bluuebunny/crossref_metadata_2025 huggingface dataset:
Second-order thermoacoustics
Authors: Waxler Roger
Published: July 2002

A theoretical study is made of nonlinear effects in thermoacoustic stacks. The equations of fluid dynamics and heat transfer are expanded about a mean state with constant pressure and zero velocity but with a nonzero temperature gradient along the stack. The expansion is carried out to second order in Mach number. Using Rott’s approach to control the linear terms, the time-independent parts of the second-order terms are studied. A set of coupled equations for the radiation pressure, velocity streaming, and time-averaged temperature are obtained and solved. The transverse heat flux into the stack is obtained. The solutions depend on the pressure and temperature at the ends of the stack, and thus on the details of the nonlinear effects in the resonator. However, it is found that the effects of the resonator on the velocity streaming and radiation pressure can be reduced to two parameters.
Quantifying correlations in pitch- and amplitude contours
Authors: Herzel Hans-Peter , Reuter Robert
Published: July 2002

Nonlinear phenomena such as subharmonics, biphonation, or deterministic chaos induce characteristic patterns in pitch and amplitude contours. A novel measure of correlations in contours termed MAC (mean average correlation) is introduced. For uncorrelated and weakly correlated noise, the expectation values and standard deviations of MAC are calculated. It is shown that trend elimination induces spurious correlations. Moreover, simulations of the circle map are analyzed and high values of MAC for subharmonic and chaotic data are found. Examples are presented of severely rough voices and excised larynx data and correlations are quantified in the corresponding contours by MAC. Finally, the connection between perceived voice quality and MAC is studied with the aid of 120 perceptually evaluated voices. [Work supported by the Deutsche Forschungsgemeinschaft.]
A design of laryngeal structures for a physiological articulatory model
Authors: Wu Chao-Min , Dang Jianwu , Honda Kiyoshi
Published: July 2002

A model of the hyoid–larynx complex is designed and incorporated into a physiological articulatory model. The previously reported articulatory model [Dang and Honda, Proc. ICSLP98] consists of the tongue, mandible, hyoid bone, and vocal tract wall. The aim of this work is to include the laryngeal structures to simulate the tongue–larynx interaction which is observed in natural speech. The framework of laryngeal structures is composed of the thyroid, cricoid, and arytenoid cartilages, which are connected by muscles, ligaments, and joints. The form and geometry of the cartilages are extracted from volumetric MR images of a male speaker and represented in a quasi-3D shape. They are modeled to yield joint rotation and translation as well as vertical movements. The ligaments and muscles are represented by mass points with viscoelastic springs, and the cartilages are composed by springs with an extremely large stiffness. Laryngeal factors of F0 control such as cricothyroid rotation and laryngeal vertical movement are demonstrated in the simulation, and plausible dynamic behaviors of the tongue–larynx interaction are observed. The performance of the model is tested by comparison of acoustic output from the model with recorded speech.
Magnetoencephalographic evidence for temporal representations in human auditory cortex
Authors: Poeppel David E. , Roberts Timmothy P. L. , Greenberg Steven
Published: July 2002

It is generally assumed that the auditory representation of speech is derived primarily from the spatial distribution of activity across the tonotopically organized neuronal axis. Such temporal parameters as spike synchronization, interspike interval, and neuronal oscillations are also regarded as potential complements to the classical tonotopic representation of the speech spectrum. The present study used magnetoencephalography (MEG) to investigate the temporal behavior of large neuronal ensembles in human auditory cortex. Experimental stimuli were sinusoidal and square-wave signals, sinusoidally amplitude-modulated tones, and synthetic vowels. The results indicate the presence of a systematic representation of the signal spectrum and pitch derived from the latency pattern of the major auditory-evoked neuromagnetic field. The magnitude of latency differential to signals of variable frequency and pitch can be as large as 25 ms. This ‘‘tono-chronic’’ analysis may offer an alternative encoding mechanism for pitch and spectral information that plays an important role in the perception of complex signals such as speech. [Research supported by the James S. McDonnell Foundation and the National Science Foundation.]
An oscillatory correlation framework for the separation of speech from interfering sounds
Authors: Brown Guy J. , Wang DeLiang L.
Published: July 2002

Speech is seldom heard in isolation—usually, a mixture of sounds reaches our ears. Hence, the auditory system is faced with a scene analysis problem—it must group together spectral components that are likely to have arisen from the same environmental source. At the neurobiological level the auditory system also faces a binding problem—it must construct meaningful wholes from features that are represented in separate neural structures. Models of auditory function based on the theory of oscillatory correlation offer a potential solution to the scene analysis and binding problems. According to this theory, perceptual streams are represented by populations of neural oscillators, such that oscillators belonging to the same stream are synchronized (phase locked with zero phase lag), and are desynchronized from oscillators belonging to different streams. A neural oscillator model of auditory function has been developed, capable of segregating speech from interfering sound sources. The core of the model is a two-layer neural oscillator network, in which lateral connections among oscillators encode harmonicity as well as proximity in time and frequency. A variant of this model is able to account for the sharing of acoustic evidence among streams, as would occur in ‘‘duplex’’ perception of speech.
Forward/backward first-order statistics algorithm for the estimation of DOA in a multipath environment
Authors: Song Joon-il , Kim Han-Su , Sung Koeng-Mo , Lim Jun-Seok
Published: July 2002

Many kinds of direction of arrival (DOA) estimation methods in the coherent environments have been studied for many decades. Most of them use spatial smoothing to tackle the coherent environments. An alternative method was proposed by Pillai which used first-order statistics on each sensor datum [S. U. Pillai, Array Signal Processing (Springer-Verlag, New York, 1989)]. The method handled the complex numbered input data with additional symmetric array so that it needed two times as many sensors as it needed in the case of the real numbered input data. In this paper a DOA estimation method using forward/backward first-order statistics is proposed which can handle the complex numbered input data without additional sensors. [Work supported by UARC.]
Coupling criteria for arrays in a channel
Authors: Gaumond Charles F.
Published: July 2002

In this paper, an analogy is made and developed between a linear array of receivers with far-field sources in an unbounded, homogeneous medium and an arbitrary array of receivers with arbitrarily placed sources in a channel. The transfer function of a signal transmitted through a channel is the well-studied transmission loss (TL). The singular value decomposition (SVD) is performed on the transfer function between sources and receivers. The three components of the SVD are related to various factors in the case of the linear array in an unbounded, homogeneous medium. The case of a translationally invariant channel described using modes is considered. This modal description is also related to the three components of the SVD. The sum of the eigenvalues of the SVD is proposed as a single number which quantitatively describes the amount of coupling between an arbitrary array and the sound channel, assuming that the distribution of sources spans the propagating modes. This single number can then be used to assess the ability of various array configurations to sample the sound channel. [Work supported by ONR.]
Modeling animal FM tones using a cubic spline interpolation
Authors: Douaze Etienne , Delory Eric , Potter John , Taylor Elizabeth
Published: July 2002

The use of FM tones for communication is widespread among birds and mammals. Analysis of these signals by biologists is currently limited to a qualitative visual inspection of spectrograms and manual logging of discrete signal features from a spectrogram window. Human-related measurement errors are usually not considered and a robust numerical model of animal FM tones is lacking. A model should take at least three signal parameters into account: frequency and amplitude modulation, and the harmonic content. In this work a numerical model for animal FM tone fundamentals is presented. A piecewise polynomial interpolation is used to fit bottlenose dolphin (Tursiops truncatus) FM tones, commonly called whistles, from manually logged spectrogram fundamentals. Cubic splines are used to achieve the best visual fit to the signal fundamental by controlling locally the slope and the curvature of the curve fit. A smoothing factor, calculated from an independent experiment on six human subjects, is applied to the interpolation to adjust for human measurement errors. Dolphin tone fundamentals are modeled with a set of 80 coefficients, a number conveniently independent of signal duration because longer dolphin whistles are observed to be associated with a lesser degree of frequency modulation.
A low-voltage silicon condenser microphone for hearing instrument applications
Authors: Rombach Pirmin , Müllenborn Matthias , Klein Udo , Nielsen Lis , Frehoff Roger
Published: July 2002

Silicon microphones have been the subject of investigations since the early 1980s. Due to their poor performance, no silicon microphone for hearing instrument applications has been commercially available until today. Usually the sensitivity of the electromechanical transducer is too low. Thus the input-related noise of the following preamplifier stage becomes dominant and results in a high equivalent input-related noise. Here a silicon condenser microphone with the potential for hearing instrument applications will be presented. To get the best properties for the different mechanical parts, e.g., membrane and back plate, a dedicated process sequence has been developed. Therefore, circuitry and mechanical parts have to be produced separately and mounted later in a stacking process. The microphone has a 2×2 mm2, 0.4 μm-thick membrane and an air gap of 1.0 μm. Wafers with different membrane stress have been produced. The microphones have been acoustically, mechanically, and electrically characterized, partly on wafer-scale, and compared with an acoustical/electrical lumped element model. The sensitivity for the low-stress microphones using a 1.5 V power supply is about 7.0 mV/Pa and the input related A-weighted noise is 23 dB SPL, including the preamplifier. Due to a conservative layout, the parasitic capacitance is about 50%. An increase of 2–3 mV/Pa sensitivity and hence 3 dB SPL less noise can therefore be achieved by design optimization.
An integrated-optical silicon microphone with a Fabry–Pérot cavity as modulator
Authors: Klaiber Martin
Published: July 2002

A new construction principle for an integrated-optical microphone is presented. It uses a Fabry–Pérot cavity as modulator. This combines the very sensitive interferometric multiple reflection scheme with a direct intensity modulation. The microphone consists of two chips, the carrier chip and the membrane chip. Both are fabricated in silicon using standard semiconductor technology. The membrane chip is mounted on top of the carrier chip, so that both chips together form the Fabry–Pérot cavity. Unmodulated primary light is generated by a laser diode and partially transmitted to the microphone via a 3-dB coupler. A single fiber, which is attached to the bottom side of the carrier chip, connects the coupler and the microphone. The incoming light is modulated in the Fabry–Pérot cavity by the membrane deflection and is reflected back into the fiber. It passes the coupler in reverse direction and is thereby partially transmitted to a receiver device, e.g., a photo diode. The main advantage of this kind of microphone is that a very low inherent noise level (<30 dB/A) can be achieved with a relatively simple design.
Active noise reduction in an ear terminal
Authors: Ottesen Georg E.
Published: July 2002

An active ear terminal is being designed to obtain both hearing protection and radio communication in adverse environments. Passive damping is sufficient for high frequencies, but low frequencies must be damped by active methods. Even when low-frequency noise is not harmful to the hearing, it heavily increases the effort of communication. Active noise reduction may use the signal from an outer microphone, or from an inner microphone. Active noise reduction obtained by these two microphone positions are simulated and discussed. Analog and digital filters are considered. The time delay budget of the active system is drawn up. Time delays should be minimized to obtain the best performance of the active noise reduction system. The physical system of transducers with their acoustic surroundings contains time delays which cannot be compensated by filtering. These time delays can be determined from the transfer functions of the system. The physical design of the ear terminal is performed with a view to reduce these delays. The signal prosessing delay adds to the delay of the physical system. The total delay determines the upper frequency limit of the active noise reduction system.
Design considerations for a low-impedance loudspeaker in active noise control exhaust systems
Authors: Kahn Steven P.
Published: July 2002

To meet the needs of automotive original equipment customers, a low-impedance loudspeaker has been designed for active noise control exhaust systems that reduces system cost, decreases overall weight, eliminates complexity, and improves reliability. The current design practice is to convert the 12-V dc supply of the car battery to 36 V dc. This supply voltage is used in conjunction with a switching amplifier to drive the noise canceling loudspeaker. By utilizing a low-impedance loudspeaker, the 12-V dc supply from the battery can be fed directly to the amplifier, thereby eliminating the dc–dc converter completely. This approach also simplifies the cable harness and reduces the number of connectors. To develop a low-impedance loudspeaker that allows the realization of these advantages, one must consider the following: output level, electrical damping, frequency bandwidth, power handling, and manufacturability. The interaction and trade-offs between these issues must be understood and optimized. Test results between the current configuration and the new approach will also be presented.
Self-adaptive absorber of underwater sound with electrorheological fluid
Authors: Guicking Dieter , Michaelis Patrick , Neumann Helge
Published: July 2002

Laboratory experiments have been made on an adaptive-passive absorber of underwater sound with viscous flow in ERF (electrorheological fluid) as absorption mechanism. The absorber model is of the velocity transformation type where a thin fluid layer is confined between an inertial back mass and a piston. Axial vibration of the piston by the incident sound wave causes a strong radial fluid oscillation and thereby enhances the sound absorption. With ERF as fluid, the reflection coefficient depends on the gap width and the field strength. In the present setup the gap width can be varied by a stepper motor. A two-parameter feedback control strategy employing a genetic algorithm has been developed to adjust the gap width and the field strength to minimize the reflectivity as measured by the two-microphone method in an impedance tube. The optimum of 10% to 20% reflection is reached within a few generations of adaptation.
Experimental and modeling results for near-grazing propagation over rough surfaces
Authors: Chambers James P. , Sabatier James M.
Published: July 2002

Recent experimental work on acoustic propagation over rough surfaces has shown intriguing propagation effects ranging from frequency shifts in the ground interference dip to surface wavelike phenomena. Attempts to model the observed data with an effective impedance formulation have had some success but have met with some problems for certain propagation geometries and surface configurations. Experimental data and modeling results will be presented and discussed for a variety of surfaces ranging in surface properties from impermeable to loosely packed soil and ranging in surface roughness scales from low sloped perturbations to steeply sloped wedges.
Acoustic remote sensing of seafloor roughness: Resolving ambiguous estimates of roughness spectrum parameters
Authors: de Moustier Christian P. , Sternlicht Daniel D.
Published: July 2002

To predict or model how sound waves interact with the seafloor some parametrization of the roughness of the sediment–water interface is required. Conversely, bottom backscattered echoes contain information about the roughness of the interface, relative to the acoustic wavelength, often mixed with contributions from inhomogeneities in the sediment volume. However, inversions of seafloor roughness parameters from high-frequency (10–100 kHz) acoustic backscatter measurements often produce ambiguous results with numerous plausible parameter combinations fitting a given acoustic backscatter angular dependence curve or an echo envelope shape. Based on the data and on direct or inferred seafloor roughness measurements reported in the literature, and assuming that the relief spectrum of seafloor topography obeys a power law, it is shown that a level of inversion ambiguity can be lifted by setting the roughness spectral exponent to a value appropriate for the sediment type and by iterating on the spectral strength and the allowable amount of sediment volume backscatter. This is illustrated with examples drawn from comparisons between seafloor echo envelope models and measured acoustic backscatter data yielding interface roughness spectral parameters and rms bottom curvatures consistent with the lithology of the areas investigated. [Work supported by ONR N00014-94-1-0121.]
Psychoacoustics and noise abatement in Austria—Some practical experiences
Authors: Kalivoda Manfred T.
Published: July 2002

In Austria there are two main fields of application for psychoacoustic methods. For noise from trade and industry the Gewerbeordnung (trade act) does not use any SPL limits. There is a more stringent regulation that says that an industrial plant has to be licensed only if neither a health risk nor intolerable nuisance or annoyance can be expected. The Umweltverträglichkeits-Prüfungs-Gesetz (environmental assessment act) from the year 1994 includes industry and all modes of transport uses the same wording as the Gewerbeordnung. Nevertheless, A-weighted equivalent levels are frequently used for noise assessment in these two fields. There are many situations where it makes no big difference if LA,eq or psychoacoustics is used. However, there is about 10% to 20% of noise problems for which SPL alone is not adequate so wrong conclusions are drawn on the noise impact. psi-A Consult is very experienced in applying psychoacoustics on real noise problems. Some examples, showing the discrepancies between conventional and psychoacoustic noise assessment, are presented here.
Investigations in ultrasonic tomography
Authors: Lefebvre Jean-Pierre , Lasaygues Philippe , Wirgin Armand , Delamare Sebastien , Mensah Serge
Published: July 2002

Ultrasonic tomography (UT) is a technique to reconstruct the spatial distribution of some physical parameter of an object from measurements of the scattered field. The measurements are made for more or less dense sets of emitter and receiver positions and of frequencies of the interrogating wave. This inverse scattering problem was solved by using a Born approximation that leads to a particularly simple and attractive linear relation between the object function (OF) and the scattered field, particularly in the far-field (2-D or 3-D Fourier transform), making it possible, in principle, to reconstruct the OF in near real time for a sufficiently large set of scattering data. Ultrasonic tomography was investigated numerically and experimentally. Numerical simulations, using ideal measures with ideal objects, allow the examination in detail of the influence of various parameters such as the object’s dimension and contrast, transducers bandwidth, etc. It allows one to analyze what happens when the Born approximation is no longer valid (high frequencies, high contrasts), to find limits of quantitative and qualitative imagery, and to imagine various improvement procedures (e.g., superresolution procedures leading to high resolution with low frequencies). Experimentations (with a mechanical and an antenna-based system) show the applicability of the method for medical and materials applications.
Generation of a standing wave field using crossing sound beams for acoustic micromanipulation
Authors: Kozuka Teruyuki , Tuziuti Toru , Mitome Hideto , Fukuda Toshio , Arai Fumihito
Published: July 2002

A novel sound-field generation method for noncontact manipulation of particles has been studied experimentally. Although it was possible to trap particles in water at nodes of a standing wave field generated between a transducer and a reflector and to transport them using a frequency-shifting operation, there were several problems, such as unstable force due to resonance of the sound field and different movement of trapped particles. The present paper describes a method to generate a standing wave field using plural sound sources without a reflector. When two transducers were settled with their sound-beam axes crossing to each other, a standing wave field was generated in the crossing region. Changing the mutual phase of transmitting signals between the transducers, the sound field was shifted laterally, keeping the same form. When polystyrene particles were poured with a pipette into the sound field, they were trapped at the nodes of the sound-pressure distribution and were transported stably. Using three transducers whose sound-beam axes were crossed at an angle of 120° to each other in a plane, a standing wave field was generated with a hexagonal node pattern, and two-dimensional manipulation of a particle was accomplished.
Water vapor and sonoluminescence
Authors: Szeri Andrew J. , Storey Brian D.
Published: July 2002

In recent work, it has been shown that segregation of gas mixtures within sonoluminescence bubbles is driven by strong thermal and pressure gradients inside the bubble [B. D. Storey and A. J. Szeri, ‘‘Mixture segregation within sonoluminescence bubbles,’’ in review]. Although these rather exotic types of diffusive transport are often neglected, they are of crucial importance in a number of applications. These effects conspire to increase the fraction of the lighter gas in a mixture of noble gases near the center, and to increase the fraction of the heavier gas near the bubble wall at the moment of extreme collapse. This results in a greater energy concentration at the bubble center. While the temperature and pressure peaks have a duration (FWHM) of only a few hundred picoseconds at most, it has been found that appreciable mixture segregation persists for several orders of magnitude longer. Armed with this new understanding of how gas mixtures behave, it is possible to investigate more deeply the question of water vapor within the bubble. Several investigators have shown that the presence of water vapor has important consequences for compression versus shock heating, for hydrodynamics of the bubble collapse, and for chemistry.
Nonlinear cochlear processing of harmonic complexes
Authors: Leek Marjorie R.
Published: July 2002

Some harmonic complexes with component phases that change monotonically with harmonic frequency are characterized by temporal waveforms with relatively flat envelopes. With an appropriate choice of component phases, the fine structure within periods represents an upward or downward sweep of instantaneous frequency. One such class of phase selections creates ‘‘Schroeder-phase waveforms,’’ named after the developer of the phase algorithm. These stimuli have been used recently in a number of psychoacoustic studies because of some intriguing characteristics of the waveforms, along with some compelling hypotheses regarding the interactions of the waveforms with cochlear processing mechanisms. Properties of Schroeder waveforms will be discussed in relation to hypotheses regarding internal alteration of waveform shapes by normal and impaired cochlear processing. Studies will be reviewed that involve the effects of phase selection on discrimination of temporal waveforms, as well as masking by Schroeder-phase stimuli in normal and impaired auditory systems. Results from these studies appear to reflect properties of the traveling wave within the cochlea, as well as evidence of active nonlinear gain in normal cochleas. [Work supported by NIH DC00626.]
Relationship between loudness perception and peripheral compression in subjects with unilateral cochlear hearing loss
Authors: Moore Brian C. J. , Vickers Deborah A. , Plack Christopher J. , Oxenham Andrew J.
Published: July 2002

Loss of compression on the basilar membrane (BM) associated with outer hair cell damage (OHC) is thought to be the main factor underlying loudness recruitment in people with cochlear heating loss. To assess this idea, subjects with unilateral cochlear hearing loss were tested. Recruitment was measured by obtaining loudness matches between tones presented alternately to the two ears. Compression was estimated from the slopes of growth-of-masking functions in forward masking. The ratio of slopes for a masker centered well below the signal frequency and a masker centered at the signal frequency gives a measure of BM compression at the place corresponding to the signal frequency [A. J. Oxenham and C. J. Plack, J. Acoust. Soc. Am. 101, 3666–3675 (1997)]. The measures of loudness recruitment were used to derive a parameter HLOHC, the amount of the hearing loss attributable to OHC damage [B. C. J. Moore and B. R. Glasberg, Auditory Neurosci. 3, 289–311 (1997)]. The value of HLOHC is directly related to the amount of recruitment. The estimated compression decreased systematically as HLOHC increased from 0 to 55 dB. These results are consistent with the idea that loudness recruitment is directly related to loss of compression on the BM. [Work supported by the MRC, UK.]
Benefits of nonlinear active gain for detection of brief probes in Schroeder-phase complex maskers
Authors: Summers Van , Seitz Philip
Published: July 2002

Masking period patterns for positive and negative Schroeder-phase complexes were determined for tonal probes at 1 and 4 kHz. Maskers included components from 200–5000 Hz of a 100-Hz fundamental. The 5-ms probes had onsets occurring 153, 155.5, 158, 160.5, or 163 ms following masker onset. For listeners with normal hearing, thresholds in the positive Schroeder-phase masker varied by as much as 25 dB depending on the position of the probe within the masker period. Thresholds in the negative Schroeder-phase maskers were nearly constant as probe onset varied. The maskers were more nearly equal in effectiveness for listeners with sensorineural hearing loss and in testing at high presentation levels. The findings support an interpretation involving differences in the shape of the basilar-membrane waveform generated by each masker, and influences of nonlinear active gain on these internal responses. The positive Schroeder-phase masker appears to produce a highly modulated basilar membrane response. Nonlinear active gain enhances the signal-to-masker ratio for signals presented during low-amplitude ‘‘troughs’’ of these modulated internal waveforms. Consistent with this account of the present findings, active gain is reduced or eliminated at high presentation levels and with cochlear damage.
Monaural and binaural minimum audible angles for virtual sound sources
Authors: Bolia Robert S. , Musicant Alan D. , Gajewski Daniel A.
Published: July 2002

A number of researchers have employed the minimum audible angle (MAA) paradigm to investigate the precision with which listeners can discriminate differences of sound-source location in the free field [A. W. Mills, J. Acoust. Soc. Am. 30, 237–246 (1958); W. M. Hartmann and B. Rakerd, ibid. 85, 2031–2041 (1989)]. Surprisingly, this has been done only for binaural listening conditions, although Man’s capacity for monaural localization is well documented [J. R. Angell and W. Fite, Psych. Rev. 8, 225–246 (1901); R. A. Butler, R. A. Humanski, and A. D. Musicant, Perception 19, 241–256 (1990)]. In the present study, monaural and binaural MAAs were determined using a broadband stimulus for seven virtual sound-source positions in the horizontal plane. Listeners indicated via a three-alternative, forced-choice paradigm whether the stimulus presented in the second interval appeared to move clockwise, counterclockwise, or not at all, with respect to that presented in the first interval. Results will be discussed in terms of data obtained in previous studies of binaural MAA, and compared with data on monaural and binaural localization acuity.
Effects of low-frequency thresholds on sound-source identification in the frontal horizontal plane using interaural time difference cues
Authors: Gatehouse Stuart , Lever Catherine , Lorenzi Christian , Howell Patrick
Published: July 2002

Eight listeners with symmetric sensorineural hearing loss were investigated using a sound-source identification paradigm, to study localization abilities in noise for sources in the frontal horizontal plane. Stimulae were low-pass filtered click trains making available the primary cue of interaural time differences. Compared to performance of a normally hearing reference group, the listeners with sensorineural hearing loss exhibited systematic deficits in localization ability which were strongly related to hearing level. However, parallel detection experiments also showed deficits and the localization data might be a function of simple audibility. The listeners were further tested using a number of binaural hearing aid fittings for which detectability was returned to levels displayed by the normal-hearing listeners. Aided localization ability did improve but still showed residual deficits compared to the normal-hearing references and, furthermore, still had a systematic relationship with hearing level. The gradient of this relationship was approximately half that in the unaided data. The results suggest that sensorineural hearing loss compromises the use of interaural time difference cues to achieve sound-source identification and that the deficit is in part independent of audibility considerations.
A vibroacoustic inverse problem regularized by SVD technique coupled with Miller’s method
Authors: Marquis-Favre C. , Hamzaoui N. , Boisson C.
Published: July 2002

This communication presents the resolution of a vibroacoustic inverse problem. This latter is a typical ill-posed problem introduced by a Fredholm integral equation of the first kind. This formulation is used in the mathematical model established for an approach dealing with numerical construction of the vibroacoustic transfer function for sound radiation prediction from an industrial structure operating in its real environment [Marquis et al., ICA (1995)]. The construction of the vibroacoustic transfer function is based on the use of acoustic pressures and vibratory velocity measurements defining the structure in situ. The acoustic pressures are data for computation of a density function allocated to points sources distributed on the structure. This calculation lies in solving a vibroacoustic inverse problem. This is carried out with the SVD technique. The regularization process is based on the Miller’s method using a priori information on the solution and on the measurement errors contaminating the data. Some numerical simulations for an academic vibroacoustic system illustrate the sensitivity of different parameters and their influence on the application conditions of the approach. Some practical indications concerning SVD and the regularization are given and some results show the regularization effects and the consequences in the presence of measurements errors.
Nonlinear modeling of the forward path in a mechanical active control system
Authors: Pettersson Linus , Hakansson Lars , Claesson Ingvar , Lago Thomas
Published: July 2002

In most active control applications the estimate of the forward path is closely related to the performance of the controlled system. Nonlinearities in the forward path are most likely to degrade the performance. In the active control of machine tool vibration in a lathe, the plant under control principally consists of two physical parts: amplifier and a tool holder construction with integrated actuators. One applicable actuator type was used and it was based on highly magnetostrictive material. However, these types of actuators generally have nonlinear behavior. To obtain deeper understanding on the dynamic properties of the plant under control and thereby obtain a solid base for the feedback control of the machine tool vibration, nonlinear modeling of the forward path was proposed. Two methods were used: The approach of neural networks and the reverse multiple-input single-output by Bandat have been used in order to describe the forward path as accurately as possible. In the active control system of machine tool vibration in a lathe the forward path is the transfer from the actuator amplifier to the output from the accelerometer mounted on the toolholder shank in the toolholder construction.
HMM-based database segmentation and unit selection for concatenative speech synthesis
Authors: Martin Holzapfel
Published: July 2002

In the proposed paper methods for improved HMM-based segmentation and unit selection for concatenative speech synthesis from a large corpus will be presented. The proposed algorithm is based on triphone HMMs. For training only the database to be segmented and its canonical phonetic transcription are used. Experiments on the optimization of these HMMs for segmenation are reported. Results in changes of the HMM topology, overadaption to the database, and variations on the acoustical features will be highlighted. Special attention is given to the tree-based clustering of the models according to their phonetic context. Contrary to other approaches, the clustering is performed on phone-model level. The clustering-tree is the basis of the phonetically motivated selection of the phone-sized speech units. Using the binary membership of those clusters as selection criterion results in a target conflict on the size of those clusters. (Tight phonetic description versus prosodic and contextual variety in one cluster.) Using the distance of the resulting nodes of the tree offers a still discrete but much finer scaled phonetic selection criterion overcoming the above-mentioned shortcomings. These techniques decreased the distances of the segmentation from manual labeling and significantly improved the overall quality of the Siemens TTS System PAPAGENO.
Toward further understanding of second language speech learning: An approach utilizing speech technology
Authors: Akahane-Yamada Reiko
Published: July 2002

Recent training studies have clarified several aspects of second language (L2) learning. However, many questions still remain unanswered, and further training studies are necessary. In this paper, today’s speech technology was applied to speech perception and production training. First, new speech resynthesis algorithm (STRAIGHT) was used to enhance training stimuli. Japanese speakers were trained to identify English /r/ and /l/ using several versions of synthetic stimuli with different manipulations. All the versions showed equivalent training effect as the original stimuli, suggesting that this resynthesis technique has a potential to enhance training materials. Second, two attempts were made to build an automated speech production training system, focusing on the issue of how to provide useful feedback for learners. Spectrogram and HMM-based speech recognition scores were attempted as feedback, and Japanese speakers were trained to produce /r/ and /l/. Subjects with either feedback improved significantly from pretest to posttest in their production ability as evaluated by human judges. A correlation was also found between evaluation scores obtained from HMM and human judges, further supporting the applicability of a HMM-based speech recognition system. Various approaches to optimize L2 speech training methods will be discussed in the context of the theories of speech perception/production development.
On what it takes to predict perceptual difficulty in cross-language vowel perception
Authors: Bohn Ocke-Schwen , Strange Winifred , Trent Sonja A.
Published: July 2002

According to current models of cross-language speech perception and second language (L2) speech learning, perceptual difficulty depends upon how listeners map the phonetic segments of the foreign language onto their native language (L1) categories. A number of indirect methods have been suggested for predicting perceptual difficulty, e.g., contrastive phonemic analyses and acoustic comparisons of L1 and L2 sounds. This presentation summarizes recent studies which directly assessed perceptual assimilation patterns for North German vowels (as perceived by American English listeners) and for British English vowels (as perceived by Danish listeners). Vowels were produced in CVC syllables in various consonantal contexts in both citation form and sentence frames. As expected, contrastive phonemic analyses were not good predictors of assimilation patterns. Somewhat surprisingly, acoustic comparisons didn’t always predict assimilation patterns either. In particular, measures of acoustic similarity from ‘‘canonical’’ forms didn’t predict acoustic similarity for coarticulated vowels, nor did these measures predict perceptual similarity. Our results strongly suggest that perceptual difficulty in cross-language speech perception cannot be predicted using indirect methods, nor can difficulty be predicted by perceptual studies of citation form utterances. Instead, successful predictions of perceptual difficulty must be based on direct assessments of assimilation patterns. [Work supported by NIDCD.]
Sound localization using other people’s ears
Authors: Middlebrooks John C.
Published: July 2002

When subjects attempt to localize virtual targets that have been synthesized with directional transfer functions (DTFs) recorded from other subjects’ ear canals, they typically show a considerable increase in localization error relative to the condition in which DTFs are recorded from their own ear canals. This study explored intersubject differences in DTFs and developed a procedure to reduce such differences, thereby improving virtual localization. DTFs varied systematically among subjects in regard to the center frequencies of spectral features such as peaks and notches. Generally, spectral features appeared at lower frequencies in the DTFs of larger subjects and at higher frequencies in DTFs of smaller subjects. Characteristic errors in virtual localization could be related to frequency differences between spectral features in the listener’s own DTFs and those with which virtual targets were synthesized. Scaling transfer functions in frequency by an optimal scalar could reduce intersubject variance in DTFs, sometimes by more than 50%. Frequency scaling of DTFs resulted in corresponding improvements in virtual localization. In the majority of cases, optimal frequency scaling reduced by more than half the increase in error that resulted from listening through another subject’s ears. [Work supported by ONR Grant No. N00014-94-0481 and NIH Grant No. DC00420.]
Propagation modeling of T-phase time series via RDOASES
Authors: Sperry Brian , Schmidt Henrik , Baggeroer Arthur
Published: July 2002

Properly situated earthquakes and explosive sources often radiate acoustic energy into the oceanic waveguide and are thereby detectable at long ranges. Details of how this energy actually couples into the ocean remain elusive. Recent development of a range-dependent version of OASES has made it possible to model acoustic propagation in range-varying, elastic environments. It has been very successful in investigating the effect of epicenter depth for T-phase excitation on a sloping bottom. The present study extends earlier work to consider time-domain characteristics of the arrival and their dependence upon properties of the source and its surrounding region. The general environment considered here is a Munk sound-speed profile and sloping bottom out to a range of 30 km, after which the bottom depth is constant. The seismoacoustic field is computed using RDOASES, and then projected onto the local mode shapes. The complex mode coefficients are propagated to a distant receiving array, where the time-domain signal is synthesized. Understanding the source excitation in terms of mode and frequency content is critical to interpreting the arrival structure. At teleseismic ranges, the dependence of group delays on mode number and frequency leads to significant time spreading of the various signal components.
Progress in the implementation of the T-phase subnetwork for monitoring the oceans
Authors: Arranz Marta Galindo
Published: July 2002

For the first time seismic technology is being used for monitoring the oceans, detecting and evaluating acoustic signals. Five seismic stations, called T-phase stations, are being established to complement six hydrophone stations, thus constituting the hydroacoustic international monitoring network. This T-phase subnetwork is mostly localized in the northern hemisphere and has been designed to work in synergy with the seismic network. Many technical difficulties arise at the time of installing new T-phase stations, due to lack of some relevant scientific knowledge. The T-phase stations are located on small islands with abrupt bathymetric properties and an open view of the ocean. Within this environment it is necessary to find the optimum distance from the coastline in order to achieve a compromise between sea-induced noise and good T-phase detection capability. Other open questions include the details of how acoustic energy is coupled into seismic energy, as well as how factors such as location and seasonal variations of the SOFAR channel affect the acoustic-to-seismic coupling. These questions are investigated at each site survey, upgrading, or new installation within the subnetwork. Results of these investigations will be presented as well as possible implications for technical specifications of the T-phase stations.
Perturbative eigenvalue techniques for global-scale hydroacoustic pulse propagation
Authors: Orris Gregory J. , Perkins John S. , Fialkowski Laurie T.
Published: July 2002

Because of the computational difficulties associated with three-dimensional acoustic propagation on global scales in the ocean, current models require a hybrid approach. In the Global-Scale Ocean Acoustic Raytrace Model, horizontal refraction of eigenrays is accomplished by first separating the vertical dependence of the field from the horizontal dependence and solving for the vertical eigenvalues and eigenmodes. This approximation is valid as long as the ocean medium is adiabatic in the horizontal direction. The eigenvalues and eigenmodes directly affect the local phase speed and group velocity, and consequently, refraction and travel time of a given ray. If Fourier synthesis is used in conjunction with another full-field method to obtain time-domain results along a horizontally refracted ray, then the computing requirements of determining the local eigenvalue spectrum as a function of frequency can become overly burdensome. An approach based on perturbation methods is introduced that can be used to compute the eigenvalue spectrum quickly at frequencies close to those at which the spectrum has already been determined, yet far enough away in frequency to be useful for the Fourier synthesis of pulse propagation. [Work supported by the U.S. Department of Energy, Office of Non-proliferation and National Security, Office of Research and Development, Contract No. DE-A101-95IS20011.]
Summary of bottom reverberation findings and model/data comparisons during three rapid environmental assessment trials
Authors: Preston John R. , Ellis Dale D.
Published: July 2002

From August 1996 through February 1998 the authors participated with SACLANTCEN in three rapid environmental assessment (REA) trials, including experiments for bottom reverberation. Eight sites were visited: three sites south of Sicily, three sites in the Ionian Sea, and two sites in the Atlantic near the Straits of Gibraltar. SUS charges were used as sources. The receivers were a nested horizontal array with 128 elements spaced at 0.5, 1, and 2 m and a second horizontal array with 32 elements spaced at 0.18 m. Sets of nearly monostatic recordings were analyzed in frequency bands from 80 to 4000 Hz. Polar plots of the beam time series were superimposed on bathymetric charts, revealing a number of scattering features not on the charts. The results illustrate the use of directional reverberation measurements as a useful remote-sensing tool. Another objective of the reverberation experiments was to quickly invert for estimates of scattering and geo-acoustic parameters that can be used in a wider variety of conditions. The horizontal array data were compared with the generic sonar model (GSM) predictions for selected cases. A summary of the model parameters found at the eight sites is presented. [Work partially supported by ONR.]
Shallow-water REA using autonomous ocean sampling networks
Authors: Schmidt Henrik , Bellingham James G. , Robinson Allan
Published: July 2002

Environmental management and naval operations in coastal waters are severely limited by a lack of oceanographic predictive capability associated with the strong spatial gradients and temporal variability inherent to such environments. A key to improving the predictive capability is a measurement system which can provide data in close to real time for assimilation into oceanographic circulation models. In that regard, the Autonomous Ocean Sampling Network (AOSN) combines small, autonomous underwater vehicles (AUV) with an underwater communication and navigation network to provide an ocean measurement system with real-time data access and adaptive sampling capabilities. The acoustic communication capability of the AOSN allows for feedback to the sensor platforms of uncertainties of the field estimates, information which can then be used to redistribute the AOSN resources optimally. The development of a new observation and prediction capability combining the AOSN with efficient coastal circulation models is the objective of an ongoing multidisciplinary research effort, but the feasibility has already been demonstrated through a series of field deployments. The results of these experiments are reviewed, and the potential for this new rapid environmental assessment capability for littoral environments is discussed.