bluuebunny/crossref_metadata_2025 huggingface dataset:
Bubble dynamics and nucleation of ice in supercooled water
Authors: Trinh E. H. , Ohsaka K.
Published: July 2002

The effects of mechanically induced impulses such as vibration and flow have been used to induce the solid–liquid phase transformation in liquids held below their equilibrium freezing point. Although processes used for grain refining in metallurgy rely on this artificially enhanced nucleation process, there is no clearly identified underlying physical mechanism. Experiments have been conducted with single bubbles immersed in a small volume of supercooled water and trapped in a 22-kHz resonant cell to study the action of both large amplitude radial and shape oscillations. The experimental observations reveal that large-amplitude radial oscillations of micron-size air bubbles induce the nucleation of ice when a minimum supercooling of 5 °C is reached. Shape oscillations and radially nonresonant bubbles do not induce nucleation of ice within the same ‘‘incubation’’ time. The appearance of the ice crystal at the location of the trapped bubble strongly suggests a cause and effect correlation between the bubble radial oscillations and the freezing onset, and both sonoluminescing and nonluminescing bubbles appear to induce nucleation. This mechanism also has the additional benefit of allowing the direct measurement of free dendritic growth velocity within a supercooled liquid bulk in the absence of any artificially introduced seed. [Work sponsored by NASA.]
Acoustic diagnosis of singing cavitation vortices
Authors: Bajic Branko
Published: July 2002

The acoustic field generated by hydrodynamic cavitation is usually random and broadband. There is, however, a case in which it is tonal. Indeed, as reported by Higuchi, Arndt, and Rogers [J. Fluids Eng. (Dec. 1989)], and Maines and Arndt [ASME Symp. Cavitation and Multiphase Flow (1995)], cavitating vortices which shed off from a body immersed in a flow may generate a harmonic acoustic signal even if involving no vibrations of the cavitating body. This paper presents the results of the parametric analysis of sound generated by the singing vortices produced in a test on a finite-span hydrodynamic profile immersed in a uniform water flow. A continuous development of the vortex through filling by air was investigated. The acoustic diagnosis of the process of oscillations of the cavitating vortex generating the singing is presented. The issues relevant to the theoretical modeling of the lock-in mechanism are investigated: necessary conditions for the onset and stability of the process, possible influence of water velocity other than through the cavitation number, influence of vortex energy, etc. The relevant peculiarities of the vortex dynamics are revealed. The question if there is a back-influence of the singing on the basic vortex dynamics is considered.
The effect of mean flow/source motion on sound-pressure fields with absorption
Authors: Kelly Jeffrey J.
Published: July 2002

An analysis of sound-pressure fields relative to both uniformly translating and space-fixed reference frames is presented. By using Fourier transforms and separation of variables, a solution is determined in the source frame. Far-field approximations are transformed to emission coordinates which model the pressure field observed in a space-fixed reference frame. The influence of mean flow/forward flight on source-fixed directivity patterns is shown. A finite cylindrical source was considered and directivity patterns were constructed which show the effect of Mach number, frequency, and displacement pattern on the sound field. The influence of flow on absorption due to viscosity is also presented.
Model-based dispersive wave processing: A recursive Bayesian solution
Authors: Candy J. V. , Chambers D. H.
Published: July 2002

Wave propagation through dispersive media represents a significant problem in many acoustic applications, especially in ocean acoustics, seismology, and nondestructive evaluation. In this paper we propose a propagation model that can easily represent many classes of dispersive waves and proceed to develop the model-based solution to the wave processing problem. It is shown that the underlying wave system is nonlinear and time-variable requiring a recursive processor. Thus the general solution to the model-based dispersive wave enhancement problem is developed using a Bayesian maximum a posteriori (MAP) approach and shown to lead to the recursive, nonlinear extended Kalman filter (EKF) processor. The problem of internal wave estimation is cast within this framework. The specific processor is developed and applied to data synthesized by a sophisticated simulator demonstrating the feasibility of this approach.
Modeling speech intelligibility and quality on the basis of the ‘‘effective’’ signal processing in the auditory system
Authors: Kollmeier Birger , Wesselkamp Matthias , Hansen Martin , Dau Torsten
Published: July 2002

A model of the ‘‘effective’’ signal processing performed by the auditory system is applied to the quantitative assessment of speech intelligibility (cf., Kollmeier et al., Proc. ICA 95, III, 81–84). It consists of a gamma one filterbank followed by envelope extraction, nonlinear adaptation circuits, and subsequent modulation low-pass filtering in each filter channel before being fed into a cross-correlation-type detector or pattern recognizer. The model is capable of predicting the speech intelligibility of both normal and hearing-impaired listeners under a wide range of signal-to-noise ratios, as well as the perceived quality of speech signals whose spectro-temporal properties are systematically degraded. The degradations are generated with a modulated noise reference unit either in a band-specific, time-independent way or in a broadband, temporally varying way. These intelligibility and quality assessment data can be used to deduce the components of the model most germane to the processing of speech, as well as provide information about how to enhance and extend the model for better predictive performance. a)Now at Siemens Audiologische Technik, Erlangen, Germany. b)Now at Widex Hearing Aids, Vaerloese, Denmark.
Estimation of time-varying signal parameters via VFM Kalman filtering
Authors: Lee Sang-Wook , Lim Jun-Seok , Jun Byung-Doo , Sung Koeng-Mo
Published: July 2002

This paper proposes a VFM (variable fading memory) Kalman filtering and applies it to the parameter estimation for time-varying signal. By adaptively calculating fading memory, the proposed algorithm does not require any fixed predetermined fading memory when estimating a time-varying signal. The time-varying signal parameter estimation performance is evaluated by computer simulations for two cases, one of which is the chirp signal whose frequency varies linearly with time and the other is the chirp signal with an impulsive outlier.
Middle ear adaptations in an old world amphibian
Authors: Narins Peter M. , Purgue Alejandro P. , Lewis Edwin R.
Published: July 2002

Each tympanic membrane (TM) of males of the Old World frog Petropedetes parkeri is decorated with a single, prominent, fleshy papilla during the breeding season. The papilla is composed of a nonossified, spongiform tissue that contain fluid-filled canals that terminate at or near the papillar surface. Roughly three-quarters of the papillar mass is located on the outer surface of the TM, while one-quarter is found on the inner TM surface. Injecting sound into the vocal cavity and measuring the vibration response of a point of the tympanic membrane using laser Doppler vibrometry revealed that the papilla vibrates little in response to sound, but clear modes are observed on the remaining portion of the TM. Moreover, the Fourier transform of the impulse response of the male’s intact TM exhibits a peak at ∼750 Hz, close to the dominant frequency of the male’s advertisement call (780 Hz). This tuning is interpreted as an adaptation for broadcasting of the call from the eardrums of the male. Surgically removing the papilla resulted in a surprising change in the TM response properties. [Work supported by NIDCD Grant Nos. DC-00222 to PMN and DC-00112 to ERL.]
An excitation of surface acoustic waves in the ear canal
Authors: Evreinov E. Grigori
Published: July 2002

Usually electrodynamic loudspeakers have a small flexibility of the diaphragm. Therefore, at sound reproduction of low frequencies, a coupling space between the diaphragm and the ear must be hermetic. In a range of high frequencies, distortions are stipulated by the presence of reflected sound waves, and depend on the ear canal’s position in the closed sound space. There is a coupled acoustic resonant system which consists of the pinna, the eardrum, and the ear canal. Sound oscillations excited in a cartilaginous part of the outer ear are partly reflected, but an absorbed component is spread through subcutaneous conjunctive tissue toward an attachment place of the tympanic membrane via surface acoustic waves (SAW). The hearing thresholds’ data received by use of a special sound conductor which is generating SAW, in a range of frequencies 125–8000 Hz, did not show statistic distinctions regarding measurings carried out via conventional tonal audiometry. It is supposed that mechanical energy necessary for excitation of auditory sensation with the help of oscillations spreading through the conjunctive tissue of the ear canal is required even less than at the creation of equivalent levels of air pressure because an amplitude of acoustic oscillations is directly attached to the eardrum edge.
Physical constraints on blue whale sound production mechanisms derived from matched-field processing
Authors: Thode Aaron M. , D’Spain Gerald L. , Kuperman William A.
Published: July 2002

Matched-field processing (MFP) is a technique for estimating the range and depth of an acoustic source using data collected from a multiple hydrophone array. Physical constraints on the sound production mechanism used by blue whales can be established using this method, including a vocalizing animal’s depth, depth and range changes during vocalization, source level estimates, and removal of propagation effects from the animal’s source signature. Data are presented from a 1996 experiment off the California Channel Islands that contain a 30-min. sequence wherein a single individual produces 10 pulses and 22 FM sweeps from distances up to 5 km away. The animal appears to vocalize at depths between 15 and 25 m during both types of vocalizations, and swims with horizontal speeds between 2 to 5 m/s. The source levels generated by a type A call vary from 176–192 dB re:1 μPa @ 1 m over a 10- to 100-Hz band, consistent with previous estimates [W. C. Cummings, J. Acoust. Soc. Am. 50, Pt. 2, p. 1193]. The type B source levels will also be analyzed and discussed. [Work supported by ONR.]
Anthropogenic ocean noise: Negligible or negligent impact?
Authors: Ketten Darlene , Potter John R.
Published: July 2002

The increasing use of the oceans is accompanied by rising acoustic pollution. From explosive transients to the nearly continuous drone of ships, anthropogenic noise accompanies virtually every human activity in the sea. Anthropogenic noise has the potential for significant impact on marine species, especially marine mammals, yet existing data are insufficient to predict accurately any but the grossest consequences. There are almost no controlled data on how the acoustic environment is changing as a result of human activity. There is also little information on how marine mammals respond physically and behaviorally, either to intense transients or to a long-term background increase in ambient noise. Ignorance and resulting caution in testing potentially harmful acoustic devices may be seriously hampering our learning progress and the development of effective acoustic deterrents that could decrease marine mammal by-catch. This work discusses the current understanding of acoustic trauma and ambient noise characteristics in the context of what is known about marine mammal hearing and its variations. The work focuses on how species vary in their potential for impact and on determining whether acoustically vulnerable species coincide with acoustic ‘‘hot spots’’ in the ocean where Man’s activities may damage hearing and disrupt key behaviors.
The investigation of physiologically controlled resonance scattering data from fish using a new acoustic approach
Authors: Feuillade C.
Published: July 2002

Low-frequency acoustic scattering from fish is typically dominated by the swimbladder resonance response. Swimbladders are physically similar to bubbles, and it has been traditionally assumed that, as with bubbles, resonance frequency measurements may be directly used to calculate the swimbladdeer volume. Typically, swimbladder models used to interpret fisheries scattering data, and thus determine fish size and abundance, have adopted this premise. Questions are raised, however, by one set of measurements on depth-adapted Atlantic cod by Sand and Hawkins [J. Exp. Biol. 58, 797–820 (1973)], which indicates resonance frequencies much higher, and with more rapid depth variations, than can be realistically explained using such simple assumptions. They argued that the anomalous resonances they observed were actively controlled by the fish, and not just a passive property of the swimbladders. To investigate this phenomenon, a new model was recently developed which incorporates both viscous and elastic properties of fish flesh as variables influencing swimbladder scattering [J. Acoust. Soc. Am. 103, 3245–3255 (1998)]. This presentation will demonstrate how this model provides a basis for understanding the physiological behavior of the fish, and a starting point for determining which species of fish may be reliably surveyed using the traditional approach.
Acoustic observations of the annual cycle of fish and plankton populations in the Beaufort Sea
Authors: Pinkel Robert
Published: July 2002

During the period November 1997 through September 1998, a 160-kHz Doppler sonar was operated at the SHEBA ice camp. During this period, the camp drifted clockwise around the Beaufort Sea. The sonar, constructed at the Marine Physical Laboratory of the Scripps Institution of Oceanography, transmitted coded pulses with an 8-kHz bandwidth and recorded echo intensity, spectral bandwidth, and mean Doppler shift. The fall 1997 data featured very weak return echoes. Distinct hard targets were observed over ∼5% of the profile. These have been tentatively identified as Arctic Cod. With changing season and camp location, both plankton and fish populations have evolved significantly. Algorithms have been implemented to track the discrete hard targets and the underlying continuum planktonic population through the course of the year.
Acoustic abundance estimation of midwater animals
Authors: Foote Kenneth G. , Roettingen Ingolf
Published: July 2002

The echo integration method of estimating the abundance of midwater animals, specifically pelagic fish and zooplankton, is reviewed. The roles of the modern scientific echo sounder and complementary postprocessing system are emphasized, but requirements imposed by availability of the target animals to surveying by vertical transducer beam, environment including weather and possible presence of other significant scatterers, and the animal itself, are mentioned. The process of converting acoustic measurements of animal density to abundance over the survey region is illustrated in detail for the case of Norwegian spring-spawning herring (Clupea harengus) when wintering in a fjord system. A simple explanation is given for continued interest in the scattering properties of midwater animals, specifically their backscattering and extinction cross sections. The constant need for positive scatterer identification, as through physical capture or optical registration, or, failing these, acoustic classification, is described. The usefulness of geostatistics in estimating abundance and the variance of the abundance estimate, as well as in quantifying the observed spatial distributional properties of the animal, is mentioned. [Partial support of the EU through RTD Contract Nos. AIR2-CT94-1007 and MAS3-CT95-0031 is acknowledged.]
Slot silicon microphone: Analytical and experimental study
Authors: Skvor Zdenek , Stoffel Axel
Published: July 2002

The diaphragm, as an acoustically active part of the electrostatic silicon transducer, is usually located in the front of the transducer, and the size of the substrate plate determines the entire size of the assembly. It is possible to place on the substrate plate a higher number of sensors and thus obtain a higher sensitivity of the whole system. The increasing number of diaphragms provokes the enlargement of the frontal area of the assembly and the device loses its miniature size. In this contribution a new arrangement is presented of electrostatic or similar pressure sensors with an indirect pressure excitation of their diaphragms and with a small frontal area of the assembly [A. Stoffel and Z. Skvor, Slot microphone, Applic. for U.S. Patent 9,010,032]. Together with an adjacent plate, the substrate plate with diaphragms forms a thin rectangular air-gap with constant or variable cross section and with frontal or frontal and lateral opening. The analytical description is based on the solution of the two-dimensional Helmholtz equation and the transfer function was derived with respect to visco-thermal losses and was experimentally verified.
Model measurement of the microphone array
Authors: Cerny Libor
Published: July 2002

The use of silicon has allowed the fabrication of microphones with integrated electronic circuitry and the development of the FET microphone. The introduction of lithographic technique has resulted in microphones with very small (1 mm2) diaphragms with specially shaped backplates. The application of corrugated diaphragms ensures promising future development of silicon microphones. On the basis of this work the arrangement can be prepared for an encapsulation of a miniature microphone chip. Frequency response of the microphone arrays of various types was measured on models in an enlarged scale. It was proven by the numerical results and by the measurements on the model that this manner of encapsulation is possible and advantageous. It was concluded that there is a satisfactory agreement between the analytical description and experimental results of model measurements. The measurement can be transposed to frequency ranges in the ratio of wavelength and dimensions and the damping of modes can be calculated. This problem solution is cheaper than measurement of the miniature microphone chip [Z. Kvor, ‘‘Study of the Air-gap and the Influence of the Microphone Position on the Substrate Plate,’’ Report Copernicus CP 940515 (1997), P. R. Scheeper, Sensors and Actuators A44, 1–11 (1994)].
The Plymouth Prowler, active noise control exhaust system. Part I: Noise reduction
Authors: Shipps J. Clay
Published: July 2002

An active noise control exhaust system or electronic exhaust system was developed for the Plymouth Prowler to demonstrate the ability to reduce backpressure greatly while also significantly reducing the exhaust noise. An electronic exhaust system is typically composed of passive and active elements. One of the major advantages of electronic exhaust systems is that most of the flow restrictions in the passive elements can be removed; this is reponsible for the backpressure reduction. Reduced backpressure typically results in a 3% to 6% horsepower gain at the rated engine speed of the vehicle and increases torque over most of the engine speed range. The electronic exhaust system also has the capability of greatly reducing the exhaust noise. The active elements can produce antinoise at very high levels such that, when combined with the unwanted exhaust noise, an overall decrease in the radiated tailpipe noise occurs. This allows the vehicle to meet the legal passby requirements with significant margin while maintaining appropraite sound quality characteristics. This paper will describe the system developed for the Prowler and the resulting performance gains.
The Plymouth Prowler, active noise control exhaust system. Part II: Noise shaping
Authors: Meeks Timothy E. , Shipps J. Clay
Published: July 2002

An active noise control exhaust system or electronic exhaust system was developed for the Plymouth Prowler to demonstrate the ability to reduce exhaust backpressure greatly while creating a legal sports car sound for the vehicle. One of the major advantages of electronic exhaust systems is that flow restrictions in the passive elements can be removed, consequently decreasing the total backpressure. This backpressure reduction typically results in a 3% to 6% horsepower gain at the rated engine speed of the vehicle and also increases torque at most engine speeds. The electronic exhaust system also has the capability of shaping the exhaust noise. The active elements can produce antinoise at very high levels such that, when combined with the unwanted exhaust noise, there is a significant reduction of selected harmonics. The software which processes the antinoise can be programmed to use engine speed information to determine which harmonics will comprise the resulting waveform. This control varies the vehicle’s sound quality characteristics while meeting passby noise requirements. This paper will describe the system developed for the Prowler and the resulting performance gains and sound quality enhancements.
Sound reflex compensation in active noise control
Authors: Ogonowski Zbigniew
Published: July 2002

Creation of a zone of quiet in an enclosure by using active noise control (ANC) methods causes problems with additional noise generated by a reflection of secondary sound. The paper presents a new method of the reflected sound compensation. The idea is to augment the main ANC controller with an auto-feedback controller (AFC). No additional measurement is needed because AFC is fed back with the main controller output. Increase of delay time of the successive reflection assures nonanticipation of the compensation. AFC corrects an activation of the secondary source of sound to cancel the reflected secondary sound generated by the main controller. AFC synthesis uses a mathematical model of the reflection. A specially designed method of the model identification using multisinusoidal excitation is described in the paper. Usually, geometry of the enclosure is constant, thus stationarity of the model is assured. The choice of the model structure implying stability issues is discussed in the paper. Stability of the AFC loop depends also on the reflection delay time. A special stabilizing design method is presented in the paper. An illustrative example of the method application is described.
Nonlinear acoustic testing of defects for the evaluation of risk for insurance of thermal electric power stations
Authors: Robsman Vadim A. , Novak Yury V.
Published: July 2002

The testing of defects in buidings, smoke tubes, groundworks of turbine generators, and other principal constructions of an electric power station without their removal from service was performed. Both externally exited acoustic pulses with narrow-band spectrum and acoustic emission pulses caused by vibration of power equipment were used as sources of structural information. Harmonics generation was observed at a small concentration of defects. Increase in concentration causes the growth of higher harmonic intensity and the broadening of their spectral lines. In highly damaged constructions, the spectral lines merge together into a continuous wideband spectrum. Acoustic emission can complicate the spectral distribution by introducing the random frequency components. It was discovered that the analysis of different stages of nonlinear evolution of spectra makes it possible to distinguish the different kinds of internal defects and, consequently, to predict the risk of emergency. This approach was put into use recently for insurance of thermal energy plants in Russia.l
Experiments in solids using air-coupled ultrasound
Authors: Hutchins D. A. , Billson D. R. , McIntyre C. S.
Published: July 2002

Electrostatic (capacitance) transducers have been designed and constructed for the imaging of solids in air, at frequencies of up to 1 MHz. The devices use a micromachined silicon backplate and a flexible membrane. The sensitivity and bandwidth of the resultant devices is such that signals can be propagated through solid samples in air. Examples will be given of imaging experiments in a wide range of materials, including fiber-reinforced polymer composites, paper, and metals. Measurements of material property can also be made, and examples will be given of how such measurements can be made at various temperatures. Particular reference will be made to measurements where it is possible to study changes in structure, by measurement of the elastic properties of the material.
Studies of particle separation using acoustophoresis
Authors: Brooks Todd L. , Apfel Robert E.
Published: July 2002

Particles in a liquid medium can be manipulated using the acoustic radiation force generated by a high-intensity ultrasonic standing wave. The resulting motion of the particles depends on the relative contrasts in particle density and compressibility with the surrounding host liquid. This allows the possibility of a separation technique that can distinguish particles based on these elastic properties, which is not possible with other separation methods. A flow-through resonator has been designed in which an acoustic standing wave is oriented perpendicular to the direction of fluid flow. This allows incoming particles to be moved perpendicular to the flow so that they can be separated into separate streams at the output end. Results will be shown for several different types of particles in a range of sizes (5−100 μm). There is also a secondary force which can cause an attraction between particles. This force is usually considered negligible except when particles are very close together. Some preliminary calculations will be shown predicting the effects of the secondary particle force and these will be compared to observations. [Work supported by NASA through Grant No. NAG8-1351.]
Agglomeration and collection of suspended particles using resonant acoustic fields within porous media
Authors: Feke Donald L. , Penrod Alan E.
Published: July 2002

The collection of micron- to millimeter-size solids from flowing suspensions is of fundamental importance in many chemical and materials processing technologies. Recently, a technique has been developed that enables the collection of such solids within porous meshes and other filtration media having pore size up to 200 times larger than the particles being retained. This is accomplished by energizing the filtration medium with a low power, one-dimensional resonant acoustic field. In laboratory tests, collection efficiencies up to 90% have been achieved. The retention of the solids by the porous medium is reversible, and the solids can be easily flushed from the filter as soon as the acoustic field is terminated. In this presentation, data that illustrate the agglomeration and collection effects will be provided. The three possible mechanisms that can cause the observed phenomena will be discussed. In order to model the particle collection phenomena, a particle trajectory analysis has been completed. In this model, the path of individual particles toward elements of the porous mesh are calculated as a function of the fluid flow conditions and the acoustic force acting between the particle and collector. Design strategies for optimization of the particle collection phenomena will also be presented.
Sonoluminescence light emission
Authors: Hilgenfeldt Sascha , Lohse Detlef , Grossmann Siegfried
Published: July 2002

Extending the hydrodynamical/chemical approach to single-bubble sonoluminescence (SBSL) to include light emission, it is shown that a few well-known, simple physical concepts are sufficient to account for the observed characteristics of emitted light pulses: Rayleigh–Plesset bubble dynamics, polytropic gas heating, dissociation of molecular gases, and thermal radiation of optically thin bodies. With these ingredients, a model can be developed which yields correct widths, shapes, intensities, and spectra for the pulses of emitted SBSL light. These quantities all depend in a characteristic way on each other, as well as on experimental parameters like driving pressure, water temperature, or dissolved gas concentration. A direct comparison of the theoretical results to experimental findings shows good agreement. The theory furthermore predicts that strongly forced xenon bubbles should show a dependence of the pulse width on light wavelength, in contrast to argon bubbles. Moreover, if upscaled SBSL bubbles of larger size at larger driving pressures can be created (e.g., using smaller driving frequencies), the resulting light emission intensities should be greatly increased.
The search for electrical discharges from single- and multi-bubble sonoluminescence
Authors: Bezzerides Vassilious J. , Matula Thomas J. , Crum Lawrence A.
Published: July 2002

Previous research supported the notion that electrical discharges resulted from the collapse of bubbles in cavitation fields [G. Giminez and F. Goby, in Cavitation and Inhomogeneities in Underwater Acoustics, edited by W. Lauterborn (Springer-Verlag, New York, 1980), pp. 101–107]. Further research was performed in a cavitation field, as well as with a single cavitating bubble. The cavitation field was generated in a standing wave-field using gassy water, resulting in a small confined region of cavitation. Single-bubble cavitation was also performed in this system, but with degassed water. Two methods were used to detect electrical pulses. In one case a small probe consisting of two fine wires coupled to a low-current differential amplifier was placed near the cavitation bubble(s). Signals were only observed when the probe was in contact with the cavitation field (no signals were observed in the single-bubble system). The second method utilized a coiled loop of wire surrounding the levitation cell, connected to a high-bandwidth lock-in amplifier. In this case, the reference frequency was scanned from about 25 kHz to more than 20 MHz. Signals were observed at harmonics of the drive frequency. [This research supported by DOE-EM.]
Amplification and compression in the cochlea
Authors: Russell Ian
Published: July 2002

The large dynamic range of the auditory system is due to processes within the cochlea which amplify low-level and compress high-level signals. Both the amplification and compression are associated with outer hair cell function and are reflected in their voltage responses to tones. Outer hair cells control the mechanical properties of the cochlear partition for frequencies close to the peak of the traveling wave which propagates along it. Thus they also control the mechanical input to the inner hair cells and excitation of the auditory nerve fibres. The amplification of basilar membrane responses can exceed 60 dB for high-frequency, low-level tones close to the characteristic frequency of the measurement place. For responses to levels above about 40 dB SPL, hair cell and basilar membrane responses are compressive and increase at a rate close to 0.2 dB per dB SPL. The cochlear efferent system has been implicated in controlling compression, but indications of the underlying mechanism of compression can be gained from measurements of the spatial distribution of basilar membrane responses both along its length and width, and from temporal changes in these responses to tones.
Indirect determination of input forces generated by the pump of a watercraft under operating conditions: Some guidelines
Authors: Janin J-M. , Champoux Y. , Attala N.
Published: July 2002

Engineers are currently able to calculate the dynamic characteristics of a complex structure using numerical tools. However, from an optimization point of view, it would be more valuable to predict correctly the actual response of the structure under working conditions. That requires that the engineers have the ability to determine precisely the input forces that are injected to the structure while operating. The direct measurement of the input forces is often impossible. That is due to the difficulties in mounting transducers and to the additional modifications it implies on the structure. Indirect measurement of forces has already been applied to a few industrial structures with some success, but no guidelines on the method have been given yet. In this paper, the work that has been done on the determination of the forces generated by the pump of a watercraft under real operating conditions, i.e., in the water, where direct measurements are made with difficulty, is presented. A comparison of the different inversion techniques (pseudoinversion, TLS) has been made. Attention here has been focused on clearly defining limitations and guidelines concerning the use of indirect force measurement methods in order to obtain results with a good confidence level.
Data-driven speech synthesis
Authors: Campbell Nick
Published: July 2002

Described in this paper are the theoretical background and implementation of a speech synthesis engine that uses an index of features describing a natural speech source to provide pointers to waveform segments that can then be re-sequenced to form novel utterances. By efficiently labelling the features in speech that are minimally sufficient to describe the perceptually relevant variation in acoustic and prosodic characterisitcs, reduce the task of synthesis to ‘‘retrieval’’ rather than ‘‘replication,’’ and is reduced reuse of original waveform segments is possible without the need for (perceptually damaging) signal processing. The drawback of this system is that it requires a large corpus of natural speech from one speaker, but current improvements in data-storage devices and cpu technology have overcome this problem. The style of the corpus speech determines the style of the synthesis, but experiments with corpora of emotional speech confirm that by switching source corpora one can easily control the speaking style. By shifting ‘‘knowledge’’ out of the synthesizer into the source data an engine is produced that can work on any adequately labelled speech corpus. The interesting work for the future lies in determining which features are relevant to capture the variation in speech, and how they can be best described.
Domain-specific prominence-based concatenation
Authors: Portele Thomas
Published: July 2002

Some recent approaches to unlimited text-to-speech conversion do not use ‘‘designed’’ inventories of concatenative units like diphones. Instead, an annotated corpus of read speech is searched for the realization of a speech segment whose features best match the corresponding ones demanded by the synthesis input. Crucial for this approach are the size and the variety of the corpus, the number and kind of annotations, and the definition of ‘‘best match’’ based on these annotations. In this implementation, the parameter ‘‘perceived prominence’’ is relied on as the most important parameter for the selection of a prosodically appropriate realization. A previous investigation with a German corpus indicated, however, that a corpus has to be very large in order to allow prosodic variations (e.g., different focus placements) of one utterance while retaining an acceptable intelligibility, if no post-selection signal modification is applied. The application of the same procedure to an American English corpus with domain-specific utterances (travel planning) indicates that the restriction to specific domains yields better results, but that signal manipulation is inevitable in order to obtain intelligible speech.
Cortical representation of language in monolinguals and bilinguals
Authors: Mehler Jacques
Published: July 2002

A presentation will be made of experiments trying to evaluate language representation in monolingual and bilingual subjects. The impact of distance of the languages on the cortical representations of L1 and L2 will be thus evaluated. Furthermore, also considered will be the age at which L2 has been acquired and the degree of proficiency that the bilinguals have attained in L2. Data will be presented from PET, fMRI, MEG, and high-density evoked potentials. Results tend to show that level of proficiency is more important than age of acquisition to explain the cortical representation of L2. Comparing and contrasting standard psycholinguistic evaluations with imaging studies, the respective advantages and shortcomings will be discussed. In the second part of the presentation the hypothesis that languages can be arranged into a limited number of classes will be explored. A review of work showing that infants discriminate a change of language only if the test and habituation languages have different rhythmical properties will be presented. The notion that languages can be sorted into rhythmical classes can be evaluated experimentally and results will be presented.
Adapting head-related transfer functions to individual listeners
Authors: Bronkhorst Adelbert W.
Published: July 2002

A highly efficient data reduction scheme will be described that captures the interindividual variability, present in head-related transfer functions (HRTFs), in only a limited number of parameters. The techniques used are principal component analysis and least-squares fitting of orthogonal spherical functions. The distortions introduced by the data reduction were evaluated using a distance measure based on the dB differences between HRTFs within 1/6 octaves. The distance measure correlates strongly with results of listening tests in which original HRTFs were compared with HRTFs reconstructed from the compressed data. Adaptation of HRTFs to an individual can be performed by reconstructing a set of HRTFs using estimated parameter values. Two methods for estimating these values will be discussed: prediction based on anthropometrical data and optimization using a simple listening test.
Acoustic source calculations for nuclear bursts
Authors: Clarke Douglas B. , Harben Philip E. , Harris David B.
Published: July 2002

Research has been conducted on the source term for long-range underwater propagation of signals from nuclear explosions in and above the ocean, in support of CTB monitoring objectives. A suite of source-region simulations is reviewed to study the variation of wave properties and source-region energy partition as a function of height or depth of burst. The multistep calculation combines LLNL’s CALE hydrodynamics code in the strong shock region with NRL’s weak shock code, NPE, at intermediate ranges. The source term calculations are intended as a starter field for long-range linear propagation models to obtain signature estimates at normal observational ranges. Calculations are presented to examine the effect of sea ice below an air burst on acoustic coupling into the water column. The ice was modeled as a continuous elastic layer 5-m thick at the water surface, and the source as a 1-Kt explosion 50 m above the ice. Calculations with and without the ice layer predict a moderate, but noticeable reduction in coupled acoustic energy in signals observed at the 10000-m range. [Work performed under the auspices of the U. S. Department of Energy by the Lawrence Livermore National Laboratory under Contract No. W-7405-ENG-48.]
Hydroacoustic monitoring of the oceans
Authors: Caristan Y.
Published: July 2002

In order to verify compliance with this Treaty, an international Monitoring System based on four technologies, seismology, infrasounds, hydroacoustic, and radionuclides, shall be implemented. Therefore, a network of hydrophones and seismic T-wave stations has been designed and evaluated in terms of detectability and location capability. Hydroacoustic waves generated by an undersea explosion or by oceanic earthquakes are propagating over several thousands of kilometers within an oceanic waveguide called So.F.A.R., channel (sound fixing and ranging) as shown by Ewing and Tolstoy (1950). When they are reaching any continental slope or island they are converted in seismic T-waves which are recorded by seismographs. By using both numerical simulations and experimental analysis, a methodology has developed which allows to better understand and quantify the effects of the source parameters and of the propagation characteristics (specifically on the So.F.A.R. channel and of the continental slope) on the recorded waveforms. The results obtained should be useful to improve the evaluation of detectability and location capability of the future hydroacoustic network and should provide some effective guidance to the site survey process. For that purpose and also to have a better understanding of the source parameters and of the acoustic wave propagation, a methodology has developed using both a numerical simulation and an experimental analysis.
Rapid assessment of undersea ambient noise
Authors: Wagstaff Ronald A. , Hollett Reginald D.
Published: July 2002

Undersea ambient noise is a critical parameter in determining the performance of passive sonar systems, and sometimes active systems. There are many properties of the ambient noise that must be considered in a rapid ambient noise field assessment, including mean and percentile levels as a function of frequency, temporal and spatial variation, vertical and horizontal directionality, and depth dependence. In addition, rapid interpretation and extrapolation of the measured and analyzed ambient noise results require a credible onboard ambient noise prediction model and collateral noise model input measurements of oceanographic, acoustic, meteorological, and noise source parameters, e.g., sea and swell height, wind speed, and the positions, speeds, courses, and lengths of near and distant ships. Finally, turnkey data processing programs and time proven analysis procedures must be utilized with onboard graphics and word processing capabilities to generate the assessment document onboard in final form. The methods and techniques used by the SACLANT Undersea Research Centre (SACLANTCEN) and the Naval Research Laboratory (NRL) for rapid onboard assessment of the ambient noise are presented and discussed. [Work supported by Project 01 of SACLANTCEN and the Office of Naval Research through the management of the WISPR Filter Development and Evaluation Project by NRL.]
In-situ update of geoparameter databases
Authors: Holland Charles , Osler John , McDonald Edward , Ferla Carlo
Published: July 2002

Low- and mid-frequency system performance is often controlled and degraded by bottom interaction, especially in littoral regions. However, performance prediction models generally give unreliable estimates of bottom interaction. The poor estimates are most often due to the databases that drive the models rather than the model physics. In this study a new approach is taken in rapid environmental assessment for overcoming this problem. The concept is to use in-situ propagation data to extract geoacoustic information about the bottom which is then used to update the bottom database. An analogous approach has long been employed for the water column, where in-situ bathythermograph (BT) data are used to update historical databases. While techniques for screening/merging BT data are now under development, there are no comparable techniques for treating bottom geoacoustic data. In this paper, a high-level methodology for updating bottom databases with in-situ measurements is outlined. The methodology includes a series of critical questions sequenced to lead to decisions about the robustness of the historical databases in a given region and the potential for database update. Results of the methodology, i.e., the in-situ measurements, modeling, and resulting bottom geoparameter database updates, from several littoral areas, will be shown.
Estimation of the channel scattering function
Authors: Kay Steven , Doyle S. Bradford
Published: July 2002

The estimation of the range-Doppler scattering function for an arbitrarily spread (underspread or overspread) channel is addressed. Conventional methods typically require collection of reverberation data from many pings over time to characterize the channel accurately. A new method is proposed based on a minimum mean-square error (MMSE) estimator of a single realization of the channel’s time-varying frequency response. Next, a two-dimensional autoregressive spectral estimation technique is used as a scattering function estimator. This approach does not suffer from the usual convolutional smoothing with the signal ambiguity function encountered with Fourier-based methods. The method is compared to traditional methods through simulation. Insight into signal design based on this new method is also provided. [Work supported by NUWC Division, Newport.]
Influence of sensory interactions between vision and audition on the perceptual characterization of room acoustics
Authors: Nathanail Chrysanthy , Lavandier Catherine , Polack Jean-Dominique , Warusfel Olivier
Published: July 2002

Broad evidence on sensory interactions between vision and audition suggests that visual information available to listeners-spectators in concert halls is likely to interfere in the evaluation process of the acoustical qual- ity. The influence of the stage visual distance on the auditory apparent distance was studied in four magnitude estimation paradigms. Sound stimuli providing successive impressions of auditory distance were generated by a virtual room acoustics processor (‘‘Spatialisateur’’) in binaural format and presented separately or coupled with 3-D concert hall pictures. Subjects were asked to judge the auditory distance. Pilot experiments I and II obtain a first qualitative observation of the effects and study a progressive change observed at the responses. Main experiments III and IV adopt two different procedures of stimuli presentation and improved sound reproduction and attempt a first objective evaluation of the results. The analysis of variance performed on the responses reveals a small offset-type influence of the visual stimuli on the auditory distance: i.e., the same auditory stimuli appear more distant when presented with distant stage views. In addition, differences between unimodal (auditory stimuli alone) and bimodal tests suggest that vision enhances the ‘‘depth’’ of the auditory field and provides a spatial frame of reference for auditory judgments.
Appropriate bandwidth and integration time for measurements of the degree of interaural cross correlation as a measure of apparent source width in concert halls
Authors: Morimoto Masayuki , Iida Kazuhiro
Published: July 2002

ISO 3382 Annex suggests using the degree of interaural cross correlation (ICC) as a measure of apparent source width (ASW), while mentioning that the choice of bandwidth for test signals and the integration time are uncertain. Needless to say, what is important for a physical measure is that it is well correlated with the subjective effect, in this case ASW. This paper describes psychological experiments to investigate the appropriate bandwidth for test signals and the integration time for measuring ICC to evaluate ASW. The experimental results indicate that (1) measurements of ICC with 1/3-oct bands are preferred for evaluating ASW, whereas the use of wide band and 1/1-oct band signals as described in the ISO standard are not, and (2) measurement of ICC for all reflections including both the early reflections and later reverberant sound is most suitable for evaluating ASW, rather than the early reflections alone.