bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic quality at workplaces
Authors: Kortchmar Lia , Vorländer Michael , Slama Jules G.
Published: July 2002

The pursuit of a methodology to elucidate the workplaces issue is a challenge within the area of acoustic quality. The matter involves the search for subjective attributes to be employed on rating acoustic quality judgements. In this search for evaluation strategies, it is appropriate to associate a physical property of the enclosure to a subjective feature. Objective measures offer an intermediate description between design and the subjective effect. The emphasis here is on the objective attributes significant for sound perception. This work looks at some practical experience gained from the connection between experimentally measured values and subjective sensations related to sound quality at workplaces. Information is crossed among the different techniques used for its evaluation. The qualitative features of extremely nondiffuse sound fields are examined by the analyses of the physical properties, objective measures, and subjective quality of various types of rooms. [This work has been carried out at the Institute of Technical Acoustics at the RWTH Aachen University in collaboration with the Federal University of Rio de Janeiro (COPPE/UFRJ).]
Directional hearing in small animals: Tympanal mechanics and evolutionary innovations
Authors: Robert Daniel
Published: July 2002

Tympanal hearing is an evolutionary innovation that occurred at least twice independently in flies (Diptera). Both in Tachinidae and Sarcophagidae fly families, hearing organs evolved that are located on the anterior thorax and that present two thin tympanal membranes rigidly connecting to a pair of mechano-receptive sensory organs. Remarkably, the small interaural distance (1 mm) constitutes a serious challenge for directional hearing at 3–6 kHz. The mechanical response of these ears to incident random noise was investigated by microscanning laser Doppler vibrometry. Mechanical transfer functions of tympanal displacements show that the ipsi- and contralateral tympana vibrate with an unexpectedly large amplitude difference and interaural delay [R. N. Miles et al., J. Acoust. Soc. Am. 98, 3059–3070 (1995)]. The deflection shape analysis shows that in both cases the tympanal membranes are mechanically coupled across the animals midline. However, morphological and biomechanical evidence indicates that the process of mechanical coupling differs between both families, indicating the presence of two alternative and distinct mechanisms for directional sensitivity. This biomechanical and morphological study exemplifies how evolutionary innovation can generate original and specific solutions to the general problem of directional auditory processing. [Work supported by NIH, NSF, and the Swiss Science Foundation.]
Use of acoustics in large tuna trophic-habitat characterization in French Polynesia
Authors: Bertrand Arnaud
Published: July 2002

Tuna have a high metabolic rate, yet they often live in regions characterized by low primary productivity rates. Improving knowledge on tuna-habitat trophic relationship is of main importance for understanding tuna distribution and catchability. Distribution of large tuna food webs, i.e., micronekton, was described in French Polynesia at depths up to 500 m using acoustics and pelagic trawls. At large scale, acoustics profile response, morphological characterization of acoustic structures, and quantitative descriptors of sound scattering variability were used. Due to water masses advection and oxygen limitation main micronektonic production was localized 8 degrees south of the equatorial upwelling. At a smaller scale, micronekton was classified into seven scattering type: gas-filled swimbladder fish, absent swimbladder fish, gas-bearing invertebrates, cephalopods, fluidlike crustacean, fluidlike jelly-fish, and elastic-shelled invertebrates. Qualitative and quantitative aspects of the scattering structures were also described. Then scattering structure composition and the tuna trophic interest of aggregations was determined.
Multi-frequency measures and models of Lake whitefish (Coregonus clupeaformis) backscatter from Lake Michigan
Authors: Jech J. Michael , Horne John K. , Powell Lee A. , Grandt James H.
Published: July 2002

To improve chances for discriminating and identifying fish and zooplankton species, more acoustic information is better. Multifrequency, digital echosounders increase information bandwidth by increasing the number discrete frequencies transmitted and received through multiple transducers. Multifrequency data provide a range of aural perspectives just as colored spotlights accent different components of a painting. Lake whitefish (Coregonus clupeaformis) represent an increasing biomass in Lake Michigan and have a disproportionately large, single-chambered swimbladder relative to the body volume. Frequency-dependent, echo amplitude predictions from a Kirchhoff-ray mode model were compared to in situ backscatter measurements of constrained and free ranging lake whitefish at five discrete frequencies (38, 50, 120, 200, and 420 kHz). In addition to geometric scattering, backscattering characteristics of whitefish in the resonance region were also modeled. This combination of theoretical model prediction and multifrequency measures allows the quantification of the importance of intraspecies backscatter variability due to fish sizes and behavior. [Work supported by ONR.]
Silicon microphones: An overview
Authors: Fischer Marc , Sessler Gerhard M.
Published: July 2002

Over the past few years, silicon micromachining has become a well-established technology. This development was promoted by the expectation that new types of sensors and actuators can be fabricated at low cost. Very early, attempts were made to realize acoustic sensors by the use of micromachining. Only now, the first commercial silicon microphones are available. This paper presents an overview of the technology and discusses different principles to build silicon microphones. There are acoustic sensors, e.g., the capacitive silicon microphone, which are designed by downscaling classical transducers. As opposed to this, there are new types of sensors, making use of novel transduction principles, or of the possibility to integrate active elements on the silicon substrate. The focus of this paper is on the following principles: the capacitive microphone in either single-chip or two-chip design, the piezoelectric microphone with organic or ceramic piezolayer, the piezoresistive microphone utilizing monocrystalline or polycrystalline silicon, the optical microphone based on modulating light propagation in a micromechanical waveguide, and the FET microphone, consisting of an acoustically modulated field-effect transistor. Each transducer will be discussed briefly, including the typical technological process necessary for its design.
Characterization of small-scale seafloor roughness using close-range digital photogrammetry
Authors: Lyons Anthony P. , Fox Warren L. J. , Pouliquen Eric , Akal Tuncay
Published: July 2002

Surface roughness is a fundamental property affecting a variety of physical phenomena including sediment transport and the interaction of acoustic energy with the seafloor. Characterization of bottom roughness and its dynamics is therefore essential for understanding and quantifying the influence of the sediment microtopography. Extensive field measurements of bottom roughness have been taken recently with an end-to-end digital photogrammetry system providing quantitative, two-dimensional surface roughness measurements on spatial scales of approximately a millimeter to a meter. Results of these measurements have shown that sediment surfaces in shallow water are often anisotropic and/or exhibit non-Gaussian height distributions, both of which have the potential to strongly affect seafloor acoustic scatter. For these kinds of surfaces, simple roughness parameters such as rms height or the slope and offset of a power-law representation of the power spectra will not give a sufficiently complete description. Two-dimensional statistical models are needed to capture the anisotropic nature of sediments with oriented features, while for seafloors with peaked forms, it is the phase information in the frequency domain that is required, as this controls the shape characteristics of a surface. Characterization of seafloor roughness based on these ideas will be presented using results from the digital photogrammetry system.
Measurement of the effective surface impedance of rough surfaces in laboratory and in situ
Authors: Nocke Christian
Published: July 2002

The normal specific acoustic impedance fully characterizes the acoustic behavior of a surface. In this contribution, laboratory and in situ results of a novel measurement procedure for this impedance are presented. The acoustic propagation above the surface is most sensitive to the acoustical properties of the surface at small angles of incidence. This effect is utilized to deduce the impedance from measurements of the interference between the direct and reflected sound of a point source above the surface under investigation. For homogeneous and plane surfaces, the absorption coefficient calculated from these impedance values compares well with the results from tube measurements. For rough and/or inhomogeneous surfaces, the measured impedance results must be interpreted as effective values. The reliability of the method has been proved in laboratory measurements for a frequency range between 80 and 4000 Hz. Furthermore, results of the new method from room acoustical applications are presented.
Response to air-cooling machinery noise: A preliminary study
Authors: Franchek Nancy , Davies Patricia , Wood Richard , Lavrich Phil
Published: July 2002

People living close to commercial properties are often exposed to air-cooling machinery noise. Of primary concern is the awareness of tones in the sounds and how that awareness contributes to the overall annoyance experienced by people exposed to the noise. For noise control, it is most straightforward to control the strength of tonal contributions from components such as fans and compressors. For the class of machinery considered in this study, fan fundamentals are usually around 60 Hz and compressor fundamental frequencies are typically between 500 and 700 Hz. A preliminary study was carried out using 27 sounds that simulated a broad range of possible noises made by this type of air-cooling machinery. The sounds were presented to people who were asked to rate them in terms of tonality and annoyance. Tonality and annoyance ratings only increased when sounds contained at least one compressor tone above a certain threshold level. High correlations were found between objective metrics and the average of the subjects’ responses. Recommendations for future research include investigations into the individual effects of changing spectral balance, overall loudness, and tonal contributions for these types of sounds, because changing tonal levels alone affects all of these sound characteristics simultaneously.
Aspects of global assessment of loudness-equalized interior car sound in stationary driving conditions
Authors: Hempel Thomas
Published: July 2002

In listening tests, loudness still turns out to be the most relevant parameter for a subject’s assessment regarding interior car sound. In order to investigate further parameters in listening tests, the influence of loudness differences between the sound samples has to be avoided. Minimizing errors, this can be done by equalizing the loudness of presentation of each sample to the median of all samples in the set. It can be found that originally soft cars now yield lower rankings of preference than before. A possible reason might be the loudness of presentation, which is higher than in the real car. A presentation of sound samples, which now are equalized to the loudness originally appearing in those soft cars, could possibly lead to different rankings. Furthermore, global assessments of loudness-equalized interior car sounds were obtained using the method of paired comparisons as well as the method of magnitude estimation.
Psychoacoustic evaluation of noise emissions
Authors: Fastl Hugo
Published: July 2002

While the psychoacoustic evaluation of noise emissions is already on its way from basic research laboratories in universities to application labs in industry, the psychoacoustic evaluation of noise emissions is not yet widely applied [http://www.mmk.e-technik.tu-muenchen.de/admin/noise.html]. Therefore, psychoacoustic evaluations of noise emissions from road traffic noise, railway noise, aircraft noise, industrial noise, and leisure noise are discussed. As a rule, the overall loudness of a noise emission is larger than the average of the instantaneous loudnesses of the acoustic events constituting the noise emission. With respect to physical measurements, statistical treatment of loudness values measured according to DIN 45 631 can predict the subjective evaluations of noise emissions for different types of noises. [Work supported by DFG.]
A new HRTF decomposition and the corresponding algorithm for sound spatialization
Authors: Dudouet Eric , Martin Jacques
Published: July 2002

Binaural simulations make extensive use of head-related transfer functions (HRTF), so spatializers have to deal with large amounts of data and long computing times to interpolate the HRTF database over the whole space. A better understanding of human auditory localization could lead to a better simulation of localization cues and allow the database and the computing time to be minimized. Principal component analysis on HRTFs, reported by D. J. Kistler and F. L. Wightman [J. Acoust. Soc. Am. 91, 1637–1647 (1992)], gave only second-order decorrelated base vectors. In order to find simple vectors (bandpass or low-pass), an independent component analysis has been computed for upper-hemisphere preprocessed HRTFs (reported in by M. Emerit et al. [Proc. 15th ICA, Trondheim, Norway, (1995) pp. 437–440]). New analyses have been done on a new HRTF data set, on different representations (time, magnitude alone and with phase). These analyses give a small number of base vectors, which can be used to reconstruct any HRTF, with a correlation coefficient near 1, and allow a new spatialization method using 3 IIR filters, derived from independent impulse responses, to be conceived. Source signals are filtered by each of these filters, and linear combination of the results gives the spatialized signal.
Amplitude dependence of down-converted acoustic waves due to a nonlinear force interaction
Authors: Dupas E. , Behme G. , Kulik A. , Burnham N. A. , Chilla E. , Froehlich H.-J.
Published: July 2002

The interpretation of atomic force microscopy (AFM) data suffers from the lack of a detailed knowledge of the nonlinearity of the tip-sample force interaction. The AFM output signal as a function of the amplitude of two independent acoustic oscillations has been experimentally investigated in order to determine the interaction’s power law. Two complementary experiments were performed: (1) Two separate 500-kHz acoustic waves were propagated within the cantilever and through a cleaved mica sample; (2) 200-MHz surface acoustic waves (SAW) excited surface oscillations on a quartz sample in contact with the cantilever. In both experiments, the signals were amplitude modulated at a low frequency in order to record the dc mixing term. The cantilever vibration amplitudes at the modulation frequency, as well as at the difference frequency, were recorded using a lock-in amplifier. The experimental results have been compared with a physical model assuming a nonlinearity of the form F(1/d)n, with F being the force, d the penetration depth, and n lying between 0 and 5. This model predicts an output signal depending approximately quadratically on each of the oscillation amplitudes. This approach to force interaction analysis will be evaluated.
Double focus technique for quantitative measurements in time-resolved acoustic microscopy
Authors: Haenel Volker , Kleffner Bernhard
Published: July 2002

Simultaneous measurement of sound velocity and thickness of thin specimens is a well-known technique in time-resolved acoustic microscopy. However, the applicability is restricted, because a substrate in contact with the back of the sample is required. Furthermore, the substrate must have an area without coating, because a reference signal from the substrate’s surface is needed. In order to overcome these performance limitations, a double focus technique has been developed. By focusing the lens on the front, and afterwards on the back of the sample, only signals from the sample itself are required. Based on a simple geometrical model, the relevant acoustical equations for sound velocity and thickness were deduced analytically. Afterwards the method was applied to PVDF film in order to experimentally examine the feasibility of the approach. The measured results were compared with reference values and showed reasonable agreement. Based on these experiments, the influence of factors limiting the precision of the model, for example the spherical aberration, will be discussed, yielding an approach for an improved model. Eventually, the significance of the results for applying the method to manufacturing processes of thin films or foils, especially for online process control, will be described.
Studying drop cluster and array behavior using an acousto-electric levitator
Authors: Zheng Yibing , Apfel Robert E.
Published: July 2002

An acousto-electric levitator is developed for studying the formation and behavior of drop clusters and arrays. The device enables us to form suspended 2-D drop clusters and arrays by using electric and acoustic fields to levitate and manipulate drops. The spacing and the size of drops can be easily controlled by varying electric and/or acoustic field intensities. Experiments have been done on single- and multi-component drops, such as octane and dodecane, which are primary materials in natural fuels. Studies are focused on the evaporation rate of each drop due to the interaction in the array. The acoustic frequency has been increased from 28 kHz to more than 150 kHz so that generation and control of smaller drops, associated with spray combustion, are possible. The formation process of clusters and arrays, and the force balance among drops are analyzed. A theoretical calculation based on the diffusion-controlled model is compared with these experimental results. This high-frequency acoustic levitator is also used to study solid particle clusters whose size is around 10 μm by using a mixture of solid particles and liquid, and then letting the liquid evaporate. [Work supported by NASA through JPL Grant No. 958722 and NASA Grant No. NAG3-2147.]
Ultrasonic cell micro-manipulation and separation
Authors: Coakley W. T.
Published: July 2002

The manipulation of small (<5-μm-diam) eukaryotic cells and of bacteria (ca. 1.5-μm diameter) has been examined here in a number of batch and flow systems containing ultrasonic standing waves. The The different systems, involving plane and tubular ultrasonic transducers and a range of acoustic pathlengths, have been designed to treat suspension volumes of analytical scale, i.e., 5 ml to 50 μl for both single-sample and ‘‘on-line’’ situations. The frequencies employed range from 1–12 MHz. Removal efficiencies of 99.5% have been achieved for yeast and for the bacterium Escherichia coli. The influence of secondary cell–cell interaction forces in determining the cell concentration dependence of harvesting efficiency in batch sedimentation systems is considered. The contributions of pulsing and of field modulation techniques to achieving high harvesting efficiencies are illustrated. Applications of standing wave radiation forces to (i) levitate cells in suspension for microscopic examination of cell interactions; (ii) to enhance particle agglutination immunoassay detection of cells or cellular products, and (iii) accelerate the aqueous biphasic separation of cells are presented. [Work supported by BBSRC and EU.]
Stimulating sonoluminescence
Authors: Holzfuss Joachim , Rüggeberg Matthias
Published: July 2002

Single-bubble sonoluminescence is characterized by the light emission of an acoustically driven bubble in a liquid. It is due to an enormous compression of the bubble volume, where a change in volume by six orders of magnitude is followed by a violent collapse. The question has been asked if the violence of the collapse can still be increased. In the theory of nonlinear dynamical systems, methods have been developed to control systems toward higher energetic response. As a complex system that is excited within a resonant cell is being dealt with, harmonic series expansions of the driving signal have been used. It is shown that already two-mode signals with optimized amplitudes and phases are able to boost sonoluminescence. The difficulties concerning multimode signals are analyzed experimentally and numerically. The effects of nonsinusoidal excitation concerning spatial and temporal bubble dynamics are presented. [Work supported by the DFG (SFB 185).]
Sonoluminescence and acoustics within nonlinearly driven cavities
Authors: Guenther Ralf
Published: July 2002

The field distributions inside a nonlinearly driven cavitation bubble are obtained as a function of the dynamics of the bubble radius. They represent a straightforward generalization of standing wave solutions within a one-dimensional cavity to nonlinearly driven cavities. By coupling the solution to the Rayleigh–Plesset equation, a bubble equation is obtained which includes heat transfer from the bubble to the liquid and the spatial variation of the pressure field. The range of validity of the perturbational solution is dependent on the nature of the bubble content, and is largest for monoatomic gases. The inclusion of the internal field distributions into the simulation of the bubble dynamics is possible with very small computational effort. Applications to sonoluminescence are discussed. [Work supported by the German Ministry of Education and Research.]
The stability of microbubbles and sonoluminescence
Authors: Augsdörfer U. H. , Evans A. K. , Oxley D. P.
Published: July 2002

Sonoluminescence (SL), the light emission of a micron-sized bubble of gas trapped in water by an acoustic field, is associated with the collapse of a bubble nonlinearly oscillating under the sound field. Recent advances have made it possible to trap a single bubble at the pressure antinode of a standing wave, where it collapses and reexpands with the periodicity of the applied sound and emits light with every cycle. A critical feature of SL is the spherical symmetry of the bubble. If the collapse is violent, irregularities in the spherical shape of the bubble may develop towards the end of the collapse. As a result, the bubble may get destroyed shortly after the collapse. Due to the microscopic size of the bubble, molecular fluctuations constitute an additional force to which the gas–liquid interface is exposed. This additional force causes irregularities in the spherical shape of the bubble. The stability of a bubble is discussed by analyzing results obtained from hydrodynamic calculations of the bubble dynamics and the fluid dynamical processes outside the bubble taking molecular fluctuations into consideration. Results of these calculations reveal a surprising stability behavior and are in good agreement with observed behavior in single-bubble SL experiments. a)ua@dmu.ac.uk
Monaural, cross-frequency coincidence detection as a mechanism for decoding perceptual cues provided by the cochlear amplifier
Authors: Heinz Michael G. , Carney Laurel H. , Colburn H. Steven
Published: July 2002

Basilar membrane (BM) response properties associated with the cochlear amplifier are consistent with broadened BM tuning as level increases from approximately 20 to 90 dB SPL or higher. Associated with broadened tuning are compressive BM magnitude responses and level-dependent phase changes. To evaluate the perceptual significance of the cochlear amplifier, the associated nonlinear response properties of the auditory nerve (AN) and physiologically realistic mechanisms for decoding the perceptual cues they provide were studied. Most AN fibers’ rate and synchrony responses to tones are saturated above 40 dB SPL; however, nonlinear phase changes with level are encoded in AN responses up to 90 dB SPL or higher. Broadened tuning produces an increase in the correlation between adjacent AN fibers. Monaural coincidence detection is a physiologically realistic mechanism that can extract nonlinear phase cues across AN fibers with similar characteristic frequencies over a wide dynamic range. Methods from statistical decision theory [Colburn, J. Acoust. Soc. Am. 54, 1458–1470 (1973)] were combined with an analytical AN model to provide a quantitative framework for evaluating the potential of a coincidence detector to explain human performance in pure-tone intensity and frequency discrimination. [Work supported by NIH Grant No. T32DC00038 and NSF Grant No. 9601215.]
Loudness fluctuation and temporal masking in normal and hearing-impaired listeners
Authors: Chalupper Josef
Published: July 2002

The model of fluctuation strength as proposed by Fastl is based on the level difference ΔL between the maximum and minimum of the temporal masking pattern of dynamically varying sounds. Compared to normal-hearing subjects this level difference can be strongly reduced in hearing-impaired listeners. Recent findings, however, suggest that perceived fluctuations in hearing-impaired listeners are magnified by recruitment. Based on fitted loudness fluctuations—measured by a categorical scaling procedure—for each subject a ‘‘uniform loud noise’’ was generated, having equal loudness across all critical bands. These noises were amplitude modulated with different rates and depths. In experiment 1, subjects had to judge ‘‘how strong (not how fast) fluctuates the loudness’’ by marking one of seven verbal categories. In experiment 2, the ΔL’s of the various noises were measured by determining the thresholds in the envelope’s maximum and minimum. Although measured level differences were markedly reduced in the hearing impaired, the perceived loudness fluctuations are nearly the same for normal and hearing-impaired listeners. These results support the idea that the model of fluctuation strength can be applied to hearing-impaired listeners by transforming the difference in level into the corresponding individual difference in loudness. [Work supported by Geers.]
Relation of long-term adaptation to the loudness-function slope
Authors: Hellman William , Hellman Rhona
Published: July 2002

The decline in loudness over time, defined as simple loudness adaptation, is modeled by introducing temporal factors into the nonadapted loudness function. This modification generates effective time-dependent intensity and threshold shifts. Model predictions for several frequencies are in accord with recent psychophysical measures [Hellman et al., J. Acoust. Soc. Am. 101, 2176–2185 (1997)]. In contrast to the time constant description of adaptation, the present method produces a direct connection to properties of the loudness function. A consequence of this connection is the demonstration that the increase in loudness adaptation with decreases in SL is linearly related to the loudness-function slope in log–log coordinates. Moreover, the good agreement between the measured and predicted loudness-adaptation data implies a consistency and stability in the use of numbers for loudness judgments.
Characterization of complex systems from inverse radiative transfer measurements
Authors: Burkhardt John
Published: July 2002

A technique is proposed for the characterization of complex systems using measurements of time-dependent backscattered acoustic intensity. Utilizing a statistical model of system disorder, radiative transfer theory is used to model backscattered acoustic intensity. An algorithm for the estimation of system characteristics from observed backscatter is developed and presented. The technique is tested by numerically simulating the backscattered acoustic intensity from a disordered acoustic half-space with prescribed characteristics. Inversely estimated system characteristics are then compared with known system parameters.
Semiactive damping of vibration in mixing or propelling systems
Authors: Vachtsevanos George , Wang Peng
Published: July 2002

Vibration damping for mixing or propelling systems is investigated in this research, through the use of semiactive control techniques. The mixing or propelling systems considered consist of a motor mounted on some support, a long shaft driven by the motor, and a propeller. Such sytems can be found in many engineering areas like chemical and materials engineering. Because the shaft is long and the propeller is working in liquids, occurrence of vibrations in such a system is inevitable. Damping of the vibration is critical to prolong the life of the motor and thus of the whole system. However, it is difficult to establish an accurate mathematical model for the operation of the mixing or propelling systems since they involve a great deal of material characteristics and structure dynamics. This paper proposes a dynamic wavelet neural network to model the relationship between vibration observations and operating parameters. This model is them employed to design a semiactive controller regulating the operating variables so that vibrations can be damped. Experimental and simulation results are presented. The experimental setup consists of an AC motor, a triaxial transducer, a data acquisition module, in addition to labview software from a Dual-400-MHz Pentium-II workstation.
Smart/intelligent vibration and shock control by active or semi-active means
Authors: Stein George J.
Published: July 2002

In this contribution, some results of investigation of an electro-pneumatic active vibration control system for use in driver’s seats will be given first. The system uses a common air spring as the force generator and either a flow control or pressure control proportional valve to generate the active force. The structures of the linear controller, employing both the feed-back ‘‘sky-hook’’-type additional damping and the feed-forward vibration control for both types of electro-pneumatic transducers, will be illustrated. Both systems, of different control structure, are capable of mitigation of the vibration exposure of the driver by 8 to 10 dB, measured by a standardized method. The theoretical possibility of use of active or semi-active vibration control systems for smart/intelligent vibration and shock control will be briefly discussed. The idea is to use already available active or semi-active means to rapidly modify the damping properties of a linear oscillatory system according to commands based on fast evaluation of input vibration and so intelligently adapt the structure to the character of vibration excitation.
Rules, measures, and corpora for speech synthesis
Authors: Sagiksaka Yoshinori
Published: July 2002

A speech synthesis system is characterized by three factors: control rules, evaluation measures for rule development, and speech corpora to quantify and test these rules. In traditional speech synthesis by rule, the latter two factors have not been clearly acknowledged as system building constituents. Unclear acknowledgment of these two factors has often made it difficult for researchers to develop and evaluate a system consistently with objective measures. At ATR, corpus-based speech synthesis was proposed more than a decade ago to cope with this deficiency. In this talk, research efforts in corpus-based synthesis are overviewed to show how these three factors have been studied. Through this introduction, it is stressed that both clear formulation of engineering problems and basic scientific investigation of human mechanism are quite important to build a system efficiently and to understand human characteristics properly in a scientific fashion. To figure out the future problems, research efforts combining subjective evaluation and objective measures will also be introduced on the control of temporal characteristics for speech synthesis.
Waveform models for data-driven speech synthesis
Authors: Macon Michael W.
Published: July 2002

Many ‘‘data-driven’’ models for synthesizing speech rely on concatenating waveforms extracted from a database. However, the number of perceptually important degrees of freedom in speech make it unlikely that enough data could be collected to cover all combinations of phonetic variables. By utilizing models that can transform the waveform in perceptually relevant ways, the space of acoustic features covered by the data can be expanded. The minimal requirement for such a model is parametric control of the fundamental frequency and duration. In addition to this, dimensions such as voice quality characteristics (breathiness, creak, etc.), phonetic reduction, and voice identity can be altered to expand the range of effects realizable from a given database. A few classes of models have been proposed to allow varying degrees of control over these dimensions. Trade-offs between flexibility, fidelity, and computational cost exist with each. This paper will describe common threads running through the best-known approaches, including handling of nonperiodic components and sensitivity to measurement errors. Although each of these approaches has its own merits, none has been proven to satisfy all the desired properties for synthesis simultaneously. The limits of some of the simplifying assumptions underlying these methods will be demonstrated, and areas needing improvement will be outlined.
Research in cross-language and second-language speech perception and production: Issues, trends, and future directions
Authors: Strange Winifred
Published: July 2002

Theoretical and methodological advances presented in the preceding papers of this special session will be summarized and critiqued. Suggestions for future research will be offered. The comments offered in this talk are intended to generate discussion by the participants and audience of the special session.
Estimation of modal eigenvalues from an arbitrary distribution of receivers in a planar array
Authors: Rajan Subramaniam D.
Published: July 2002

Acoustic propagation in shallow-water areas is strongly influenced by the acoustic properties of the sediment layers. Many approaches have been proposed for obtaining the acoustic properties of the sediment layers. A simple and powerful approach has been the inversion algorithms based on modal eigenvalues [Rajan et al., J. Acoust. Soc. Am. 82, 998–1017 (1987)]. The performance of this approach has been demonstrated in field experiments. In these field experiments, the data were obtained on a linear synthetic aperture horizontal array. In recent years there has been an effort to extend the procedure to cover wider areas so that the range-dependent properties of the sediment can be determined. A means of achieving this objective is to collect the acoustic data on a planar array of receivers from which the range-dependent properties of the sediment can be obtained. Schemes for extracting modal eigenvalues of the propagating modes from data acquired on a curvilinear track with nonuniform grid spacing will be presented and the performance of the schemes evaluated using synthetic and field data.
Acoustic measurement of Shakespeare’s Globe Theatre, London
Authors: Richardson Russell , Shield Bridget M.
Published: July 2002

In the late 1960’s, Sam Wanamaker initiated the reconstruction in London of the theatre built for William Shakespeare’s company of players in the late 16th century. The theatre was built in the traditional manner to be as faithful to the original as surviving documentation allows, and has a circular auditorium with an open roof. The theatre was opened for performances in the summer of 1997. An acoustic survey of the theatre has been carried out. Measurements have been made of reverberation time (RT), early decay time (EDT), clarity index (C50), definition (D50), and rapid speech transmission index (RASTI) within the theatre’s auditorium. Various source positions were chosen on the stage after consultation with voice coaches and actors at the theatre, and receiver positions were chosen to provide uniform coverage of all the major audience areas. The results of these measurements indicate that, despite the theatre’s open-air construction, the amount of early energy provided to most audience areas results in reasonable levels of speech intelligibility, although achieving an adequate speech sound level is more of a problem. This is clearly exacerbated by the theatre’s open-air design, and its position in central London under the main flight path to London’s Heathrow Airport. The reverberant characteristics of the auditorium are generally within acceptable limits for modern theatres.
Acoustical evaluation of auditoria in Taiwan built before 1975
Authors: Chiang Weihwa , Juang Zenjang
Published: July 2002

Investigations were conducted regarding the acoustical characteristics of existing auditoria built before 1975. Discussions of the historical evolution since Japanese occupation were followed by architectural and acoustical analysis of 21 cases. Although many auditoria built during Japanese occupation were not designed for musical performances, they are more reverberant but smaller than the ones built after World War II. However, 62% of the selected cases were constructed with windows with 4 mm or thinner glazing on sidewalls or around the upper perimeter. Analysis of two auditoria were made regarding the performances of room acoustics, background noise control, and sound system. The room acoustics in both auditoria are suitable for chamber music but are too dry and too loud for orchestra music. Average background noises reach NC40 with AC off and NC50 with AC on. Both sound systems could not give correct directions of sound images, although they provide acceptable output level (96 dB SPL) and speech intangibility (STI 0.57) in the 2000-Hz band. With growing concerns about energy preservation in Taiwan, the current study could provide useful information for renovating or remodeling an existing auditorium. [Work supported by the National Science Foundation of Taiwan.]
Perception of sweep sounds caused by reflective pulse train in a rectangular reverberation room
Authors: Kiyohara Kenji , Furuya Ken’ichi , Tanaka Masashi , Kaneda Yutaka
Published: July 2002

Elevated frequency sweep sounds were perceived when a pulsive sound (a handclap) was generated in a rectangular reverberation room. The mechanism of this phenomenon was investigated. First, reflection of the pulsive sound (a pulse train) was simulated using the image method. The sweep sounds were also perceived in the simulated pulse train. It was assumed that the pulse train, whose intervals decreased inversely in proportion to time, caused the sweep sound sensation. The existence of such a pulse train was confirmed in the simulated pulse train. Then, this phenomenon was investigated theoretically based on algebraic number theory. A pulse train whose intervals decrease inversely proportional to time becomes a pulse train of equal intervals when it is represented on the power of time (t2) axis. On the other hand, considering a cubic room for simplicity, reflective pulse sounds arrive at (l2+m2+n2)L2/c2 on the t2 axis (l, m, n: positive integer, L: room dimension, c: sound velocity). The number theory proves that (l2+m2+n2) covers almost all integers. In other words, reflective pulsive sounds arrive with equal time intervals of L2/c2 on the t2 axis. This algebraic regularity causes a listener to experience the sweep sound sensation.
Biophysics of peripheral auditory systems in three teleost fish
Authors: Hastings Mardi C.
Published: July 2002

Many types of auditory systems are found in the more than 20000 different species of teleost fish. These unique sensors may detect extremely small acoustic particle velocities as well as acoustic pressure. Auditory bandwidths may range from very low frequencies to ultrasonic regimes depending on the species. Several experimental studies conducted over the last 15 years have yielded data which indicate that auditory sensitivity, bandwidth, and source localization are primarily controlled by the biomechanics of the periphery. In these studies in vivo displacements of peripheral auditory organs were measured noninvasively in response to tones emitted from an underwater speaker. The measurements were then correlated with mathematical models formulated from the basic physics of each system. Results are presented for three different species, goldfish (Carassius auratus), oscars (Astronotus ocellatus), and shad (Alosa sapidissima), and compared with auditory capabilities determined in behavioral and physiological studies reported in the literature.
Cochlear fine structure in chinchillas
Authors: Long Glenis R. , Shaffer Lauren , Murphy William J. , Talmadge Carrick L.
Published: July 2002

Spontaneous otoacoustic emissions (SOAEs) and distortion product otoacoustic emissions (DPOAEs) were evaluated in both ears of nine apparently normal-hearing chinchillas selected because preliminary screening indicated that they had SOAEs in at least one ear. All chinchillas show DPOAE fine structure in both ears (even in ears without detectable SOAEs). The spacing of the SOAEs and the characteristics of DPOAE fine structure are compared with OAE measures (obtained using the same procedures) from two species of kangaroo rat (Dipodomys merriami and Dipodomys spectabilis) that have no SOAEs and no DPOAE fine structure, and human subjects who have fine structure. Similarities and differences between the species are used to evaluate this model of cochlear fine structures further [Talmadge et al., J. Acoust. Soc. Am. 98, 1517–1543 (1998)].
Modeling structure-borne sound transmission via pipe systems to building structures
Authors: van der Jagt G. Susanne , Martin Heiko J.
Published: July 2002

In residential buildings many people are annoyed by noise caused by their neighbors, including structure-borne sound caused by drinking water, sanitary, and other installations. In different building regulations, limits are set to the maximum allowable sound level caused by installations in a room. However, predictions of the sound level in the design stage are not possible. In this research, a framework for structure-borne sound transmission models for pipe systems, including source characterization, has been developed. Further, the application possibilities of existing calculation methods in the quantification of the sound transmission have been investigated. For example, the application possibilities of the finite-element method (FEM) and statistical energy analysis (SEA) depend on the element type, including dimensions and material (pipe, mounting, or building structure), the wave type, and the frequency area. By combining both methods, calculations for the whole audible frequency area seem possible. Some models are presented, together with calculated values of characteristic parameters. Measurements can be used to complete and validate the models, for example by using a measurement setup based on the plate method. The interaction between calculations and measurements will form the basis for future research. The future research setup is also presented.
Improved spectral resolution via the reassigned Fourier transform
Authors: Day Charles R. , Ainsworth William A. , Meyer Georg F.
Published: July 2002

A modified form of the Fourier transform, the reassigned Fourier transform (RAFT), uses phase as well as magnitude information. Its properties are shown to be superior for both pure-tone and speech signal analysis. The RAFT technique has the potential to deliver higher resolution spectrograms than the fast Fourier transform (FFT) for a given signal analysis window. The reassignment of energy with respect to both time and frequency, such that the reassignments model the time–frequency fluctuations of the sampled signal, allows the RAFT to deliver higher resolution than the FFT (with the FFT, spectral/temporal analyses place all of the energy at points lying in the center of either a time/frequency sample-window, respectively). In particular, the improved spectral resolution of the RAFT for a given signal analysis window size is compared with that of the FFT for a fundamental speech processing application: pitch tracking. The results indicate that use of the RAFT instead of the FFT allows shorter signal analysis windows to be used. The pitch tracking performance of the RAFT with short analysis windows (e.g., 25 and 12 ms) is better than the FFT. [Work supported by the UK’s EPSRC.]
Effects of L1 phonotactic constraints on L2 speech perception and production
Authors: Imaizumi Satoshi , Tamekawa Yuji , Itoh Hidemi , Deguchi Toshisada , Mori Koichi
Published: July 2002

Effects of L1 phonotactic constraints on L2 speech perception and production were analyzed during the course of audio-visual perceptual training for Japanese adult learners of English by observing articulation of words containing /l/, /r/, and /w/. The speech identification score drastically improved during the training. The improvement in non-native word-initial rl distinction was clearly associated with the changes in the perceptual, articulatory, and neuronal spaces, which represent dissimilarities between the non-native and native phonemes in each domain assessed through perceptual, palatographic, and neuromagnetic measurements. Significant difficulty in identification of consonant-rl clusters, however, remained for some trainees even after one year of training. Analyses of palato-lingual contact patterns during word articulation suggested that more articulatory errors tend to occur in consonant-rl clusters than in other phonotactic contexts in such a way that both /l/ and /r/ are substituted with Japanese /r/ sometimes associated with a vowel inserted. Results suggested that the L2 phonological system can be acquired through a multimodal training, although it is strongly interfered with by L1 phonotactic constraints.
Neurophysiologic correlates of cross-language perception of phonetic categories
Authors: Sharma Anu , Dorman Michael
Published: July 2002

This study examined neurophysiologic correlates of categorical perception of native and non-native phonetic categories. Behavioral and electrophysiologic responses were obtained in Hindi and English listeners, in response to a stimulus continuum of naturally produced, bilabial CV stimuli that differed in VOT from −90 to 0 ms. These speech sounds constitute phonenically relevant categories in Hindi but not in English. As expected, the native Hindi listeners identified the stimuli as belonging to two distinct phonetic categories (/ba/ and /pa/) and their discrimination of a stimulus pair across categories was more accurate than their discrimination of a stimulus pair (of equal VOT change) within the /pa/ category. English listeners discrimination accuracy of within- and across-category pairs was at chance. N100 and MMN cortical-evoked potentials which are considered neural indices of preattentive sensory encoding and discrimination were measured. A double-peaked N100 was elicited in English and Hindi listeners. The first peak was elicited by syllable onset, while the latency of the second peak reflected the VOT lead time. Hindi listeners exhibited robust MMNs for the within- and across-category pairs, while English speakers did not, suggesting that the central sensory representation of certain phonetic categories is at least partially altered by linguistic experience. latency of the second peak reflected the VOT lead time. Hindi listeners exhibited robust MMNs for the within- and across-category pairs, while English speakers did not, suggesting that the central sensory representation of certain phonetic categories is at least partially altered by linguistic experience.