bluuebunny/crossref_metadata_2025 huggingface dataset:
Numerical simulation with experimental validation for nonlinear standing wave phenomena
Authors: Zhu Zhichi , Zhou Anqi , Huang Dongtao , Liu Ke
Published: July 2002

The study of nonlinear standing waves is of great significance to nonlinear acoustics. The strong nonlinear standing wave phenomena are constantly encountered in the experimental studies of acoustics, such as multiplying growth of higher harmonics, saturation of harmonics, bifurcation, and chaos. The experimental study of strong nonlinear standing waves is highly restricted by the sound source, which is very difficult to build up when its sound power becomes very strong. Therefore, in this paper, a numerical approach to the problem was adopted. By the use of Euler equations and MacCormack fourth-order difference method, the multiplying growth and saturation of higher harmonics in a nonlinear standing wave have been numerically simulated. By increasing the sound-pressure level of the excited source from 125 to 190 dB, the entire development process of the nonlinear standing wave was clearly illustrated. Some new interesting results were obtained from the simulation. For example, the sound-pressure level at zero frequency enlarges quickly with increasing the intensity of the excited source. A detailed comparison between numerical simulation and the relevant experimental results shows that the numerical investigation is successful. [Work supported by NSFC.]
Analytical method for radiation and scattering problems in noncanonical domains
Authors: Grinchenko Victor T.
Published: July 2002

One of the tendencies in modern acoustics is to use analytical solutions of the Helmholtz equation to present the sound field around radiators and scatterers of noncanonical shapes. The T matrix and internal source density methods are examples. In the present paper a method called a partial domain method is discussed. An essential point of the method is the notion of general solution of the boundary-value problem for the Helmholtz equation. The boundaries of the objects under consideration are parts of coordinate surfaces in separable systems. However, surfaces may be either coordinate surfaces of different types (as in the case of a finite cylinder) or coordinate surfaces of different systems (as in the case of a cylinder with spherical lids). Theoretical problems of both the method and numerical implementation of one are discussed. Consideration of such aspects of the method as singularity of the field near corner points, nonuniqueness of the wave functions, and extension of boundary conditions gives a basis to make the computation process more effective. The numerical results for some cases of radiation by finite cylinder and cylinder with hemispherical lids are used to illustrate the method.
The auditory attentional blink in a congenitally blind population
Authors: Goddard Kim M. , Slawinski Elzbieta B. , Isaak Matthew I.
Published: July 2002

The attentional blink (AB), a temporary information processing deficit that follows an attended-to target, has been demonstrated most recently using pure-tone stimuli in a rapid auditory presentation (RAP) paradigm. The current study compared this auditory AB in a group of congenitally blind individuals to nonblind individuals. RAP streams were created using 25 equally loud, randomly generated 1000–2490-Hz tones (11 tones/s). Tones were 85 ms in duration and were separated by a 5-ms silent interval. Target and probe tones were 1500-, 2000-, or 2500-Hz tones increased in intensity by approximately 10 dB SPL relative to the stream tones. In the experimental condition, participants identified the target and probe according to pitch (low, medium, or high). In the control condition, participants performed only the probe task. Probe detection accuracy was measured. Results revealed that in the control condition, participants reported a single loud tone with near ceiling accuracy. In contrast, in the experimental condition, probe detection accuracy was impaired for approximately 270 ms following the target and this impairment was significantly attenuated in the blind population. Findings are discussed in terms of compensatory attentional and temporal processing mechanisms. [Work supported by NSERC.]
Monitoring the simultaneous presentation of multiple spatialized speech signals in the free field
Authors: Nelson W. Todd , Bolia Robert S. , Ericson Mark A. , McKinley Richard L.
Published: July 2002

The effect of spatial auditory information on listeners’ ability to detect, identify, and monitor the simultaneous presentation of multiple speech was evaluated in the free field. Factorial combinations of four variables, including the number of localized speech signals, the angular separation of the speech signals, the location of the speech signals along the horizontal plane, and the sex of the speaker were employed using a completely within-subjects design. Participants were required to detect the presentation of a critical speech signal against a background of nonsignal speech events. Speech stimuli were derived from a coordinated call sign test which consisted of a call sign (‘‘Ringo’’), a color (‘‘red’’), and a number (‘‘five’’). In addition to having high face validity for aviation communication tasks, this measure has been successfully employed in competing message experiments. The experiment was conducted at the USAF Armstrong Laboratory’s Auditory Localization Facility—a 277-speaker geodesic sphere housed within an anechoic chamber. Performance efficiency was evaluated in terms of percentage of correct detections and false alarms, reaction time, and signal detection theory indices of perceptual sensitivity and response bias. Implications for the design of spatial auitory displays to enhance communication effectiveness and situation awareness are discussed.
Selective attending to auditory streams in complex sequences: Frequency and/or temporal expectations?
Authors: Drake Carolyn , Brochard Renaud , Adenier Matthieu
Published: July 2002

Both frequency and temporal information influence the perceptual organization of complex sequences: stream segregation is more likely with wider frequency separations and at faster rates. The influence of frequency and temporal information in directing attention to a stream within a complex sequence is investigated. Complex sequences of three subsequences (each with a specific frequency/tempo combination) were preceded by a cue sequence containing different types of information (FT=frequency + tempo, F=frequency, T=tempo, S=silence). The ability to selectively attend to one subsequence was measured by correct detections of a small temporal irregularity in the cued subsequence. When stream segregation was easy (large frequency separations), performance was similar for FT and F (93%), intermediate for T (79%) and random for S (64%): frequency information facilitated selective attending more than temporal information. When stream segregation was hard (small frequency separations), performance was poorer and similar for all conditions (65%): temporal information became predominant when frequency information was less useful. Thus listeners adapt their attending strategies to coincide with the most pertinent information. Results are discussed in terms of diverging predictions of the theories of Bregman [1990] and Jones [1993].
A new finite-element analysis approach for the prediction of minimum and maximum vibroacoustic response of solid–fluid systems
Authors: Genot Jean-Sébastien , Charron François , Atalla Noureddine
Published: July 2002

Engineers are aware that identical products do not have necessarily the same dynamic response because of small differences in their dimensions or material properties. The classical finite-element method cannot take account of this fact. Statistical tools are needed. In order to consider the input data variability, new modules have been integrated in a finite-element code to estimate the maximum and minimum response levels (mean quadratic velocity and radiated power) of a structure. Up to now, numerical modules were generally based on Monte Carlo or perturbation methods. Both have limiting and complementary drawbacks: The Monte Carlo method is accurate but far too slow, even for small finite-element models, while the perturbation scheme is fast, but unable to predict nonlinear behavior of the dynamic response properly with respect to the input variables. The developed approach takes advantage of the response modal decomposition to identify intermediate variables which are nearly linear. The intermediate variables (eigenfrequencies, dynamic factors) are easily interpolated, thus requiring little computation. Results are presented to prove the efficiency of the method.
Hybrid control for vibration and acoustics
Authors: Clark Robert L. , Bernstein Dennis
Published: July 2002

The ‘‘standard problem,’’ frequently discussed in the controls literature, is detailed for application to noise and vibration control. Specifically, the feedback, feedforward, and hybrid (combination of feedback and feedforward) control system architectures are developed in the framework of the standard problem. The hybrid measurement structure and problem was formulated to demonstrate that the LQG compensator design for control of stochastic inputs is separable with respect to inputs which can be measured directly. Thus, one can design a feedback controller in the absence of measurable input disturbances if the objective is to combine both feedback and feedforward or adaptive feedforward control for stochastic and measurable inputs, respectively. Once the feedback controller is designed for the stochastic inputs, the feedforward controller required for the measurable input disturbances can be designed in a subsequent formulation of the control problem. Realizing that the design of such compensators is separable serves to simplify the hybrid design process.
Fifty-four years in speech research
Authors: Fant Gunnar
Published: July 2002

This is a brief presentation of my scientific career with outlooks on the development of speech research in the last 50 years. It started with an electrical engineering thesis at KTH in 1944–1945. The topic was intelligibility loss as a function of bandwidth reduction in telephony and related problems in assessing the effects of various types of hearing loss. In 1945–1948 I was employed by the Ericsson Telephone Company where I conducted basic studies of the spectral characteristics of Swedish speech sounds. This early period and my two years at MIT, 1949–1951, provided an entry to a pioneering era of speech research with new tools for speech analysis and synthesis and speech production modeling. It was during this time I started my work on the acoustic theory of speech production and established my cooperation with the linguists Roman Jakobson and Morris Halle and with Ken Stevens. I have followed the advent and the progress of speech research through the analog, the digital, and the computer age. Speech research now has the challenge to provide a deeper and more integrated understanding of all levels of the speech communication process. This is the ultimate requirement for the realization of advanced applications.
Speech research at the I. P. Pavlov Institute in Leningrad/St. Petersburg
Authors: Chistovich L. A. , Pickett J. M. , Porter R. J.
Published: July 2002

The ‘‘Leningrad Group,’’ led by L. A. Christovich and V. A. Kozhevnikov, carried out an extended research program to describe relations between speech perception and speech production and build a model of perception. The work first explored auditory processing mechanisms that could afford closely interdependent connections between auditory information and the control signals for motor production. The resulting model for auditory analysis employed amplitude modulation detectors as well as spectral measurements. The model was evaluated in perception tests with listeners mimicking and shadowing speech sounds and was also employed in a spectrographic display and a scheme for automatic speech recognition.
Chasing ideas in phonetics
Authors: Ladefoged Peter
Published: July 2002

In my early career I never stayed long enough in a particular field to be contradicted. I started as a poet learning about the sounds of words with David Abercrombie. Then, remembering my background in physics, I moved to studying acoustic phonetics. From there I became a pseudo-psychologist testing perceptual theories, until a meeting with a physiologist led to work on the respiratory muscles used in speech. Eventually I landed in Africa teaching English phonetics and learning about African languages. So, by the time I was asked to set up a lab at UCLA, I was a specialist in nothing. However, I was able to use my background to describe the sounds of a wide range of languages, becoming a sort of linguist. Computers and bright students led to other ways of analyzing sounds. Building a research group who felt that they had a stake in the development of the lab taught me their varied ideas from statistics to engineering, and the philosophy of linguistics. Now, still looking for the growing edge of the field, I think it might be in the physically observable activity of the brain; but perhaps this is because I have never been a pseudo-neurologist.
Acoustics and psychoacoustics of workstation audio systems
Authors: Toole Floyd E.
Published: July 2002

Workstations are widely used as production environments for multimedia, film, and audio programs. For many consumers, they are also the delivery systems for these programs, as well as for others, like music, television, and films, intended for reproduction in various conventional systems and environments. Listeners in workstations therefore must be able to perceive sounds with the appropriate sense of timbre, direction, distance, and ambiance whether the program was created in two-channel stereo, or any of the multichannel surround or binaural 3-D formats. The space constraints, and the usual restriction to two loudspeakers, put special demands on the design of these audio systems. Binaural image-steering techniques will be the basis for most interactive games and, through crosstalk-cancellation, binaural technology will enable realistic replications of two, four, and five-channel audio for music, films, etc. This paper reviews the requirements for workstation audio systems, discusses the key enabling technologies, and summarizes what is known about the appropriate technical specifications, measurement techniques, and subjective evaluation criteria and methodologies for evaluating workstation audio systems.
Distributed-mode loudspeakers and their impact on intelligibility in multimedia and sound distribution
Authors: Mapp Peter , Azima Henry
Published: July 2002

A new class of acoustic radiator has been investigated and shown to exhibit unusual radiation characteristics that can produce enhanced intelligibility under given listening conditions. The improvement mechanisms are investigated and shown to be related to a number of unique features. These include transient response, synergetic decay, lower inherent distortion, directivity, and the diffusivity of acoustic radiation. In a sound system design, it is desirable to produce an even coverage in both spatial and spectral domains. Conventional radiators, the majority of which work under a pistonic regime, exhibit two important characteristics that limit their effectiveness in real-world applications. These are the well-known narrowing of the acoustic radiation (dispersion) angle with increasing frequency and the coherency of the radiation. In contrast, the distributed-mode radiators, an emerging new technology, tend to overcome these obstacles and offer a new approach to improved sound distribution and multimedia. They combine a flat power response with an essentially constant and wide directivity (independent of frequency), augmented by a spatially and temporally diffuse radiation characteristic. The fast transient response results in a signal broadcast with very high clarity. The diffuse nature of the radiation suggests a lower degree of boundary interaction and spectral interference.
Reproduction of CD by providing ultrasonic atmosphere
Authors: Suzuki Jouji , Kaneko Shin’ichiro , Ohno Kiyomitsu
Published: July 2002

It is said that the LP sound reproduced by a well-prepared system is superior to the sound of a CD and the difference stems from whether the signal is recorded by analog or digital form. Considering master tape of LP is recorded in digital form, the main difference exists in the bandwidth of the reproduced signal. The signal of a CD is restricted to 20 kHz. On the other hand, the spectrum of an LP signal is extended to 100 kHz. Though the upper limit of hearing acuity of a human is 20 kHz, superior components will contribute to offer more rich and realistic atmosphere. In order to produce a higher-frequency component from CD sound, the folding spectrum of the sampling frequency of 44.1 kHz is utilized. The output signal of a DA converter of a CD player is filtered with BPF of 24–42 kHz. This signal is added to the CD output and presented to listeners. This system is named PULSA (providing ultrasonic atmosphere). A preference test is conducted on conventional CD sound and the sound by PULSA through a wideband static headphone on three minuets of orchestra and opera. It is revealed that PULSA sound is better than the conventional by the ratio of 65 to 35.
A multi-processor system for the production of virtual sound fields
Authors: Ordunã-Bustamante Felipe , Boullosa Ricardo R. , López Antonio P.
Published: July 2002

A multi-processor system is described for the production of virtual sound fields using small arrays of up to four high-quality loudspeakers and up to four control microphones distributed around an artificial head and torso in a lightly damped listening room. The digital signal processing engine consists of four floating point processors interconnected for parallel computation, together with digital and analog converters. The performance of the system is analyzed in terms of the tradeoff between the maximum sampling rates that can be used and the complexity of the digital filter matrices in the different operating modes of the system such as system identification, single-input multiple-error adaptive filtering, and full multichannel filtering with fixed coefficients.
Trends in modeling of structural-acoustics systems with structural complexity in low- and medium-frequency ranges
Authors: Soize Christian
Published: July 2002

This paper gives a comprehensive survey of a theoretical approach for predicting the frequency response functions of general complex structural acoustics systems (CSAS) in the low- and medium-frequency ranges. For such a CSAS, the complex structure is made of a master structure coupled with internal substructures presenting structural complexity and corresponding to a large number of secondary subsystems, such as equipment units or internal secondary structures, attached to the master structure. The master structure is a general 3-D dissipative structure with an arbitrary bounded geometry and is made of an anisotropic, inhomogeneous, viscoelastic medium. This complex structure is coupled with an external acoustic fluid via the master structure. The approach is based on the use of fuzzy structure theory, introduced by the author in 1985, including recent developments concerning identification of the fuzzy substructure model parameters. This theory allows the effects of the internal structural complexity on the master structure to be modeled. In addition, the construction of an intrinsic reduced model of the frequency response functions of this CSAS adapted to the medium-frequency range (recently proposed by the author) is presented, which allows simplifications in the calculation of the responses to any deterministic or random excitations.
Vessel avoidance of wintering Norwegian spring spawning herring
Authors: Vabo/ Rune , Olsen Kjell , Huse Ingvar
Published: July 2002

Vessel avoidance during acoustic abundance estimation of herring was investigated. Echo energy values measured from a submerged 38-kHz transducer were recorded and values at the time of passage were compared with corresponding echo energy values from a hull-mounted 18-kHz transducer on the passing vessel. A quantification of the loss in echo energy due to avoidance behavior is presented. Experiments are grouped into nighttime and daytime experiments. While a considerable drop in echo energy at the time of passage was observed at night, insignificant loss was found during the day. A general avoidance reaction pattern was identified in all the nighttime experiments. The magnitude of the vessel avoidance reaction was found to be strongly dependent on depth. Between 40 and 100-m depth some nighttime experiments showed loss in echo energy of the order of 90%. The effect of avoidance behavior decreased with depth below 100 m, but was seen down to 150 m at night. Low daytime response is due to deep daytime herring distribution. Body tilting and vertical and horizontal swimming are suggested explanations for the lower echo energy during passage.
The digital transducer, new sonar technology
Authors: Acker William
Published: July 2002

The new digital architecture in sonar technology designed by BioSonics, Inc. has many advantages when compared with traditional analog technology. The echo signal is digitized inside the transducer at a very low signal level. This removes problems of coupled or induced noise when sending the signal through long lines. The system operates with a 3 to 4 dB noise figure in all modes of operation. Another important feature of the system is very high dynamics (>132 dB continuous). This allows receiving of low signals from small targets (i.e., plankton) simultaneously with high level signals (i.e., bottom) without saturation. The echo signal is sampled at a high rate and stored with high accuracy (0.013%) and high resolution (1.8 cm or less). The sonar is PC controlled for simple operation. The large dynamic range means no adjustment in system gain or transmit power. The data acquisition system is virtually automatic because most decisions are in postprocessing. Raw data is stored on the PC’s hard drive or similar device. The sonar is available in a wide frequency range (38 kHz–1 MHz) and single, dual, or split beam transducer configuration depending on the application, i.e., biomass, layering effects, bottom typing, etc.
An acoustic tag ‘‘radar-type’’ tracking system for fish behavior studies near structures
Authors: Hedgepeth John , Fuhriman Dave , Johnson Robert , Geist David
Published: July 2002

Recent studies of fish behavior around hydroelectric dams have used acoustics with split-beam methodology. A complementary methodology called the tracking transducer takes advantage of split-beam capabilities for expanding fish behavior investigations. The principle of tracking radar, aligning the antenna beam with a target, was applied with an acoustic transducer and dual-axis rotators for tracking individual fish over long periods of time. Deviation of the target from the beam axis produces a correction to point the axis toward the target. Two of these tracking systems have been used to triangulate the position of a small acoustic transmitter implanted in salmonid fish in the forebay of Lower Granite on the Snake River. This paper describes the system design, development, and early implementation. In October 1997 a 6-mm-diameter 200-kHz acoustic tag was placed in a rainbow trout and fish movement was measured near a surface bypass collector. The tracking equipment consisted of two 201-kHz, 6 deg half-power, full-beamwidth transducers placed 12.2 m apart and 3.0 m below the water surface. Each split-beam transducer was mounted on a high-speed, dual-axis, motorized armature. These armatures were computer controlled and the system automatically followed the tag by using angle estimates provided by the echosounders.
Noise characteristics of Japanese fisheries’ research vessels
Authors: Takao Yoshimi , Sawada Kouichi , Miyanohana Yoichi , Okumura Tsuyoshi , Furusawa Masahiko , Hwang Doojin
Published: July 2002

Noise is one of the important problems for hydroacoustic surveys of fisheries resources, because it may cause a large error in estimated results and it may shorten detectable range. It is necessary to estimate and reduce the contribution of acoustic noise for precise and accurate surveys. Noise is classified into low-frequency noise (audible by fish) and high-frequency noise (affecting echo sounders). High-frequency noise usually results in an overestimate of acoustic return. The average power of the high-frequency noise was measured by the echo-integrator. The echo-integrator output was converted into the equivalent noise spectrum level which can be compared with noises received by other quantitative echo-sounding systems or environmental noises. The noise measurements of several fisheries research vessels, ranging from 29.5-m length (RV ‘‘Taka Maru’’) to 93-m length (RV ‘‘Kaiyo Maru’’), were conducted to know their characteristics of noise. The dependence of noise upon the ship speed and the screw propeller’s setting was investigated for each vessel. The vessel’s performance for hydroacoustic survey is discussed from a viewpoint of noise level.
Can SWL-induced cavitation and renal injury be separated from SWL-induced impairment of renal hemodynamics?
Authors: Evan Andrew P. , Willis Lynn R. , Connors Bret A. , McAteer James A. , Lingeman James E. , Cleveland Robin O. , Bailey Michael R. , Crum Lawrence A.
Published: July 2002

SWL to one kidney causes localized tissue damage and impairment of tubular function, but reduces renal plasma flow (RPF) in both kidneys. We examined the effect of SWL voltage (kV) and inversion of the waveform (IW) on these localized and bilateral effects of SWL. Five-week-old pigs were anesthetized for either sham-SWL or SWL (2000 shocks, unmodified HM3) at 12, 18, or 24 kV or 2000 shocks, 24 kV, with a reflector that inverts the waveform. RPF and tubular extraction of PAH (EPAH) were measured 1 h before and 1 and 4 h after SWL. EPAH estimates tubular secretion function. SWL significantly reduced RPF to similar degrees at each kV. EPAH was not significantly reduced in the 12-kV group, but was reduced to progressively greater degrees in the 18 and 24 kV groups. IW eliminated ultrasonic evidence of cavitation, produced minimal tissue damage, and eliminated the reduction of EPAH. It did not eliminate the reduction of RPF. The data suggest that shock-wave voltage and cavitation may be related to the tissue injury and reduced EPAH induced by SWL, but suggests that neither may be directly related to the impairment of RPF. [Work supported by NIH, PO1 DK43881.]
SWL cavitation damage in vitro: Pressurization unmasks a differential response of foil targets and isolated cells
Authors: McAteer James A. , Stonehill Mark A. , Colmenares Karin , Williams James C. , Evan Andrew P. , Cleveland Robin O. , Bailey Michael R. , Crum Lawrence A.
Published: July 2002

To better understand the role of cavitation in kidney stone comminution and renal injury pressurization (Delius UMB23:1997) to regulate cavitation bubble activity in vitro was used. A low acoustic impedance chamber (8×15 cm) that only minimally altered lithotripter shock-wave pressure and waveform was constructed to house targets (aluminum foils or vials of kidney epithelial cells) under pressure (0–1400 psi). Cells exhibited lytic injury and foils sustained pitting at atmospheric pressure, but high pressure (1400 psi) prevented lysis and pitting. The pressure threshold for reduced damage was dramatically different for cells versus foils. Cell lysis was prevented at very low pressure (> 30 psi), while substantially greater pressure (∼600 psi) was needed to prevent pitting. Modest pressure (25–300 psi) actually enhanced pitting. SWL in vitro cell lysis and foil pitting are both likely due to cavitation. However, interactions between cavitation bubbles and these two targets appear to be quite different. Our observations are consistent with those of Delius and support his suggestion that it may be possible to reduce cavitation-mediated cell damage by regulating ambient pressure. The finding that slight pressure increased the number of cavitation events on foils suggests that modest overpressure might enhance stone comminution. [Supported by NIH PO1-DK43881.]
Physics of the Music Laboratory at the University of Washington
Authors: Chaloupka Vladimir , Hughes Keith
Published: July 2002

The Physics Department at the University of Washington recently completed the acquisition of new equipment for teaching the Physics of Music course, as well as for performing various independent study projects in musical acoustics at both undergraduate and graduate levels. The lab setup will be described, as well as the experience after the first year of operation. Many aspects of the Physics of Music provide excellent opportunities for involving undergraduate students in small, but meaningful and interesting, research projects. Specific examples will be discussed.
Low-frequency noise around airports
Authors: Vallet Michel , Bruyere Jean Claude
Published: July 2002

The sound frequency weighting PNdB has been proposed by Kryter (1959) to assess the noise from jet engine aircrafts. Due to the technical evolution of aircraft engines, and to choose a frequency weighting common to all transport systems, the dB(A-weighted) has begun to be used as the acoustical unit around airports; the PNdB is still used for noise certification purposes. A study of noise measurements that was carried out around French airports allows one to see some noticeable differences between acoustical levels, respectively, expressed in dB(A) and dB(C). The difference varies between 4 dB for the landing of recent aircrafts to 14 dB at take-off for 50/70-seat propeller aircrafts. If one considers the additional frequency decrease due to the walls and windows in housing, the pertinence of dB(A) weighting could be questioned for aircraft noise received by people living a long distance from the airport. This question is of interest in light of the new ANSI method for assessing combined noises (Schomer, 1997): this method suggests using the C frequency weighting for sounds with strong low-frequency content.
Converting noise level into number of noisy events: A new method to quantify k factors reveals area-dependent over- and underenergetic responses around the same airport
Authors: Kalveram Karl Th.
Published: July 2002

For aircraft noise, caused by N flight movements within observation time T, with average maximum noise level Lmax and duration dt, the energy equivalent noise level Leq3 is approximated by Q(k,d)=Lmax+k* log N+d* log(dt/T), if k=d=10. This seems to enable a trade-off between Lmax, N, and dt. However, the problem is not to keep the Leq3 constant, but the subjective annoyance, and this in turn depends on whether people process alterations of Lmax, N, and dt corresponding to energy equivalence (k=d=10) or not. Therefore, valid measurements of public noise responses by Q(k,d) require the estimation of k and d from empirical data. However, multiple correlation techniques mostly used for this purpose lead to contradictory and invalid estimates for k. This paper presents an alternative method, utilizing dose-response curves based on regression of annoyance with Lmax at fixed numbers of overflights. Results are independent from the selected annoyance ratings and furthermore also valid for nonlinear dose–response curves. Based on curves characterizing small (N<5000), medium (50000<N<200000), and large (20000010 for low k and k<10 for high noise levels, if this airport changes from medium to large.
Noise in non-premixed turbulent syngas flames
Authors: Klein Sikke A. , Kok Jim B. W.
Published: July 2002

A turbulent syngas flame may generate acoustic noise of high acoustic intensity in a combustion chamber. This may lead to the failure of construction components in a gas turbine engine in periods of the order of 1–100 hours. The research as described in the literature has almost exclusively been performed on the generation of noise in premixed methane or propane flames. Syngas fuel is a mixture of hydrogen and carbon monoxide, and the burners used are of the non-premixed type. In this research, the effect of turbulence and syngas composition on noise generation is investigated. A laboratory is set up to test syngas flames of a thermal power of 50 kW in a cylindrical air-cooled combustion chamber. Experiments are performed at several fuel compositions and burner inlet conditions. The flame sound intensity is measured in the combustion chamber equipped with acoustic dampers. The paper discusses the measured sound spectra. A model is derived for the generation of sound in a turbulent non-premixed flame. In this model it is shown that the sound generation is related to the dependence of density on mixture fraction in a flame with fast chemistry. A fluctuation in mixture fraction will lead to sound generation.
Visualization of a phase structure of sound field in a Bragg cell
Authors: Goncharov Vadim , Ilchenko Leonid , Moskalev Vladimir , Smirnov Eugene , Yurchenko Alexander
Published: July 2002

A phase structure of a sound field in a Bragg cell was investigated experimentally using a dual-beam, dual-frequency laser system. The technique used allows the recording of a spatial phase structure of diffracted light field to visualize phase nonuniformities of a sound field. The underlying physical principle of this technique is to probe an analyzed phase object using two slightly separated in space light beams with different optical frequencies. A difference frequency signal, which is obtained as a photodetector output in an optical heterodyne system, contains information about a phase difference of the adjacent areas of the sound field. When the probing beams are diffracted by a Bragg cell, their optical frequencies are shifted by the same acoustic frequency, so that the difference is the same at any Bragg cell operating frequency. A Bragg cell on paratellurite operating around 100 MHz was investigated. Focused laser beams with a difference of optical frequencies equal to 1 MHz were used as probing beams. Pictures were obtained which demonstrated dependencies of a phase structure of diffracted light on frequency and sound power. Phase nonuniformities across the sound field up to three wavelengths have been estimated. [Work supported by EOARD.]
Perceptual consequences of amplitude perturbation in the wavelet coding of speech
Authors: van Schijndel Nicolle H. , Houtgast Tammo , Festen Joost M.
Published: July 2002

Wavelet coding of sound is a frequency–time representation that may be able to mirror the properties of auditory coding at a peripheral level, provided that we use the proper so-called mother wavelet. Previous experiments based on intensity discrimination [van Schijndel et al., J. Acoust. Soc. Am. (submitted)] suggested a Gaussian-shaped wavelet with a bandwidth of 1/3 oct. Based on this function, a wavelet coding and reconstruction algorithm has been developed. The frequency–time plane is split up: one wavelet per three periods along the time axis and nine wavelets per octave along the frequency axis. In this study the amplitude of the wavelet coefficients is manipulated before the stage of reconstruction. The perceptual effect of amplitude perturbation on speech perception is studied. A (partial) answer to the following questions will be presented: (1) what degree of amplitude perturbation is just noticeable to the subject, and (2) how is speech intelligibility affected by suprathreshold amplitude perturbation. The results may provide insight into the acuity of the auditory amplitude coding and its effect on speech intelligibility. [Work supported by the Netherlands Organization for the Advancement of Pure Research (N.W.O.).]
Application of auditory models to discrimination thresholds of voicing parameters
Authors: Campbell Shari L. , Arehart Kathryn H. , Scherer Ronald C. , Butler Amy N.
Published: July 2002

A recent study [Scherer et al., J. Voice (to be published)] reported discrimination thresholds for synthesized complex tones emulating glottal flow waveforms and output pressure waveforms for the vowel /a/. Each signal type was varied along both an open quotient and a speed quotient continuum using standard values associated with normal voice production. jnds for the output pressure signals were significantly larger than for the glottal flow signals. The current experiment compared quantities derived from ‘‘auditory-perceptual’’ representations of the acoustic signals. These quantities (centers of gravity and Euclidean distances) were computed for both the raw amplitude spectra and for modeled excitation patterns and specific loudness patterns. Overall loudness and overall sharpness were also estimated. The comparisons indicated ways in which auditory modeling can enhance our understanding of basic issues in voice and speech perception. [Work supported by UGA College of Education.]
Audio signals in domestic appliances evaluated in terms of the hearing ability of older adults
Authors: Kurakata Kenji , Kuba Yasuyoshi , Kuchinomachi Yasuo , Matsushita Kazuma
Published: July 2002

The audio signals used in domestic appliances currently available on the market in Japan were recorded to identify suitable signals for the hearing ability of older adults. The results of the analysis indicated the following three problems: (1) Some appliances use high-frequency tones around 4000 Hz. Since these sounds are hard for older adults with presbycusis to hear, it would be better to use signals with lower frequencies. However, the problem here is that if the frequency is lowered, then the signal might be masked by domestic sounds whose power would be greater relative to the low-frequency signals. (2) The signals used by some appliances are too soft. The intensity of some signals should be adjusted to compensate for the hearing loss among older adults. (3) The sounds used in these appliances are often very similar in terms of both timbre and temporal ringing patterns. That may cause confusion because it is difficult to identify which appliance is signaling.
Further effects of cochlear loss on ICP loudness adaptation, and loudness constancy
Authors: Ludwig Rebecca Schnieder , Dou Hongwei , Weiler Ernest M. , Kretschmer Laura W. , Sandman David E. , Stromberg Eleanor
Published: July 2002

ICP loudness adaptation depends on an ipsilateral comparison paradigm with a short increasing or decreasing referent value to reveal loudness adaptation. Janson et al. [Br. J. Audiol. 29, 288–297 (1996)] found the intensity-based function for ICP loudness adaptation for those with a high-frequency loss differed significantly from normal listeners. Loss of loudness constancy in normal listeners was consistent, loss in those with a cochlear loss showed a negative progression as intensity increased, as if it were the inverse of recruitment. The purpose of the present study was to compare ICP adaptation to recruitment. ICP adaptation and recruitment were tested in ten high-frequency loss (probably noise induced) and ten age-matched normal hearing individuals. Although the evidence for recruitment was generally found in those with hearing loss, there was no simple relationship to ICP adaptation. Variation among measures of recruitment, implications of abnormal loudness constancy for speech perception, and the possible use of measures of otoacoustic emissions for future studies will be discussed.
Effects of naturally occurring external loads on jaw movements during speech
Authors: Shiller Douglas M. , Gribble Paul L. , Ostry David J.
Published: July 2002

The achieved positions of speech articulators are determined by a combination of intrinsic forces due to muscles and extrinsic forces due to gravity and to head and body motion. If not compensated for, changes in these extrinsic forces will affect articulator positions in behaviors such as speech. In this paper, the extent to which speakers adjust control signals to jaw muscles to compensate for the effects on the jaw due to changes in speaker orientation relative to gravity is assessed. Whether control signals are adjusted to compensate for loads arising during locomotion is also assessed. In one study, subjects produce speech utterances where head and body orientation relative to gravity are varied. In another study, subjects produce speech sequences during locomotion such that the timing of the step cycle applies the maximum load force during either the consonant related (jaw elevated) or vowel related (jaw lowered) phase of movement. The empirically observed patterns of jaw motion are compared with predictions of a model of jaw and hyoid motion which includes muscle properties, dynamics, and modeled neural signals. The hypothesis that during speech, control signals to jaw muscles remain unchanged with variation in these extrinsic loads is specifically tested.
The relationship between subjective preference for sound fields with vocal music and brain activity
Authors: Mouri Kiminori , Ando Yoichi
Published: July 2002

The purpose of this study is to reconfirm the theory that the subjective preference for music sound fields depends on its autocorrelation function (ACF) of the source signal by means of analyzing continuous α-brain waves. Ando’s theory describes that the most preferred delay time of early reflections and reverberation time are calculated by the effective duration (τe) of the ACF of music signals [Y. Ando, J. Acoust. Soc. Am. 74, 873–887 (1983)]. On the other hand, it was demonstrated that the τe of the ACF of the α-brain wave range (8–13 Hz) from the left and right temporal lobes was changed by temporal factors relating to the subjective preference of sound fields [Y. Ando and C. Chen, J. Archit. Plann. Environ. Eng. AIJ 488, 67–73 (1996); C. Chen and Y. Ando, ibid. 489, 73–80 (1996)]. When a sound field has a constant delay time as a usual condition, it is assumed that continuous α-brain waves may be affected sequentially by the different running τe values of the ACF of the source signal.
Bottom volume scattering—Modeling and data analysis
Authors: Li Dan , Frisk George V. , Tang Dajun
Published: July 2002

As part of the Acoustic Reverberation Special Research Program (ARSRP), low-frequency bottom scattering from a deep ocean sediment pond was measured using an omnidirectional source and a vertical line-receiving array deployed near the bottom. Sediment volume heterogeneities were found to be the major contributor to the measured scattered fields [Tang et al., J. Acoust. Soc. Am. 98, 508–516 (1995)]. Taking advantage of the experimental geometry, a model is developed which can generate realizations of three-dimensional random inhomogeneities in sediments and can simulate scattered fields based on these realizations using a perturbation approach. In this model, the broadband propagator (the Green’s function) is obtained using an exact numerical method. Therefore, the model can handle any layered sediment environments with scatterers distributed within any of the layers. The model predictions are compared to the experimental data across the available frequency band, and the results are favorable. [Work supported by ONR.]
Multistatic scattering from anisotropically rough interfaces in horizontally stratified waveguides
Authors: Lee Jaiyong , Schmidt Henrik
Published: July 2002

A model of multistatic scattering and reverberation from an anisotropically rough seabed in shallow water has been developed by combining a scattering theory based the method of small perturbations [W. A. Kuperman and H. Schmidt, J. Acoust. Soc. Am. 79, 1767–1777 (1986)] with a seismo-acoustic propagation model [H. Schmidt and J. Glattetre, J. Acoust. Soc. Am. 78, 2105–2114 (1985)]. This new computational model provides full 3-D wave theory simulations of multistatic rough bottom reverberation in horizontally stratified waveguides. The seabed and subbottom roughnesses are represented by anisotropic roughness patches covering the actual sonar footprint. The scattering theory convolves the incident sonar field with the roughness to provide a distribution of virtual source panels that are then fed into the propagation model to provide a numerically efficient azimuthal Fourier series of wave number integrals for evaluating the 3-D reverberant field. The efficiency of the new hybrid model enables the modeling of 3-D field statistics by Monte Carlo simulation, as well as time-domain field simulations by Fourier synthesis. The model has been verified by comparison with analytical solutions for the simple problem of determining the 3-D multistatic scattering strength of a randomly rough interface separating two fluid half-spaces. Here the new model is used to investigate the dependence of the multistatic reverberant field on environmental parameters such as anisotropic roughness statistics, and bottom stratification and composition. [Work supported by ONR.]
Classroom acoustics I: The acoustical learning environment: Participatory action research in classrooms
Authors: Hasell Mary Jo , Abbott Philip , Siebein Gary W. , Gold Martin A. , Lee Hee Won , Lehde Mitchell , Ashby John , Ermann Michael , Crandell Carl C.
Published: July 2002

This pilot study used participatory fieldwork in a number of kindergarten through eighth grade classrooms to evaluate the acoustic setting that supports learning. Dynamic interactions among administrators, teachers, students, parents, and research team members makes a difference in learning. It was found that classroom interaction depends on the social make-up, anticipated behavior, intellectual level, teaching method and theories, and hearing capacity of participants, as well as the physical characteristics of the classrooms. Participatory action research (PAR) resulted in the development of appropriate classroom observation tools utilizing checklists, summary scales, and hierarchial rating scales. The learning climate, lesson clarity, instructional variety, task orientation, and student engagement were used to identify variables for acoustical studies. The PAR was instrumental in determining the protocol for the acoustical measurement studies described in Classroom Acoustics II. Field observations began with an understanding of the adaptive behaviors of the classroom participants such as how the teachers modified their location and sometimes even the furniture to maintain short speaker to listener distances. The findings indicate a range of solutions are needed to improve the acoustical learning environment including innovative teaching methods, improving room acoustical design, reducing background noise levels, and, in some cases, using amplification systems.
Assessing speech intelligibility in classrooms at the University of Washington
Authors: Heerwagen Dean , Khiati Tarik
Published: July 2002

The University’s Committee on Accessibility has initiated an investigation of classroom acoustical conditions. This investigation has been undertaken in response to complaints made by University faculty members who state that they are experiencing difficulties in understanding student’s speech during class sessions, when students ask questions or engage in discussion. A survey of University faculty identified 75 problematic classrooms (out of the 300+ classrooms on the University campus). A sample of generally representative classrooms has been selected: these classrooms (and class sessions held in the classrooms) are being systematically measured to determine their acoustical properties. Seven parameters have been anticipated as the causes of inferior speech intelligibility: to strong background noise levels, too weak signal-to-noise ratios, excessive reverberation, too great speak-to-listener distances, various speaker idiosyncrasies (e.g., accenting, mumbling, rapidity, using unfamiliar verbiage, and so forth), acoustical defects of the classroom, and faulty amplification systems. To date, measurements show that the principal acoustical faults causing these intelligibility difficulties are high background noise levels, inadequate amounts and locations of absorptive materials and assemblies, and student speech characteristics including too-long distances separating faculty and students. In selected classrooms corrections are being undertaken.