account which is mainly used in service is increasing accordingly as the information and banking service on communication network increases. This paper presents the efficient Korean speech recognition method for name and digit string on a communication network. Because most of the traditional methods recognize the full name at once, there are some problems, in that they must have all patterns of the name or require very accurate segmentation of the phoneme, etc. So the unit of Korean spelling pronunciation that is contained in the name is recognized, and then a connection is made to the full name recognition. In the recognition of the digit, semisyllable and the combination of a connected semisyllable are used. That is, 20 semisyllables from 0 to 9 are used as the pattern of the start semisyllable or the end semisyllable of the digit string in recognition, and 100 connected semisyllables from 00 to 99 are used as the pattern of the connection part of each digit string. In comparison with the existing method, all names and digits, regardless of their kind and figure can be recognized.
A multilevel training procedure for speech recognition in noise
Authors: Tan Chin Tuan , Tong Yit Chow , Chang Joseph S. , Chan Lai Peng
Published: July 2002

A multilevel training procedure is proposed for automatic speech recognition in noise. In the multilevel training procedure, six levels of Gaussian noise, corresponding to six signal-to-noise ratios (SNR), were added to the training samples. The procedure was tested in three speech recognizers implemented, respectively, on the basis of the following three speech recognition techniques: dynamic time warping (DTW), discrete hidden Markov modeling (DHMM), and neural network (NN). The tests on the three recognizers were conducted using a common database for isolated word recognition of the ten digits. The results of the first series of the tests showed that when the recognizers were trained using clean training samples without noise and tested with noisy test samples with additive Gaussian noise, the performance of the three recognizers degraded significantly as the SNR decreased. In the second series of tests, the multilevel training procedure was applied. The results showed that, when the recognizers were trained with noisy training samples and tested with noisy test samples, the three recognizers were, in contrast to the first series, much better able to maintain their performance as the SNR decreased. The multilevel procedure is therefore effective in improving the performance of recognizers for automatic recognition in noise.
End corrections in an acoustic model of the vocal tract
Authors: Dang J.
Published: July 2002

Open end corrections are required in an acoustic model to compensate for the effects caused by abrupt area changes in the vocal tract, such as the open ends of the side branches and the opening between the teeth. The end corrections were studied by means of a number of mechanical models which imitated vocal tract geometries. The results showed that the end correction greatly depends on the local geometries in the vicinity of the open ends. On the basis of these measurements, two empirical formulas have been proposed to account for the acoustic effects of the geometries [Dang et al., TR-H-219 (1997)]. In this study, these formulas were applied to morphological data of the vocal tract. The range of the open end correction coefficient (OECC) for estimating an accurate vocal tract transfer function was evaluated. The OECC of the nasopharyngeal port was found to be about 0.8 for the oral tract side and 0.3–0.6 for the nasal tract side. For the ostia of the paranasal cavities, the value was about 0.8 for the cavity side and 0.28 for the nasal tract side. An OECC of about 0.4 was sufficient to compensate for end radiation of the laryngeal tube outlet for front vowels.
Stabilization of a capillary bridge far beyond the Rayleigh–Plateau limit using active feedback and acoustic radiation pressure
Authors: Marr-Lyon Mark J. , Thiessen David B. , Marston Philip L.
Published: July 2002

A liquid bridge between two solid surfaces is known as a capillary bridge. Long bridges naturally become unstable to a symmetric mode by bulging near one end while the opposite end thins. Suppression of this mode is important for managing fluids in low gravity. For a cylindrical bridge in low gravity of radius R and length L, the slenderness S=L/2R has a natural (Rayleigh–Plateau) limit of π beyond which the bridge breaks. Using acoustic radiation pressure to control the bridge’s shape, and an optical sensor to detect the bridge’s shape, an active feedback system in simulated low gravity was constructed. This system stabilized bridges significantly beyond the Rayleigh–Plateau limit, with S as large as 4.3 [Marr-Lyon et al., J. Fluid Mech., accepted for publication]. To be useful in low gravity, this technique will have to be modified to work on liquid bridges in air. In preparation for an experiment to be performed on NASA’s low gravity KC-135 aircraft, acoustic resonators in air that have the required property that the sound amplitude can be spatially redistributed rapidly are being investigated and tested in normal gravity using soap-film bridges filled with sulfur hexafluoride. [Work supported by NASA.]
Surfactant-bearing drop dynamics in microgravity
Authors: Chen Xiaohui , Apfel Robert E.
Published: July 2002

Experiments of large amplitude nonlinear oscillations of axially symmetric surfactant-bearing (triton or bovine serum albumin) water drops with different initial aspect ratios and drop volumes were performed during the second United States Microgravity Laboratory, USML-2 in November 1995. Through the numerical simulation of the oscillations of these surfactant-bearing water drops using the boundary integral method, the surface dilatational viscosity and the surface shear viscosity are found to play a major role in these oscillations. How they influence the damping coefficient and the drop oscillation frequency will be presented. The numerical and analytical results and a physical picture of the relation between the surface viscosities and the damping coefficient will be described. In addition, the relation between the initial drop aspect ratio, the surface viscosities, and the surface rotational velocity will also be presented. [Work supported by NASA through JPL, Contract No. 958722.]
Regular interval stimuli with more than one pitch
Authors: Yost William A.
Published: July 2002

Iterated ripple noises (IRN) are an example of regular interval stimuli (RIS) that can have more than one pitch. IRN stimuli are generated by a cascade network of delay (d in ms), attenuate (−1≤g≤1), and add circuits. When g=−1.0 and the number of cascaded circuits (number of iterations) is less than four, the pitch of IRN stimuli has two values. The addition of two IRN stimuli, each generated with different values of d, can also have two pitches. Listeners were asked to discriminate between an IRN stimulus generated with g=−1.0 and the sum of two IRN stimuli with each IRN stimulus generated with a different value of d. Performance was measured as a function of the difference in the two values of d. The center value of d, g, and the number of iterations were also parameters of the experiments. For certain separations of d and for certain values of g, listeners had difficulty discriminating between the two stimuli. The discrimination procedure allows for an objective identification of the pitch of IRN stimuli when g=−1.0. The results will be discussed in terms of models of complex pitch. [Work supported by NIDCD and AFOSR.]
Comparison of calculated and measured data for the attenuation of improved noise barrier
Authors: Yamamoto Kohei
Published: July 2002

The comparison between calculated and measured data was made to the additional noise attenuation of a barrier with an absorptive device on the top edge. The attenuation was determined at a highway with a noise barrier of 3 m height. By the use of the boundary-element method (2D), noise attenuation due to the mounting of the device was estimated. The results showed an agreement between the calculated and the measured value. However, the calculated values showed fluctuation when they were plotted against distance.
Active noise barrier based on the boundary surface control
Authors: Ise Shiro , Tachibana Hideki
Published: July 2002

Based on the principle of the boundary surface control, it is possible to make a surface which reflects acoustic energy flow by controlling the pressure and its gradient on the surface. In order to realize this method, a system has to be made using multiple speakers and multiple microphones. The system should be adjusted so that the output signals of the multiple microphones which are located on the surface considered to be the location of the barrier are minimized. By changing the distance between the microphones, the best effect of the active noise barrier can be obtained. For instance, in a level lower to the ground, many microphones are needed to control higher frequencies. On the other hand, in a level higher from the ground, it is not necessary to use so many microphones to control lower frequencies. The merit of this method is that the apparent height of the barrier, depending on the frequency considered, can be changed. In this paper, the active noise barrier based on the boundary surface control is introduced and the effect of this method is calculated using BEM.
Atmospheric sound propagation near the ground
Authors: Larsson Conny
Published: July 2002

The most important meteorological effects on sound propagation are refraction-ground attenuation, atmospheric absorption, and scattering by turbulence. The wind and temperature gradients cause refraction of the sound rays, and hence influence the sound level. The curvature of a nearly horizontal sound ray can be calculated by using measurements of wind and temperature. Atmospheric absorption is important for outdoor sound propagation. The absorption varies for different parts of the world. Results from some stations in Scandinavia are given. Atmospheric absorption for the high frequencies changes during different times of the year. The absorption for low frequencies varies both over the day and the year. The size of the annual and diurnal variance is frequency dependent. An increase in the accuracy of calculations of outdoor sound levels can be made by considering the local climate. The values of atmospheric absorption need to be calculated for the local climate instead of using some global mean value. The time has come to use data from local meteorological stations. Turbulence scatters sound into sound shadow zones and causes fluctuations of the phase and the amplitude of the sound waves.
Annoyance and health effects of entertainment noise
Authors: Rice C. G.
Published: July 2002

There is no doubt that over the past 20 years awareness in the possible adverse annoyance and health effects which can result from entertainment and leisure activities has increased, and the topic is becoming more worthy of research effort in its own right. A review of the literature has established certain knowledge base gaps in the causal relationships between such noise and its adverse effects. Indeed, apart from noise-induced hearing loss there appears to be a lack of in-depth research from which unequivocal quantitative conclusions can be drawn. While damage to hearing from certain activities has been established, in general nonauditory health effects are not apparent although annoyance effects and provoked violence are becoming increasing causes of major concern. However, bearing in mind the relatively low integrated noise exposures which personally accrue from participation in and exposure to entertainment and leisure noise activities, care should be taken not to overemphasize the adverse effects to the prejudice of the other positive benefits created by participating in such activities.
Environmental pollution and noise control of two power generator sets placed into a recreation area
Authors: Rastelli Victor , Montbrun Nila , Buoza Alexis
Published: July 2002

This paper deals with the noise control of two power generator sets of 545 kW each, located in a Venezuelan resort. The generator sets made a noise level of 73 dB(A) at a distance of 47 m from the source, where are located the guest rooms. This level was far from the legal regulations for allowable environmental conditions. A noise enclosure with air silencers was proposed to reach a final level of 53 dB(A) at the guest accommodations. Two aspects must be taken into account, the noise reduction and the ventilation specifications of the generator sets, in order to keep the operational temperature. The final noise level was measured at 54 dB(A) and the pressure loss of the silencers was well placed into the ventilation specifications.
Variations of the leaky-integrator model and validation with detection of sinusoidal and percentage duty-cycle modulated noise
Authors: Nelson John A. , Bentler Ruth A.
Published: July 2002

The leaky-integrator model is commonly used to describe auditory temporal processing. In the current study, the low-pass filter and decision device of the model were investigated. Maximum-to-minimum envelope magnitude, rms envelope power, envelope fourth moment, and envelope crest factor were implemented as potential decision devices [E. A. Strickland and N. F. Viemeister, J. Acoust. Soc. Am. 99, 3638–3646 (1996)]. Model predictions demonstrated interaction between the decision device and low-pass filter cutoff frequency. To validate model variations, detection thresholds of sinusoidal and percentage duty-cycle modulated broadband noise were measured at octave frequencies 2–512 Hz for listeners with normal hearing. To ensure subjects responded exclusively to temporal cues, the modulated noise was compensated for overall average power. Further, an intensity roving technique was used between observation intervals. The data show low-pass transfer function characteristics across all modulation functions. A two-line regression fit of the data revealed that the cutoff frequency, slope, and overall gain varied as a function of the duty-cycle modulation function. The nonlinear growth of the various decision devices as a function of the low-pass filter is used to explain individual variability among subjects. [Work supported, in part, by the American Academy of Audiology Student Investigator Award.]
Variable-duration notched-noise experiments in a broadband-noise context
Authors: Hant Jim J. , Strope Brian P. , Alwan Abeer A.
Published: July 2002

To model the confusion of short-duration speech signals in noisy environments, it is important to characterize how humans resolve the spectral components of short-duration signals in background noise. Toward that end, a variable-duration notched-noise experiment was conducted in a noise context. Broadband noise preceded and followed a tone and notched noise of similar duration. Thresholds were measured at four durations (10, 30, 100, and 300 ms), two center frequencies (0.6, 2.0 kHz), and five relative notch-widths (0.0, 0.1, 0.2, 0.4, 0.8). At 0.6 kHz, 10-ms thresholds decrease 6 dB across notch-widths while 300-ms thresholds decrease over 35 dB. These durational trends are similar but less pronounced at 2 kHz. In a second experiment, the short-duration notched noise was replaced with a flat noise, which provided an equivalent amount of simultaneous masking to that of the first experiment, and thresholds dropped by as much as 20 dB. A simple filtering model with either a linear or nonlinear combination of simultaneous and nonsimultaneous masking is unable to predict these results. Instead, it appears that the elevated thresholds at short durations are dependent on the spectral shape of the simultaneous masker, implying reduced frequency selectivity. [Work supported by NIH-NIDCD and The Whitaker Foundation.]
Vibrational modes of a tenor steel pan
Authors: Hansen Uwe J. , Rossing Thomas D.
Published: July 2002

The tenor or lead pan has 28 or 29 notes covering the range of C4 to E6. In spite of its being the lead voice in a typical steel band, little research has been reported previously on its modes of vibration or its acoustical behavior. Using electronic TV holography, we have studied the modes of vibration of the various note areas in tenor pans by two different makers. Note areas generally have one or two overtone modes tuned harmonically to the fundamental [Rossing et al., Phys. Today 49(3), 24–29 (1996)]. Special attention has been paid to the strong acoustical and mechanical coupling between the note areas as they vibrate.
Annular time-domain acoustic source for horn acoustics
Authors: Agullo Joaquim B. , Barjau Ana
Published: July 2002

The time-domain description of horns is needed in time-domain simulations of acoustical systems such as wind instruments and is very helpful to understand their transient behavior. Unidimensional horn wave equations have been used as a first approximation. The Webster horn equation, however, is too unrealistic for horns with intense flare. The improved horn equation developed by Keefe et al. [Proceedings of the SMAC-93 (1993), pp. 496–500] may be promising. A discretized approach that assumes the horn is made of cylindrical and conical elements with plane and spherical waves, respectively, has been widely used. More general alternatives based on the Kirchhoff integral are attractive because they can handle 2 and 3-D acoustic fields and lead to a surface discretization instead of volume discretization (as required in FEM). For axisymmetric horns, the use of an annular time-domain unit source, not implemented in commercial codes, dramatically reduces the number of elements. This paper presents the development of such a source and its application to time-domain horn acoustics. An example is presented for a cathenoidal horn and its results are compared to time-domain measurements.
Determining resonance frequency changes due to bore irregularities in woodwinds
Authors: Nederveen Cornelis J. , Dalmont Jean-Pierre
Published: July 2002

To determine tuning changes due to deviations from an ideal cylindrical or conical bore constituted by holes and by diameter variations, these deviations are modeled by local changes in compressibility and inertance. The magnitude of a compressibility change, being proportional to the volume change of a perturbation, is easily determined, but inertance changes pose more problems, since they may be less than proportional to the volume change. Determining these inertance changes is possible in various ways. An approximation (with a systematic error) is obtained by depicting the 3-D flow as stratified and integrating 2-D results obtained from Schwarz–Christoffel transformations, for which various accurate analytical and numerical solutions are available. Exact results can be obtained with finite difference methods: they are simple, have no systematic error, and only demand a large computing power. Results will be presented for correction impedances at hole junctions, hole ends with various flanges, and toroidal bends. A method of experimental verification and some results will be discussed.
Reactions to aircraft noise
Authors: Gjestland Truls
Published: July 2002

The data from three Norwegian surveys on reactions to aircraft noise have been reanalyzed. Data from more than 9000 respondents have been included. The response to a general annoyance question is compared to the response to questions on special effects concerning speech communication, sleep and rest disturbance, ambient noise, behavioral patterns, etc. The established dose–response relationships will form the basis for new Norwegian guidelines for assessing aircraft noise.
Psychological evaluation of sound environment with mixed sources
Authors: Kuwano Sonoko , Kaku Jiro , Kato Tohru , Namba Seiichiro
Published: July 2002

In order to make the experimental situations closer to actual situations, experiments have been conducted in field conditions, where various sounds were audible. The results suggested that generally speaking, LAeq was a good measure for the evaluation of environmental noises. However, there were some cases where LAeq did not show good correspondence with subjective impressions. On the basis of the results of the field experiments, an experiment was conducted in a laboratory situation with precisely controlled stimuli in order to examine the applicability of LAeq to sound with mixed sources. Sounds which consisted of aircraft noise, road traffic noise, and train noise were used. Three events of each noise source were included in 15-min stimuli and the duration of road traffic noise was varied in three stimulus conditions. Subjects were asked to judge instantaneous loudness using the method of continuous judgment by category. After instantaneous judgment was over, overall impression of loudness of each sound source as well as the overall loudness of 15-min stimuli were judged. The following analyses was conducted; (1) comparison among three sound sources, (2) comparison among three experimental conditions, and (3) overall evaluation of sound environment with mixed sound sources.
The effects of exposure of intense free-field impulse noise on humans wearing hearing protection: Implications for new criteria
Authors: Patterson James H. , Johnson Daniel L.
Published: July 2002

In a series of studies, human volunteers wearing earmuffs were exposed to high-intensity free-field impulse noise which simulated various military weapons noise. By detonating increasing amounts of explosive material, the exposure level was varied from one at which no effects were expected to a maximum level just below the threshold of injury to the lung and upper airway. The number of blasts was varied from 6 to 100. Temporary threshold shifts (TTS) measured 2 to 6 min after each exposure were used as the measure of effects on hearing. The highest level for each number of blasts at which 95% of the exposed population would not show a significant TTS was used to establish the maximum safe exposure levels as measured in the open and under the earmuffs. These maximum safe exposure levels exceed the exposure limits used in the United States and other countries. Also, these results suggest that the number-intensity trading rule is more shallow than that used in most criteria. The exposures measured under the earmuff provide a good indication of the hazard. Implications of these results for impulse noise exposure criteria will be discussed.
Generation of harmonics in the near field of finite amplitude sources
Authors: Kozaczka Eugeniusz , Grelowska Grayna
Published: July 2002

It is well-known that the fine structure of the near field brings difficulties to both the theoretical and the experimental investigations. However, in many applications the phenomena occurring in the near field are the most important ones. It is caused by such a measurement setup arrangement and the choice of the parameters of used transducers that the measured or diagnosed objects are situated in the near field. The paper presents the results of the experimental investigations of the finite amplitude wave radiated by mono- and biharmonic sources of different radiating apertures. The measurements were carried out in the near field of the transducer using the high-precision device which controlled the movement of the receiver. The pressure was measured using a 1-mm-diam PVdF needle hydrophone. The results of investigations will be presented as the 3-D distribution of a primary wave and its harmonics. The distributions of chosen harmonic will be shown in planes perpendicular to the acoustic axis of the transducer or along this axis. The experimental results were compared with the numerical ones.
‘‘Induced transparency’’ of an acoustical waveguide containing a liquid with nonlinear viscosity
Authors: Zozulya Oleg M.
Published: July 2002

Sound propagation in an acoustical waveguide containing a liquid that depends on a shear rate γ̇ viscosity is considered. A number of rheological models adequately representing such dependence in a wide range of γ̇ variation are well known (power law, Carreau model for stuctural viscosity, etc.). In most of the models viscosity decreases with the increase of γ̇. It can result from the destruction of some internal microstructures in the liquid and their self-reconstruction with a reduction of load (as, e.g., in a sol–gel transition). For such liquids the effect of ‘‘induced transparency’’ for intense sound waves is expected. In this work the effect is determined for harmonic waves, propagating in an acoustical waveguide with cross−section dimensions that are small in comparison with the sound wavelength. The dependence of viscosity on γ̇ is taken into account by retention of the first nonlinear term in the expansion of shear tension στ in shear rate γ̇: στ=ηγ̇−ξγ̇3, where η and ξ are linear and nonlinear parameters of viscosity, respectively. Sound attenuation in acoustical waveguides is due to the partial transmission of acoustical energy to the viscous waves being generated in the vicinity of waveguide walls. It is shown that the nonlinear viscosity gives rise to the effective viscous wave self-interaction, which results in the reduction of the energy dissipation. The application of the effect to the measurements of the nonlinear viscosity parameter is considered and the importance of this parameter for the evaluation of the oil quality is discussed. Work was partially supported by the Russian Foundation of Basic Research (project 97-08-17789).]
A descriptive model of under- and over-shoot patterns in the temporal masking function of a narrow-band noise
Authors: Formby C. , Rutledge J. C. , Heinz M. G. , Sherlock L. P. , Alexsandrovsky I. V.
Published: July 2002

Complex temporal patterns of under- and over-shoot (i.e., enhanced and diminished detection) at onset and offset of a 200-ms, 400-Hz-wide noise masker (2500 Hz center frequency) have been reported for a temporal masking paradigm [Formby et al.., J. Acoust. Soc. Am. 102, 3161(A) (1997)]. The paradigm measured detection of a 5-ms sinusoidal signal (1900 or 2100 Hz) at 43 temporal positions preceding, during, and following the masker. Efforts to model under- and over-shoot patterns in the temporal masking functions will be considered. The modeling strategy is similar to that used in the study of vestibular adaptation [Formby et al., 21st Meeting of Association for Research in Otolaryngology, 1988], and allows for quantification of the relative amplitudes, time constants, and delays of excitatory and adaptive processes. Linear combinations of exponential functions are used to represent the effects of excitatory masking and adaptation. Suppression and adaptation of suppression may also play roles in the masking process. These effects can be evaluated in our scheme by including corresponding exponential functions to represent the processes. The least-squares fitting model will be described and parameter estimates for the model will be presented for a range of masker levels. [Work supported by ONR.]
Global and continuous judgments of sounds with time-varying intensity: Cross-modal matching with a proprioceptive input device
Authors: Susini Patrick , McAdams Stephen
Published: July 2002

Conventional psychophysical methods such as paired comparisons and psychophysical scaling present problems in the case of real-time perceptual evaluation of long-duration nonstationary sounds. In order to obtain loudness profiles of such sounds and to determine their relation to global loudness judgments, a continuous judgment method was employed using cross-modal matching of loudness with muscular force in a proprioceptive input device. This device was perceptually calibrated with 1-kHz stationary pure tones using S. S. Stevens’ cross-modal matching paradigm. The loudness scale is equivalent to that obtained with ratio and magnitude production methods. The device was then tested on pure tones varying in intensity from 10 to 53 s. In order to observe the different memory processes involved in the global evaluation, several types of contours were used with from one to three peaks. The global judgment was influenced both by the duration and the level of the last contour peaks. The continuous judgment task also appeared to entail an auditory and gestual learning of the signal contour.
Scattering from fluid-loaded elastic targets near penetrable sediments
Authors: Bishop Garner , Smith Judy
Published: July 2002

A T-matrix formalism developed for plane-wave scattering from a fluid-loaded elastic spherical shell near a rough fluid elastic interface [J. Acoust. Soc. Am. 101, 767–788 (1997)] is generalized to include scattering from targets for which a T-matrix is known and constructed on a spherical wave basis, and from the interface of penetrable sediments for which a T-matrix is known and constructed on a Weyl plane-wave basis. A variety of numerical results are obtained for plane-wave scattering from elastic spheres and elastic spherical shells, as well as a finite elastic cylinder with hemispherical end caps near rigid, soft, fluid, elastic, poroelastic, and layered fluid sediments with planar boundaries that demonstrate some of the effects’ target-boundary scattering on resonant and nonresonant scattering.
Research and development of robust speech recognition
Authors: Aikawa Kiyoaki
Published: July 2002

This paper describes recent research and development activities on robust ASR (automatic speech recognition) in NTT Human Interface Laboratories. ASR system design has been changing from the experimental to the commercial level. A relevant issue in achieving practical ASR is the robustness against environmental noise, and speaker and circuit differences. Adaptation technique has been widely investigated for improving the robustness. It should be noted that customers or users generally prefer simple procedures and quicker response. Thus real time speaker/noise adaptation is getting more and more popular. Confidence measure is also effective in enhancing speech recognition performance by rejecting out-of-vocabulary words, lip, and breath noises. Another relevant issue is how to quickly reach the goal of the human-machine dialog via the telephone without visual information. The voice quality through a telephone is not good enough to provide perfect syllable intelligibility even for a human. This suggests that an appropriate human-machine dialog is needed to support an ASR system in acquiring the information or request intended by the user. Recent research topics including spectral estimation, dialog control, and other new approaches related to above discussion will be shown in the presentation. ASR application examples and related problems will also be shown.
From speech recognition to understanding: Shifting the paradigm to achieve natural human–machine communication
Authors: Juang B. H.
Published: July 2002

Automatic speech recognition has made substantial progress in the last decade due to applications of statistical methods. The task of speech recognition is often described as converting a sequence of sounds into a sequence of linguistic events (words). This is in essence transcription: given an acoustic signal, the system determines which word it represents. The implicit assumption in this traditional formulation of speech recognition is that the speech is well-formed, free from extraneous components such as disfluency, partial words, repairs, hesitation, etc. The statistical framework that supports the formulation is the Bayes decision theory. For natural human–machine communications, however, it is expected that the spoken input to the system will inevitably contain extraneous components, treatments of which have to be developed in a broadened framework utilizing the Neymann–Pearson Lemma. The departure also inspired a new formulation based on the signal detection theory for robust speech understanding. In this talk, the rationale for the new formulation will be discussed and new results towards speech understanding using the detection-based (as opposed to the transcription-based) approach will be presented. It will be shown that the new approach achieves substantial improvements in accuracy for natural but ill-formed utterances over the conventional approach.
Virtual reflecting walls for improving the acoustics of defective halls
Authors: Vian Jean-Paul , Meynial Xavier
Published: July 2002

Many performing art centers and multipurpose halls with poor acoustics cannot be improved for economical reasons by architectural correction works or by mechanical moving devices. The principle of virtual acoustic wall [Guicking et al., J. Acoust. Soc. Am. 78 (1985)] has been successfully applied to control the reflectivity and diffusivity of concert halls existing walls. Several independent active cells composed of a microphone, an electronic unit, and a loudspeaker form a virtual wall. By appropriately controlling the open loop gain in the cells of each virtual wall it is possible to create different sound fields fully coherent in space and time with the preexisting one. The first reflections can thus be reinforced and a natural reverberation enhancement effect is obtained by the acoustic exchange between the virtual walls. Computer sound simulation of three virtual walls in a concert hall was performed and experimental results, both objective and subjective, are shown for two concert halls in which a virtual wall system was installed.
Characteristics of the low- and high-frequency ambient sound from breaking waves in the surf zone
Authors: Kolaini Ali R. , Nystuen Jeffrey A.
Published: July 2002

An experiment was conducted along the Scripps Institution of Oceanography Pier to study the characteristics of the ambient sound in the surf zone. The role of the single large bubble’s (order of a few centimeters) oscillations versus the collective oscillations of bubble clouds as possible mechanisms of generation of the low-frequency sound, especially the relatively broadband peak that appears around 20 Hz in the sound spectrum, is discussed. The frequency band at which ambient sound from breakers in the surf leaks out is identified. The acoustic attenuation which makes the surf zone extremely quiet above 2 kHz, and a possible method to use ambient sound to infer bubble size distribution in shallow water near the surf zone, are discussed. [Work supported by ONR.]