bluuebunny/crossref_metadata_2025 huggingface dataset:
Parameter estimation of layered structures with one accessible surface
Authors: Dixon G. , Horner J. L. , Petersson B. A.
Published: July 2002

It is frequently desirable, for inspection purposes, to test a layered structure nonintrusively. In practice, it is not always possible, or convenient, to access both surfaces. The aim of this paper is to determine the accuracy with which the parameters of a layered structure can be found, measurements being taken from only one surface. Using a matrix propagator model, a layer over a half-space is examined. Modeling point excitation, the sensitivity of the response to changes in the parameters of the layers are presented. The information gained will support an extended study, focusing upon systems with many layers, some of which may be composed of inhomogeneous materials.
Real-time control of a vibrating plate excited by a turbulent boundary layer
Authors: Friot Emmanuel , Georgelin Marc , Durand Christophe , Robert Gilles
Published: July 2002

Optimal feedback control was designed and successfully implemented to reduce the vibrations of a steel plate excited by a turbulent boundary layer. One side of the plate was equipped with piezoelectric sensors and actuators. The other side was imbedded in the wall of the CNRS-LMFA anechoic wind tunnel. Measurements of the wall pressure at flow speed ranging from 70 to 130 m/s showed that the plate was excited by a standard boundary layer; the parameters of a Corcos model for the wall pressure were identified from flow data. For control a state-space model for the plate was first identified from the actuator-to-sensor transfer functions; nine plate modes in the 0–1000-Hz range were taken into account. Covariance matrices for the boundary layer excitation were also identified from vibration data. An optimal linear quadratic Gaussian (LQG) controller was then designed from the plate and noise model. The LQG algorithm was implemented on the CNRS-LMA multichannel control system and tested at three flow velocities. A 10-dB reduction of the vibration level was achieved for the three modes mostly excited by the boundary layer.
An overview of Microsoft’s Whistler text-to-speech system
Authors: Huang X. D.
Published: July 2002

The data-driven approach can significantly facilitate the process of creating text-to-speech (TTS) systems for a new language, a new voice, or a new style. As such, Whistler TTS engine was designed to benefit from automatically constructed model parameters. Efforts to improve Whistler with the use of additional training data and better learning algorithms that make full use of these data will be reviewed. Training data have been augmented for a number of speakers. To better use these data, the hidden Markov model speech recognition system has been used to segment the training corpora and select more representative acoustic units. The classification and regression tree was used for both grapheme to phoneme conversation and unseen triphone generalization. Speech signal reconstruction was based on the mixed excitation source-filter model that leads to better compression of the acoustic inventory. A number of ways to smooth the spectral parameters were also studied to minimize the concatenation distortion. To improve automatically extracted prosodic templates, the learning process was refined with an analysis-by-synthesis approach. However, the coverage remains a challenge for the data-driven approach to make Whistler produce synthetic speech that resembles the original speaker. This is especially true for the prosody model.
Concatenative synthesis with subwords for restricted-prosody applications
Authors: Macchi Marian , Kahn Dan
Published: July 2002

For applications of text-to-speech in which the message prosody is restricted, very-high-quality speech can be achieved by concatenation of prerecorded words without physical modifications to the recorded words, if the words were spoken with the appropriate prosody. However, if the vocabulary is unlimited or changeable, it is necessary to have strategies for producing out-of-vocabulary words. Experiments were done with concatenating subword units from a prerecorded inventory to produce names (First name + last name pairs) spoken in isolation, a message set that is rich in phonemic and word-level prosody but extremely restricted in phrasal prosody. In the present approach, the use of rules predicting duration, pitch, and coarticulatory smooth was minimized. Those phonetic events that minimize distortion when functioning as splice points were identified, and then units were defined as the interval between these phonetic events. The units are labeled in terms of constituent phonemes, stress, word structure, intonation, and adjacent phonemes. Some of the labels were then collapsed into equivalence categories. Physical manipulations of the units were limited to time-domain smoothing of pitch at the concatenation boundary. Work is under way to optimize the set of splice points and maximize the size of the equivalence classes.
The role of speaking rate in the perception of L2 speech
Authors: Munro Murray J.
Published: July 2002

Previous work has shown that late second language (L2) learners typically speak more slowly than native speakers, and that speaking rate tends to correlate with overall L2 proficiency. However, the role of rate in native listeners’ judgments of the global accentedness of L2 speech has not yet been established. Because rate tends to covary with proficiency, it is not known, for instance, whether the poorer global accent judgments typically assigned to slow L2 speech are due to rate differences or to other properties (such as segmental accuracy). This paper is a synthesis of the results of a series of experiments addressing this relationship. The effect of rate differences on evaluations of global accentedness of sentence-length utterances was assessed by compressing and expanding L2 speech and presenting it to listeners for accentedness ratings. Rate differences accounted for a small but significant portion of the variance in listeners’ judgments. Overall, slightly accelerated L2 speech tended to be rated more highly than slower speech. These results are discussed in the context of rate measurements from a variety of types of L2 speech samples. The discrepancy between typical L2 production rates and native listeners’ rate preferences is examined. [Work supported by SSHRC.]
Modeling hydroacoustic waveform envelopes for comprehensive test-ban treaty monitoring
Authors: Dorfman Yevgeniy Y. , Pulli Jay J.
Published: July 2002

Robust and accurate localization of explosion events in the oceans requires modeling signal arrival times for a number of receivers. Additional information contained in the waveform envelope may also be used to recover other source properties and reduce the probability of false alarms in nuclear explosion monitoring. The feasibility of efficient envelope modeling was investigated by comparing adiabatic normal-mode model predictions with explosion data. The kraken computer code, com-bined with appropriate dispersion curve interpolation, was used to compute modal shapes and wave numbers in each range segment. To facilitate CTBT knowledge base development, the model has been used to illustrate the various factors that contribute to the shape of the received hydroacoustic waveform envelope, and provide design feedback for sensor placement and data analysis algorithms. The model was compared with data for two source–receiver paths: (a) the Chase21 ship scuttling explosion on 25 June 1970 off the New Jersey coast and recorded at the Ascension Island hydroacoustic array; and (b) nuclear explosions on Mururoa Atoll and received at the hydrophone at Point Sur, California. Adiabatic normal-mode approach was found inadequate for source and receiver located far from the SOFAR axis. [Work supported by U.S. DOE, Office of Nonproliferation and National Security.]
Simulated acoustic field in mudejar-gothic churches
Authors: Galindo Miguel , Zamarreno Teofilo , Giron Sara , Sendra Juan J. , Navarro Jaime
Published: July 2002

The most significant acoustic parameters in a set of mudejar-gothic churches were measured (all the churches were located in the city of Seville). From the analysis of these data important conclusions were otained about the behavior of the reverberant field versus source–receiver distance [J. Sendra et al., Computational Acoustics and its Environmental Applications—II (Computation on Mechanics Publications, Southampton, 1997), pp. 139–148] from Barron‘s revised theory [Barron et al., J. Acoust. Soc. Am. 84, 618–628 (1988)] and the relations between several acoustics quality descriptors: C80, D50, and RASTI [Zamarreno et al., Proceedings Acustica-98, Lisbon (1998)]. The three most representative churches were chosen to carry out a computer simulation with the raynoise software. Absorption coefficient values used in the model have been adjusted using measured reverberation time data, and thus the results were obtained for the other acoustic parameters in the different octave bands. It was proven that, in general, simulated values are coherent with the experimental results. Moreover, according to other authors [L. Marshall, J. Acoust. Soc. Am. 94, 2251–2261 (1994)], it has also been proven that C80 and D50 indices need no spectral analysis and their broadband values and their averages in significant bands (500–2000 Hz) are indifferent.
Development of a software package calculating zones of influence on marine mammals around industrial noise sources
Authors: Erbe Christine , Farmer David M.
Published: July 2002

A software package is presented which estimates zones of interference around underwater noise sources affecting marine mammals. An ocean sound propagation model based on ray theory computes the spreading of complex underwater sound such as broadband animal vocalizations and man-made noise. On a grid of receiver locations (representing the affected marine mammal), the received signal and noise sound spectra are compared. Given a species-specific audiogram, the software package plots zones of audibility around the noise source. Given species-specific vocalizations, zones of masking are plotted based on results obtained during an earlier study which measured masked hearing thresholds of a beluga vocalization in icebreaker noise with a trained beluga whale. Tools developed during this study (such as an artificial neural network and critical band methods) are incorporated in the software package and can be used to predict zones of masking for industrial noise other than the types directly measured with the beluga. Zones of discomfort, injury, and hearing loss could be plotted if thresholds were known or using current estimates. The software package is applicable to a variety of ocean environments requiring location-specific physical oceanography input data.
Resonance frequencies of absorption lines due to pelagic fish
Authors: Diachok Orest
Published: July 2002

Resonance frequencies, f0, of absorption lines due to pelagic fish with swim bladders change at twilight in accord with systematic changes in their depths and separations. Measurements of f0 at night, and calculations of f0, based on measurements of the effective radius, r0, and eccentricity, ε, of sardine swim bladders, and the average depth of dispersed sardines at night, are in good agreement. This result implies that swim bladder compression by internal organs did not have a large effect on r0 and ε during this experiment. Close agreement between laboratory measurements and calculated values of f0 of individual physotomes suggests that this inference is generally valid. Possible exceptions to this ‘‘rule’’ will be considered in the context of Ona’s (1990) laboratory observations of the effects of internal organs on swim bladder compression, and measurements of the state of internal organs of fish in the ocean. The resonance frequencies of absorption lines attributed to sardines in schools, f0′≊0.6f0. This result is consistent with a modified form of d’Agostino and Brennan’s (1988) equation for the fundamental mode of a bubble cloud. Their equation accounts for realistic numbers of fish per school (104–105). Modifications, which account for realistic separations between fish in schools and school eccentricity, will be presented. [Work supported by ONR.]
Modeling the target strength of Calanus finmarchicus
Authors: Francis David T. I. , Foote Kenneth G. , Knutsen Tor , Calise Lucio
Published: July 2002

The boundary element method is applied to the copepod Calanus finmarchicus, treated as a composite body, with fluidlike oil sac embedded in a fluidlike, mostly transparent prosome. The generally complex shapes of the two bodies are modeled on the basis of the actual dorsal- and lateral-aspect cross sections, as observed by videomicroscopy with the living, unanaesthetized animal encased in a droplet of sea water. Physical properties of the two bodies, namely mass density and longitudinal-wave sound speed, are derived through a combination of measurement and inference. Computations of backscattering cross section as a function of orientation and frequency are presented over the range 25 kHz to 3.2 MHz for a number of specimens. A sensitivity analysis is performed to quantify some uncertainty in the assumed values of the physical properties. [Support by the following is acknowledged: EU through RTD Contract No. MAS3-CT95-0031, Norwegian Research Council through Grant No. 113809/122, and Bergen Large-Scale Facility (LSF) for Marine Pelagic Food Chain Research.]
Comparison of Ayu target strength estimates determined from tank measurement and calculations
Authors: Sawada Kouichi , Yoshida Masahiko , Miyanohana Yoichi , Shirakihara Kunio
Published: July 2002

Ayu, Plecoglossus altivelis, is one of the most popular freshwater fish which can be seen in many rivers in Japan. Especially in the North Basin of Lake Biwa, Ayu is the highest predator. It is important to know the spatial and temporal variations of Ayu biomass to understand the predator’s effect to the food chain in the lake. Acoustic survey has been conducted since 1995. There is little information about the target strength (TS) of Ayu. Since Ayu has an open swimbladder, it is necessary to measure TS before the swimbladder was deflated by the suspension method in a tank. Target strength of live Ayu was measured in a tank and a three-dimensional image of swimbladder was obtained by an x-ray computer tomography (CT) system. In addition, a side and a dorsal image of the swimbladder were obtained by a soft x-ray system. All measurements were done in live condition. Kirchhoff approximation was used for the calculation on the basis of the three-dimensional image of swimbladder. A deformed cylinder model and a prolate spheroid model were also used for the calculation on the basis of side and dorsal image of the swimbladder. Calculation and measurement are compared and discussed in this paper.
Analyzing ship-borne acoustic Doppler current profiler measurements to estimate the vertical and spatial distribution of southern Black Sea zooplankton
Authors: Erkan Funda , Gucu Ali C.
Published: July 2002

The acoustic Doppler current profiler was used in Summer–Autumn 1996 and 1997 cruises of R/V bilim of Middle East Technical University to study the distribution of zooplankton in the southern Black sea. The backscattered signal intensity is compared with macro and mesozooplankton samples collected with Nansen closing net and 30-L Niskin water samplers. Analyzing the taxonomic groups and the length distribution of the individuals in the samples, size classification has been done. Total cross-sectional areas of the zooplankton in the length classes were compared with the resultant MVBS values. Highly significant correlations were observed. The relations between acoustic units and the size classes of zooplankton were presented by an empirical model which described a high percent of the total variance. The data sets from basin-wide surveys were used to obtain spatial distribution of zooplankton in the southern Black Sea. The spatial distribution of zooplankton was to a great extent determined by hydrography of the Black Sea. Daily migration patterns of zoo- plankton were studied, and the descending and ascending rates of different zooplankton groups were also calculated.
Improving echograms’ resolution by wideband pulse compression and dynamic focusing
Authors: Zakharia Manell E.
Published: July 2002

Improving the resolution of fisheries’ sounder echograms (i.e., reducing the confusion volume) can be achieved by improving either the axial or the angular resolution. On one hand, the range (i.e., time) resolution can be improved without any degradation of the signal-to-noise ratio by using frequency-modulated signals associated with pulse compression. Axial resolution can be considerably enhanced by using wideband signals. On the other hand, due to the angular spreading of the sonar beam, the lateral resolution is proportional to the range. Improving the angular resolution can only be achieved by using a larger array (only possible on the receiving side). Nevertheless, such a solution has a severe limitation as it reduces the volumes sampled at short range, leading to an angular under-sampling of the biomass. Dynamic focusing can be used to overcome this problem. Its principle is simple: the shorter the range, the smaller the array (i.e., the larger the angular aperture). The use of such a technique needs a compromise between the acceptable beam divergence and the system complexity. Both techniques will be applied to real data and echograms will be displayed and compared to the ones obtained with a standard system.
A simple optical microphone
Authors: Niehoff Wolfgang
Published: July 2002

In the past, a number of different working principles have been suggested for optical microphones, but many of these require a high degree of technical sophistication. Thus some suggestions are based on the interferometric principle, whilst others would utilize frequency modulation of laser light via a variable mirror (membrane) situated in the laser resonator. Light intensity modulation is a much more simple approach: A light beam emitted from an optical fiber is aimed at a reflecting membrane. The beam reflected from the membrane is captured by a second optical fiber and fed into a photodiode. Depending on the movement of the membrane, more or less light is launched into a second optical fiber. The photoelectric current of the photodiode thus expresses the movement of the membrane, and with it the sound field, as an electrical signal. As optical fibers transmit light with very few losses, the actual optoacoustic transducer can be positioned at a long distance from the light source and the photodiode. This allows problem-free use of the optoacoustic transducer in the presence of very strong electromagnetic fields (e.g., MRI). It is possible to improve upon this principle via simple modifications. The electroacoustic characteristics achieved are discussed and a model is presented.
New design concepts for the construction of an omnidirectional sound source
Authors: Goussios Christos A. , Papanikolaou George V. , Kalliris George M.
Published: July 2002

This paper presents a study for the optimum design and construction of an omnidirectional point source. The point source model of theoretical acoustics is used as an elementary unit for electroacoustic measurements in the area of room and free-field acoustics. The spherical radiation requirement for the entire acoustic spectrum and the finite dimensions of the point source model contradict the construction in practice. This is because of high-frequency beaming and the existing dimensions of radiators. A new design approach of using horn loaded cone loudspeakers and properly placed high-frequency diffusers was used to minimize these problems. Twelve radiators loaded on pentagon-shaped horns were placed in a dodecahedron topology to build a practical point source with a useful frequency response of 80–16000 Hz and an omnidirectional radiation pattern up to 12000 Hz. The results display improvement in the radiation pattern compared to those of the already existing point sources, due to the use of horns and diffusers. Increase in the sound pressure level is also displayed because of the obvious increase of the directivity index of each individual radiating element.
Acoustical training for stringed instrument makers
Authors: Richardson Bernard E.
Published: July 2002

The vast majority of popular stringed instruments found in western culture have been the product of generations of empirical experimentation on the part of craftsmen trying to provide a suitable vehicle for musical expression. In previous centuries, skills and knowledge were passed down from master to pupil during long apprenticeships, but whenever the continuity has been broken, there has been a feeling that various ‘‘secrets’’ have been lost. In more recent years, increased leisure time and better education has encouraged many individuals to take up instrument making, either as amateurs or as professionals. Most of these makers work in isolation, so there is now an even greater impetus to ‘‘rediscover’’ and communicate past arts. Some of these makers have turned to science to provide rational explanations of the many questions which are inevitably raised during the pursuit of making ‘‘fine’’ instruments. Based on the author’s experience of his interactions with pupils of violin and guitar making, this paper seeks to explore how scientific investigations can best be communicated to makers and examines to what extent ‘‘scientists’’ and ‘‘makers’’ can learn from each other.
Illustrating music performance principles by synthesis
Authors: Friberg Anders , Sundberg Johan
Published: July 2002

Music is generally not performed exactly as written. Instead, many small systematic variations of tone duration, pitch, articulation, etc. are found in a performance. These variations may serve the purpose of communicating different emotions or to outline the musical structure. An overview of commonly used principles is presented concerning the performance of different rhythmic patterns, the marking of important notes or chords, musical punctuation, and the hierarchical phrase structure. These principles are encoded in the form of rules implemented in a computer program, where they can be applied on any musical example. The possibility to isolate and change the quantity of the effects generated by each rule makes the system a unique tool for illustrating music performance to music students.
Music acoustics and contemporary musical composition
Authors: Bennett Gerald
Published: July 2002

Although it may seem that in the past musical composition got by well without specific knowledge of music acoustics, in fact the acoustical knowledge of each epoch has always determined the representational framework within which the composer formulates his ideas. For the last 50 years, however, composition in the field of electroacoustic music has required far more explicit and detailed acoustical knowledge than was the case for instrumental music. The relationship between music acoustics and composition today is one of reciprocal influence: frequently research in music acoustics is put to good (if sometimes surprising) use by composers. Sometimes, however, composers’ needs have led to important research in music acoustics. The present paper illustrates this reciprocity with examples from the areas of sound synthesis, physical modeling, and the simulation of reverberant spaces, and speaks of the importance of research for contemporary composition. In addition, the author presents a wish list for future work in music acoustics.
Acoustical contribution analysis of railway wheels and tracks
Authors: Katzenschwanz C. , Hölzl G. , Willenbrink L. , Lölgen T.
Published: July 2002

In a numerical analysis the contribution of different railway wheels and the track on the sound power and the sound pressure was determined. Two different wheels under different loads and positions were compared. In a first step a unit mechanical excitation in the connection point between the wheel and the rail was used to analyze the structural displacements of each configuration with finite elements. The structural energy on the surface of the wheel and the rail was used to select critical frequencies using boundary elements. In a second step the sound pressure was displayed on visualization planes and the contributions of the wheel and the rail to the sound power and the pressure at different points were discussed for the critical frequencies. It could be shown that the main sound pressure was dominated by the wheel. The bending modes of the wheel contribute a lot to the sound pressure. For optimized wheels the bending stiffness should be therefore increased. The position of the wheel—over the sleeper or in between—did not influence the contribution very much as well as the loading of the wheel.
Calculation of the three-dimensional sound radiation of rails using the multipole approach
Authors: Zeitler Berndt W. , Moeser Michael
Published: July 2002

Railway noise is of concern where an important factor is the sound radiated from the rail. This has been studied by using a multipole approach. The three-dimensional sound radiation of a vibrating rail can be calculated by approximating the normal velocity on the surface of the rail with the velocity produced by multipole sources within the rail. Their amplitudes are gained by minimizing the surface error by a method of least squares. The sound radiation is then the superposition of the pressure generated by all such multipole sources. It has been calculated considering one or two rails, coherent or not, each moving arbitrarily in the x and y directions (perpendicular to the length of the rail), with or without frequency-dependent ground absorption. The results of the approach are demonstrated.
A DRC-based analysis of the auditory hazard from exposure to impact noises generated by carpentry tools and tasks
Authors: Lilkendey Robert M.
Published: July 2002

To assess the risk of hearing from impulsive sounds several countries have developed damage risk criteria (DRC) which establish limits for an exposure to impulse noise based on key parameters of the impulse and a threshold for acceptable hearing loss. The German, French, Dutch, and an adjusted CHABA DRC were used to assess the potential for carpentry-based impact noises to present an auditory hazard to construction workers under normal working conditions. Measurements of impact noise events from compressed air nailguns and common hammering tasks were made in the field and simulated in a laboratory. The peak sound-pressure level, duration, and A-weighted energy of each impact were determined by digitizing the event and then processing the waveform using two impulse noise analysis programs. The maximum number of allowable exposures was calculated for each event based on the levels set by the DRCs and compared to published and observed productivity rates. The results showed that among the four DRCs, the German was the least protective, finding only 5 of the 12 exposures to be hazardous, and the French (7), Dutch (11), and adjusted CHABA (12) incrementally more protective.
Blast noise propagation above a snow cover at Haslemoen, Norway
Authors: Albert Donald G. , Hole Lars R.
Published: July 2002

During the short-range winter measurements at Haslemoen, Norway, a series of blast wave forms produced by 1-kg charges of C4 explosive fired 2 m above a dry snow cover 11 to 20 cm thick were recorded after propagating horizontally 100 to 1400 m. In attempting to model these experiments, it was found that the usual viscoelastic treatment of the ground was unable to match the recorded wave forms. A rigid-porous model of the snow cover was required to obtain good agreement. The measured wave forms, with a peak frequency around 50 Hz, show pulse broadening effects similar to those previously seen in pistol shot experiments using higher frequency waves over shorter propagation distances. The modeling predicts an average snow cover depth of 15 cm and effective flow resistivities of 30−41 kPasm−2, in agreement with snow pit observations and with previous acoustic measurements over snow. For propagation in the upwind direction, the pulse broadening caused by the snow cover interaction is lessened, but the overall amplitude decay is greater because of refraction of the blast waves.
CAA activities at ONERA using LES
Authors: Manoha Eric , Troff Bruno , Seror Christelle , Rahier Gilles , Sagaut Pierre
Published: July 2002

Most CAA applications rely on hybrid methods providing the radiated sound (using the Lighthill analogy, Kirchhoff integral or linearized Euler equations) from unsteady CFD. In that context, LES is widely considered as a compromise between full DNS and unsteady RANS. For two years, ONERA has undertaken studies aiming at understanding and applying LES in CAA. Former works focused on coupling LES to the Lighthill analogy. One typical application was the prediction of the noise generated by vortex shedding at the trailing edge of a thick flat plate. The incompressible LES was performed on a Cartesian mesh using the ONERA ‘‘pegase’’ code. The far-field noise was computed using the Curle formulation and favorably compared to airfoil noise experiments. More recently, a theoretical study analyzed the contribution of the LES subgrid scale model, or cutoff wave number, to the noise production computed from the Lighthill tensor, concluding that modeling this contribution may be important to correctly estimating the acoustic field. Other CAA applications of LES are now under progress: flap side-edge noise from high-lift devices of landing aircraft, and jet noise, with comparison of the Kirchhoff integral and Lighthill analogy. Coupling LES with linearized Euler equations is also a topic of interest.
On the coupling of the inverse problem with computational aeroacoustics
Authors: Li Xiadong
Published: July 2002

This presentation is to survey the recent developments and advances of inverse problems arising from various aeroacoustic applications, with the emphasis on the inverse problem of the Ffowcs Williams and Hawkings (FW–H) equation. Generally an inversion physical model should be first established to relate the sound source with its radiated acoustic field. The main difficultly is that most inverse problems are mathematically ill posed. The singular value decomposition (SVD) technique combined with the Tikhonov regularization method has shown to be a very powerful tool. Also, the generalized cross-validation (GCV) and L-curve criterions are rather suitable for choosing an optimal regularization parameter. Several studies have shown that the aeroacoustic source strength distributions can be reconstructed using the radiated acoustic field data, such as a propeller [X. D. Li and S. Zhou, AIAA J. 34, 1097–1102 (1996)] or a streamlined body [S. P. Grace, H. M. Atassi, and W. K. Blake, AIAA J. 34, 2233–2240 (1996)]. The inversion techniques can also be applicable for the spatial transformation of sound field and active noise control. Finally, the possibility of the coupling of the inverse problem with computational aeroacoustics is to be discussed. [Work supported by NSF and ASF of China.]
Reduction of boundary effects in acoustic analogy application to CFD databases
Authors: Wilde Andreas , Rose Marco
Published: July 2002

The calculation of noise generation and radiation using an acoustic analogy is very sensitive to the degree to which the sound-generating flow or rather the flow-simulation model meets the very strict assumptions made when deriving the aeroacoustic analogy. In many cases of technical interest, the popular assumption that there are no nonacoustic flow fluctuations on the boundary of the considered volume (control volume) is not satisfied. In these cases, the analogy has to be extended or an additional volume has to be attached, in which these nonacoustic fluid motions are damped. In this work, two examples of extension of the analogy formulation and of damping zones are studied. While all schemes yield reduction of noise prediction errors, the damping zone schemes are most effective in cancelling spurious noise sources. On the other hand, the extension of the acoustic analogies provides cost-effective ways to reduce noise prediction errors even at low frequencies.
How many ways is it worth slicing the social survey data cake?
Authors: Fidell Sanford
Published: July 2002

As when serving portions of cake, social survey findings on the prevalence of noise-induced annoyance may be sliced in many ways to create dosage–response relationships. Schultz (1978) divided ‘‘clustering’’ from ‘‘nonclustering’’ surveys; FICON (1992) sliced off a few inconvenient data points; several researchers and agencies have found reasons to develop separate relationships for certain noise sources; while yet others question the rationale for developing dosage–response relationships in the first place. Are the various dosage–response relationships usefully different from one another, and artfully enough sliced to support regulatory policymaking or theory development? Might differences among dosage-response relationships inferred by different means reflect statistical artifacts, or the influences of nonacoustic factors? This paper discusses applied and other uses for different slices of the same cake, and how many (more) slices might be useful.
Soundscape in cities, limits to the noise limits?
Authors: Kihlman Tor , Berillon Jean
Published: July 2002

Today traffic noise due to cars, trains, and airplanes is the main noise source in urban areas. In accordance to the European Commission’s Green paper on Future Noise Policy, more than about 250 million people are exposed to A-weighted outdoor levels higher than 55 dB. To improve the situation is a tremendous challenge and the question arises if we are really able to provide a good environment (i.e., levels below 55 dB) for all people in today’s cities. In a previous paper [Kihlman and Kropp, Proceedings International Congress on Acoustics, Seattle (1998)], different cities with different structures and automobile dependencies are compared in a general study based on statistical data. A flat regular town structure was assumed without shielding due to, for instance, buildings. In this paper the studies are extended to more common situations including detailed information about traffic flow and the screening of traffic noise due to the building structure. Results are compared with sound maps for different cities.
Advances in pulse tube refrigeration
Authors: Radebaugh Ray
Published: July 2002

The pulse tube refrigerator is now being developed for a wide variety of cryogenic cooling applications, and it is beginning to replace other types of cryocoolers in many of these applications. This widespread interest occurs because there are no moving parts in the cold end. Most of this development in pulse tube refrigerators began in the mid-1980s with the introduction of the orifice to provide the proper phase relationship between the gas velocity and the pressure. The pulse tube refrigerator has now become the most efficient cryocooler for temperatures of 60 to 120 K. The current state-of-the-art in pulse tube refrigerators is discussed in this paper. Temperatures as low as 2.2 K with two stages and efficiencies as high as 19% of Camot at 80 K in one stage have been achieved. The operating fundamentals are explained very well through the use of thermoacoustic modeling techniques. Recent R&D areas to be discussed are the use of additional orifices, inertance tubes, tapered pulse tubes, dc flow, and thermoacoustic drivers. The fluid dynamics within the pulse tube is still not well understood, and further research in this area is needed to improve performance even further.
A general model of inner ear pressure responses including ultra-low frequencies
Authors: Haberland Ernst-J. , Neumann Hans J.
Published: July 2002

Various volume displacement processes act on the inner ear fluids, e.g., via the stapes, via the capillary bed, via the cochlear aqueduct or in the case of ultra-slow processes via pressure homeostasis from tissue. Frequencies below the audible range make a pressure difference in the perilymphatic scales but without any audible traveling wave dispersion. In experiments in guinea pigs, interference patterns were found between audible and low-frequency nonaudible perilymphatic flows in CM recordings. The function of perilymph can be subdivided into the dispersional part and the conventional part. Franke and Dancer [Arch. Otorhinolaryngol. 234, 213–218 (1982)] described a model of the guinea pig’s cochlea in accordance with pressure measurements in the two perilymphatic scales up to low frequencies at the lower boundary of the audible range. In our own experiments, it was measured pressure responses to slow and ultra-slow changes of perilymphatic infusions and aspirations and superimposed incremental volume pulses into the perilymphatic system [Neumann and Haberland, Biophysikalische Untersuchungen (Tectum Verlag, Marburg, 1998)]. From these experimental findings was developed an extended model of the passive pressure characteristics of the inner ear system that takes into account nonlinearities and valve characteristics in the convection-relevant slow and ultra-slow movements of perilymph.
Radiation of sound from a monopole source near a free-flooded cylindrical shell
Authors: Kim K. Steven
Published: July 2002

Radiation of sound from an acoustic source changes if there is a structure nearby. Radiated and scattered acoustic pressure fields are obtained when a free-flooded cylindrical shell of finite length is subjected to acoustic waves from a monopole source. By solving integral equations formulated for an elastic thin shell with simply supported boundary conditions at both ends, far-field acoustic pressures are calculated. Cases of various source locations inside and outside a shell are discussed.
Estimating radiation modes for broadband measurement
Authors: Sommerfeldt Scott D. , Lin Dong
Published: July 2002

Acoustic radiation modes have been investigated as a means of estimating far-field radiated acoustic power, based on structural vibration measurements. One of the difficulties associated with radiation modes is the fact that the mode shapes are frequency dependent. Much of the research in the past associated with radiation modes has focused on the low-frequency range, where the radiation modes are nearly frequency independent. The work reported here is aimed at developing a measurement approach based on radiation modes, which is applicable in the low- and mid-frequency range, where the frequency dependence of the radiation modes cannot be ignored. In general, if a limited number of point vibration sensors are to be used to estimate the radiated acoustic power, the appropriate locations on the structure for those sensors will change for each frequency. However, it will be shown that if one uses distributed sensors, it is possible to use a single sensor configuration to estimate the acoustic radiation over a broad range of frequencies. Numerical results will be shown that indicate the measurement accuracy that can be expected over a range of frequencies up to a normalized frequency of kL=15.
An active silencer for harsh environmental conditions
Authors: Krüger Jan
Published: July 2002

Active noise control has proven effective in the reduction of low-frequency noise in ducts, e.g., in air conditioning systems. However, in many applications the capability of proper operation of an active silencer is severely limited by the harsh environment. In fact, higher temperatures, static and dynamic pressure loads, extreme sound power levels, and abrasive flow often prohibit the use of active silencers at all. Appropriate design details for an actively absorbing silencer are presented, leading to a new construction with high performance and durability. Using standard components (microphone, loudspeaker), robust protection measures are developed which can be adapted to particular applications. A computational model derived from the theory of active silencers allows the variation of design parameters and the prediction of the insertion loss. To verify the numerical results, experiments under several acoustical and thermal conditions were conducted.
Active noise control of higher-order modes in a rectangular duct using stabilized fast transversal filters
Authors: Opfer Holger , Guicking Dieter
Published: July 2002

For active noise control of sound propagating down a duct in the fundamental mode commercial systems are available. Cancellation of sound also propagating in higher-order modes adds the requirement of higher computational power for the signal processing unit, and stability problems due to source coupling effects. The widely used LMS algorithms need little computational effort, are very stable, but also have a slow tracking behavior to changes in the input signal. Depending on the eigenvalue spread of the input signal’s autocorrelation matrix they lead to slow convergence. One method to overcome this problem is to use stabilized fast RLS algorithms which are independent of the statistics of the input signal. A multi-channel stabilized fast transversal filter algorithm has been investigated experimentally in a rectangular duct with respect to its stability, convergence, and performance.
A hybrid active–passive system for controlling locomotive exhaust noise
Authors: Remington Paul J. , Knight Scott , Hanna Douglas
Published: July 2002

Diesel electric locomotives represent the primary source of motive power in the U.S. railroad industry, and are a significant source of environmental noise. There is a variety of noise sources on a locomotive; however, two sources will have to be controlled before significant reductions in locomotive noise can be achieved: diesel engine exhaust noise and cooling fan noise. Here a system is described that uses both active and passive components to control broadband exhaust noise below 4 kHz. The active system utilizes eight actuators and eight residual microphones in an adaptive feed-forward configuration to control tonal noise below 200 Hz. The passive system is a compact exhaust silencer designed to control exhaust noise above that frequency while still fitting within the limited space available beneath the locomotive hood. Both active and passive systems will be described in detail, and practical problems encountered in the design of the combined systems will be described. Performance estimates for both the active and passive system will be presented along with the available measurements. [This work was supported under contract to the U.S. Federal Railroad Administration.]
Parameter estimation from speech signals for tube models
Authors: Schnell Karl , Lacroix Arild
Published: July 2002

Discrete time tube models implemented by lattice structures can be used to simulate the propagation of sound waves in the vocal tract. To cover a wide variety of speech sounds, it is necessary to extend the standard all-pole tube model by coupling the nasal tract, and, furthermore, enable the excitation of the tube system at different places in between. The resulting filter structure H(z) now shows poles as well as zeros in the transfer function. The impedance of the lip opening is realized using a model from Laine [U. K. Laine, ‘‘Modeling of lip radiation impedance in the z-domain,’’ Proc. ICASSP-82, 1992–1995 (1982)], which includes one pole and one zero in the z-plane. The parameters of the tube model will then be estimated from speech by minimization of an error, which describes the spectral distance between H(z) and the prefiltered speech signal. The prefilter is used for the separation of signal components due to excitation and radiation. A gradient-based optimization is implemented to find the minimum. Following this procedure, the frequency response is able to approximate the spectral envelope of the speech signal. The resulting areas correspond to those obtained by x-ray or NMR investigations.
The influence of vowel context on the Japanese listener’s identification of English voiceless fricatives
Authors: Martens William L. , Lambacher Stephen
Published: July 2002

In order to examine the influence of vowel context on the ability of native Japanese speakers to distinguish between English voiceless fricatives, recognition rates from a five-alternative, forced-choice (5AFC) test were analyzed in terms of receiver operating characteristics. This signal-detection-theoretic analysis showed contrasting effects of vowel context on sensitivity versus response bias. Subjects heard each of 75 nonsense syllables spoken by three native speakers of English, and were asked to report whether the syllable they heard contained /f/, /s/, /∫/, /θ/, or /h/. Fricative identifiability (measured by the index of sensitivity, d′) appeared to be modulated primarily by phonemic and phonetic factors. In contrast, the likelihood of giving one fricative response over another without regard for which fricative is presented (measured by the index of bias, β) appeared to be modulated by more subjective factors such as familiarity with common loan words (words of foreign origin). A cross-linguistic examination of phonemic and phonetic factors revealed possible reasons why the Japanese listener’s identification of English fricatives is considerably poorer in particular vowel contexts, and how differences in fricative production can make it more difficult to make certain phonetic distinctions.
Can English listeners learning Japanese as a second language employ a mora-based segmentation procedure?
Authors: Otake Takashi , Yoneyama Kiyoko
Published: July 2002

Previous research has shown that monolingual listeners exploit language-specific segmentation routines, based on linguistic rhythm, not only with native but also with foreign input. Thus monolingual Japanese listeners used a mora-based segmentation procedure with both Japanese and English, while monolingual English listeners, whose linguistic rhythm is different from Japanese, did not use this procedure [Cutler and Otake, JML 33, 824–844 (1994)]. Bilinguals, however, can inhibit an inappropriate rhythm-based segmentation procedure [Cutler et al., Cogn. Psychol. 24, 381–410 (1992)]. The present study explored whether English listeners learning Japanese as a second language could employ a mora-based segmentation procedure in Japanese. Three groups of American subjects, whose Japanese proficiency differed, participated in the experiment, using the phoneme monitoring materials of Cutler and Otake (1994). The prediction based on the findings above was that none of these subjects would be sensitive to morae in listening to Japanese. The results, however, showed that all the subject groups were consistently sensitive to morae, which suggests that English listeners can employ a non-native segmentation procedure in a foreign input as long as they have at least begun to acquire the foreign phonology. [Work supported by Fulbright senior research grant and the Telecommunications Advancement Foundation.]
A study of the language dominance effects on bilinguals’ voice quality
Authors: Bruyninckx Marielle , Harmegnies Bernard
Published: July 2002

This paper examines whether language dominance exerts influences on speaker’s voice quality. Ten French–Polish bilingual subjects, split into a French-dominant group and a Polish-dominant group, were selected. They were asked to record five utterances of two balanced texts (one for each language). One FFT-based LTAS was computed for each utterance. The so-obtained 100 spectra were compared by means of a pattern recognition procedure: the SDDD dissimilarity index. Two types of within-speaker comparisons were performed for each subject: within-language comparisons and between-language comparisons. The results show that the average SDDD index values derived from the between-language comparisons score higher than those drawn from within-language matchings. This confirms our earlier results based on different types of bilinguals. The within-language FR/FR SDDD index values tend to score higher in the French-dominant group than the POL/POL ones; the opposite relationship can be observed in Polish-dominant subjects. These results empower one to generalize the idea that language dominance seems to be a significant variable in the characterization of the phonetic behavior of speakers using more than one language.