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Journal of the Acoustical Society of America

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May 2012

Volume 131, Issue 5, pp. EL355-4232

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Flow noise of an underwater vector sensor embedded in a flexible towed array

Vladimir I. Korenbaum and Alexander A. Tagiltsev

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3755-3762 (2012); (8 pages)

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The objective of this work is to simulate the flow noise of a vector sensor embedded in a flexible towed array. The mathematical model developed, based on long-wavelength analysis of the inner space of a cylindrical multipole source, predicts the reduction of the flow noise of a vector sensor embedded in an underwater flexible towed array by means of intensimetric processing (cross-spectral density calculation of oscillatory velocity and sound-pressure-sensor responses). It is found experimentally that intensimetric processing results in flow noise reduction by 12–25 dB at mean levels and by 10–30 dB in fluctuations compared to a squared oscillatory velocity channel. The effect of flow noise suppression in the intensimetry channel relative to a squared sound pressure channel is observed, but only for frequencies above the threshold. These suppression values are 10–15 dB at mean noise levels and 3–6 dB in fluctuations. At towing velocities of 1.5–3 ms−1 and an accumulation time of 98.3 s, the threshold frequency in fluctuations is between 30 and 45 Hz.
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43.30.Yj Transducers and transducer arrays for underwater sound; transducer calibration
43.60.Uv Model-based signal processing
43.28.Ra Generation of sound by fluid flow, aerodynamic sound and turbulence
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Experimental evaluation of time domain models for ultrasound attenuation losses in photoacoustic imaging

H. Roitner, J. Bauer-Marschallinger, T. Berer, and P. Burgholzer

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3763-3774 (2012); (12 pages) | Cited 2 times

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Understanding and compensating ultrasound attenuation losses is an important issue in photoacoustic imaging. To contribute to this effort, simulated attenuated time domain waveforms are compared to experimental waveforms. The experimental waveforms are acquired by transmitting broadband ultrasound pulses through distilled water and porcine fat tissue. Three well-known modeling approaches are examined in detail with regard to accuracy and computation time. Furthermore, the influence of attenuation on imaging resolution is addressed. In the present paper, the focus lies on the calculation of attenuated detector signals. The results, however, also provide clues about the quality of image reconstruction.
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43.35.Bf Ultrasonic velocity, dispersion, scattering, diffraction, and attenuation in liquids, liquid crystals, suspensions, and emulsions
43.20.Hq Velocity and attenuation of acoustic waves
43.20.Bi Mathematical theory of wave propagation
43.35.Wa Biological effects of ultrasound, ultrasonic tomography

Sensitivity of the resonant ultrasound spectroscopy to weak gradients of elastic properties

Hanuš Seiner, Petr Sedlák, Lucie Bodnárová, Alena Kruisová, Michal Landa, Angel de Pablos, and Manuel Belmonte

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3775-3785 (2012); (11 pages) | Cited 2 times

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The applicability of resonant ultrasound spectroscopy on materials with weak spatial gradients in elastic coefficients and density is analyzed. It is shown that such gradients do not affect measurably the resonant spectrum but have a significant impact on the modal shapes. A numerical inverse procedure is proposed to explore the possibility of reconstructing the gradients from experimentally obtained modal shapes. This procedure is tested on synthetic data and applied to determine the gradient of the shear modulus in a continuously graded silicon nitride ceramic material. The results are in a good agreement with the gradient calculated for the examined material theoretically as well as with the results of other experimental methods.
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43.35.Cg Ultrasonic velocity, dispersion, scattering, diffraction, and attenuation in solids; elastic constants
43.40.At Experimental and theoretical studies of vibrating systems

Sound field reconstruction using acousto-optic tomography

Antoni Torras-Rosell, Salvador Barrera-Figueroa, and Finn Jacobsen

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3786-3793 (2012); (8 pages) | Cited 1 time

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When sound propagates through a medium, it results in pressure fluctuations that change the instantaneous density of the medium. Under such circumstances, the refractive index that characterizes the propagation of light is not constant, but influenced by the acoustic field. This kind of interaction is known as the acousto-optic effect. The formulation of this physical phenomenon into a mathematical problem can be described in terms of the Radon transform, which makes it possible to reconstruct an arbitrary sound field using tomography. The present work derives the fundamental equations governing the acousto-optic effect in air, and demonstrates that it can be measured with a laser Doppler vibrometer in the audible frequency range. The tomographic reconstruction is tested by means of computer simulations and measurements. The main features observed in the simulations are also recognized in the experimental results. The effectiveness of the tomographic reconstruction is further confirmed with representations of the very same sound field measured with a traditional microphone array.
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43.35.Sx Acoustooptical effects, optoacoustics, acoustical visualization, acoustical microscopy, and acoustical holography

On the Coriolis effect in acoustic waveguides

Henry Wegert, Leonard M. Reindl, Werner Ruile, and Andreas P. Mayer

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3794-3801 (2012); (8 pages)

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Rotation of an elastic medium gives rise to a shift of frequency of its acoustic modes, i.e., the time-period vibrations that exist in it. This frequency shift is investigated by applying perturbation theory in the regime of small ratios of the rotation velocity and the frequency of the acoustic mode. In an expansion of the relative frequency shift in powers of this ratio, upper bounds are derived for the first-order and the second-order terms. The derivation of the theoretical upper bounds of the first-order term is presented for linear vibration modes as well as for stable nonlinear vibrations with periodic time dependence that can be represented by a Fourier series.
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43.35.Ty Other physical effects of sound
43.20.Bi Mathematical theory of wave propagation
43.40.At Experimental and theoretical studies of vibrating systems

Non-linear inverse scattering: High resolution quantitative breast tissue tomography

J. Wiskin, D. T. Borup, S. A. Johnson, and M. Berggren

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3802-3813 (2012); (12 pages) | Cited 1 time

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Recent published results in inverse scattering generally show the difficulty in dealing with moderate to high contrast inhomogeneities when employing linearized or iteratively linearized algorithms (e.g., distorted Born iterative method). This paper presents a fully nonlinear algorithm utilizing full wave field data, that results in ultrasound computed tomographic images from a laboratory breast scanner, and shows several such unique images from volunteer subjects. The forward problem, data collection process and inverse scattering algorithm used are discussed. A functional that represents the “best fit” between predicted and measured data is minimized, and therefore requires a very fast forward problem solver, Jacobian calculation, and gradient estimation, all of which are described. The data collection device is described. The algorithm and device yield quantitative estimates of human breast tissue in vivo. Several high resolution images, measuring ∼150 by 150 wavelengths, obtained from the 2D inverse scattering algorithms, using data collected from a first prototype, are shown and discussed. The quantitative values are compared with previous published work.
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43.35.Wa Biological effects of ultrasound, ultrasonic tomography
43.60.Lq Acoustic imaging, displays, pattern recognition, feature extraction
43.60.Rw Remote sensing methods, acoustic tomography
43.80.Vj Acoustical medical instrumentation and measurement techniques
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Interior and exterior sound field control using two dimensional higher-order variable-directivity sources

M. A. Poletti, T. D. Abhayapala, and P. Samarasinghe

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3814-3823 (2012); (10 pages) | Cited 1 time

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Spatial sound reproduction systems aim to produce a desired sound field over a volume of space. At high frequencies, the number of loudspeakers required is prohibitive. This paper shows that the use of loudspeakers with up to Nth order directivity allows reproduction over N times the bandwidth and produces a significantly attenuated exterior sound field. If the constraint on exterior cancellation of the field is removed, reproduction is possible over approximately 2N times the bandwidth. The use of higher order loudspeakers thus allows a significant reduction in the number of loudspeaker units, at the expense of increased complexity in each unit. For completeness, results are included for the generation of an exterior field with or without cancellation of the interior field.
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43.38.Md Sound recording and reproducing systems, general concepts
43.60.Sx Acoustic holography
43.55.Jz Sound-reinforcement systems for rooms and enclosures
43.60.Tj Wave front reconstruction, acoustic time-reversal, and phase conjugation
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Impedance of infinite Kirchhoff and Mindlin plates with a rigid circular massless plug

Jeffrey A. Zapfe and James A. Moore

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3824-3832 (2012); (9 pages)

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Point force impedance expressions have been previously developed for infinite Kirchhoff and Mindlin plates. The present work develops impedance expressions for the more general case of an infinite plate with a circular, massless, rigid plug using both Kirchhoff and Mindlin plate theories. The models have been developed to analyze vibration propagation in buildings. The plate with the rigid plug provides a more reasonable model of the kinematic constraint at the column/floor interface. The models are used to investigate the potential benefits of using thick floors to block the transmission of structure-borne vibration in buildings.
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43.40.Dx Vibrations of membranes and plates
43.40.At Experimental and theoretical studies of vibrating systems

Development of a pseudo-uniform structural quantity for use in active structural acoustic control of simply supported plates: An analytical comparison

Jeffery M. Fisher, Jonathan D. Blotter, Scott D. Sommerfeldt, and Kent L. Gee

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3833-3840 (2012); (8 pages)

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Active structural acoustic control has been an area of research and development for over two decades with an interest in searching for an “optimal” error quantity. Current error quantities typically require the use of either a large number of transducers distributed across the entire structure, or a distributed shaped sensor, such as polyvinylidene difluoride. The purpose of this paper is to investigate a control objective function for flat, simply-supported plates that is based on transverse and angular velocity components combined into a single composite structural velocity quantity, termed Vcomp. Although multiple transducers are used, they are concentrated at a single location to eliminate the need for transducers spanning most or all of the structure. When used as the objective function in an active control situation, squared Vcomp attenuates the acoustic radiation over a large range of frequencies. The control of squared Vcomp is compared to other objective functions including squared velocity, volume velocity, and acoustic energy density. The analysis presented indicates that benefits of this objective function include control of radiation from numerous structural modes, control largely independent of sensor location, and need to measure Vcomp at a single location and not distributed measurements across the entire structure.
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43.40.Yq Instrumentation and techniques for tests and measurement relating to shock and vibration, including vibration pickups, indicators, and generators, mechanical impedance
43.50.Ki Active noise control
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A mode matching approach for modeling two dimensional porous grating with infinitely rigid or soft inclusions

Benoit Nennig, Ygaäl Renou, Jean-Philippe Groby, and Yves Aurégan

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3841-3852 (2012); (12 pages) | Cited 1 time

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This work investigates the acoustical properties of a multilayer porous material in which periodic inclusions are embedded. The material is assumed to be backed by a rigid wall. Most of the studies performed in this field used the multipole method and are limited to circular shape inclusions. Here, a mode matching approach, more convenient for a layered system, is adopted. The inclusions can be in the form of rigid scatterers of an arbitrary shape, in the form of an air-filled cavity or in the form of a porous medium with contrasting properties. The computational approach is validated on simple geometries against other numerical schemes and with experimental results obtained in an anechoic room on a rigid grating embedded in a porous material made of 2 mm glass beads. The method is used to study the acoustic absorption behavior of this class of materials in the low frequency range and at a range of angles of incidence.
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43.55.Ev Sound absorption properties of materials: theory and measurement of sound absorption coefficients; acoustic impedance and admittance
43.20.Fn Scattering of acoustic waves
43.20.Ks Standing waves, resonance, normal modes
43.20.Gp Reflection, refraction, diffraction, interference, and scattering of elastic and poroelastic waves

Sound absorption and transmission through flexible micro-perforated panels backed by an air layer and a thin plate

Teresa Bravo, Cédric Maury, and Cédric Pinhède

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3853-3863 (2012); (11 pages)

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This paper describes theoretical and experimental investigations into the sound absorption and transmission properties of micro-perforated panels (MPP) backed by an air cavity and a thin plate. A fully coupled modal approach is proposed to calculate the absorption coefficient and the transmission loss of finite-sized micro-perforated panels-cavity-panel (MPPCP) partitions with conservative boundary conditions. It is validated against infinite partition models and experimental data. A practical methodology is proposed using collocated pressure-velocity sensors to evaluate in an anechoic environment the transmission and absorption properties of conventional MPPCPs. Results show under which conditions edge scattering effects should be accounted for at low frequencies. Coupled mode analysis is also performed and analytical approximations are derived from the resonance frequencies and mode shapes of a flexible MPPCP. It is found that the Helmholtz-type resonance frequency is deduced from the one associated to the rigidly backed MPPCP absorber shifted up by the mass-air mass resonance of the flexible non-perforated double-panel. Moreover, it is shown analytically and experimentally that the absorption mechanisms at the resonances are governed by a large air-frame relative velocity over the MPP surface, with either in-phase or out-of-phase relationships, depending on the MPPCP parameters.
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43.55.Ev Sound absorption properties of materials: theory and measurement of sound absorption coefficients; acoustic impedance and admittance
43.40.At Experimental and theoretical studies of vibrating systems
43.55.Rg Sound transmission through walls and through ducts: theory and measurement
43.50.Gf Noise control at source: redesign, application of absorptive materials and reactive elements, mufflers, noise silencers, noise barriers, and attenuators, etc.

Effects of source and receiver locations in predicting room transfer functions by a phased beam tracing method

Cheol-Ho Jeong and Jeong-Guon Ih

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3864-3875 (2012); (12 pages) | Cited 1 time

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The accuracy of a phased beam tracing method in predicting transfer functions is investigated with a special focus on the positions of the source and receiver. Simulated transfer functions for various source-receiver pairs using the phased beam tracing method were compared with analytical Green’s functions and boundary element solutions up to the Schroeder frequency in simple rectangular rooms with different aspect ratios and absorptions. Only specular reflections were assumed and diffraction was neglected. Three types of error definitions were used: average error level over a narrow band spectrum, average error level over a 1/3 octave band spectrum, and dissimilarity measure. The narrow band error and dissimilarity increased with the source-to-receiver distance but converged to a certain value as the reverberant field became dominant. The 1/3 octave band error was found to be less dependent on the source-receiver distance. The errors are increased as the aspect ratio becomes more disproportionate. By changing the wall absorption from 0.2 to 0.8 for a rectangular room, the average narrow and 1/3 octave band error are deviated by around 1.5 dB. A realistic non-uniform distribution of the absorption increases the error, which might be ascribed to wave phenomena evoked by the impedance-discontinuous boundary.
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43.55.Ka Computer simulation of acoustics in enclosures, modeling

Numerical analysis of eigenproblem for cavities by a particular integral method with a low frequency approximation of surface admittance

Alexandre Leblanc and Antoine Lavie

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3876-3882 (2012); (7 pages)

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In this paper, a three-dimensional boundary element method for the eigenanalysis of complex-shaped cavity is presented. A particular integral method is proposed with general absorbing boundary conditions, well suited for determination of the lower modes. In this approach, a polynomial approximation of surface admittance is used with a recent class of compactly supported radial basis function. Two common absorbent models are employed in order to demonstrate the relevance of high-order approximation of the admittance. Resulting eigenproblems of several orders (linear to cubic) are thus performed on basic geometries and a car interior. Results show significant improvements for the computed damped eigenfrequencies and the associated modal reverberation time while using an approximation polynomial matching the surface admittance variation order.
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43.55.Ka Computer simulation of acoustics in enclosures, modeling
43.55.Br Room acoustics: theory and experiment; reverberation, normal modes, diffusion, transient and steady-state response
43.20.Ks Standing waves, resonance, normal modes
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Acoustic particle velocity horns

Dimitri M. Donskoy and Benjamin A. Cray

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3883-3890 (2012); (8 pages)

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The paper considers receiving acoustic horns designed for particle velocity amplification and suitable for use in vector sensing applications. Unlike conventional horns, designed for acoustic pressure amplification, acoustic velocity horns (AVHs) deliver significant velocity amplification even when the overall size of the horn is much less than an acoustic wavelength. An AVH requires an open-ended configuration, as compared to pressure horns which are terminated at the throat. The appropriate formulation, based on Webster’s one-dimensional horn equation, is derived and analyzed for single conical and exponential horns as well as for double-horn configurations. Predicted horn amplification factors (ratio of mouth-to-throat radii) were verified using numerical modeling. It is shown that three independent geometrical parameters principally control a horn’s performance: length l, throat radius R1, and flare rate. Below a predicted resonance region, velocity amplification is practically independent of frequency. Acoustic velocity horns are naturally directional, providing maximum velocity amplification along the boresight.
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43.58.Fm Sound level meters, level recorders, sound pressure, particle velocity, and sound intensity measurements, meters, and controllers
43.20.Mv Waveguides, wave propagation in tubes and ducts
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A directionally tunable but frequency-invariant beamformer on an acoustic velocity-sensor triad to enhance speech perception

Yue Ivan Wu, Kainam Thomas Wong, Xin Yuan, Siu-kit Lau, and Shiu-keung Tang

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3891-3902 (2012); (12 pages) | Cited 2 times

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Herein investigated are computationally simple microphone-array beamformers that are independent of the frequency-spectra of all signals, all interference, and all noises. These beamformers allow the listener to tune the desired azimuth-elevation “look direction.” No prior information is needed of the interference. These beamformers deploy a physically compact triad of three collocated but orthogonally oriented velocity sensors. These proposed schemes’ efficacy is verified by a jury test, using simulated data constructed with Mandarin Chinese (a.k.a. Putonghua) speech samples. For example, a desired speech signal, originally at a very adverse signal-to-interference-and-noise power ratio (SINR) of −30 dB, may be processed to become fully intelligible to the jury.
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43.60.Fg Acoustic array systems and processing, beam-forming
43.60.Dh Signal processing for communications: telephony and telemetry, sound pickup and reproduction, multimedia
43.60.Gk Space-time signal processing, other than matched field processing
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Modeling auditory evoked brainstem responses to transient stimuli

Filip Munch Rønne, Torsten Dau, James Harte, and Claus Elberling

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3903-3913 (2012); (11 pages) | Cited 1 time

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A quantitative model is presented that describes the formation of auditory brainstem responses (ABRs) to tone pulses, clicks, and rising chirps as a function of stimulation level. The model computes the convolution of the instantaneous discharge rates using the “humanized” nonlinear auditory-nerve model of Zilany and Bruce [J. Acoust. Soc. Am. 122, 402–417 (2007)] and an empirically derived unitary response function which is assumed to reflect contributions from different cell populations within the auditory brainstem, recorded at a given pair of electrodes on the scalp. It is shown that the model accounts for the decrease of tone-pulse evoked wave-V latency with frequency but underestimates the level dependency of the tone-pulse as well as click-evoked latency values. Furthermore, the model correctly predicts the nonlinear wave-V amplitude behavior in response to the chirp stimulation both as a function of chirp sweeping rate and level. Overall, the results support the hypothesis that the pattern of ABR generation is strongly affected by the nonlinear and dispersive processes in the cochlea.
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43.64.Bt Models and theories of the auditory system
43.64.Qh Electrophysiology of the auditory central nervous system
43.64.Ri Evoked responses to sounds

Transmission of cochlear distortion products as slow waves: A comparison of experimental and model data

Aleš Vetešník and Anthony W. Gummer

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3914-3934 (2012); (21 pages) | Cited 1 time

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There is a long-lasting question of how distortion products (DPs) arising from nonlinear amplification processes in the cochlea are transmitted from their generation sites to the stapes. Two hypotheses have been proposed: (1) the slow-wave hypothesis whereby transmission is via the transverse pressure difference across the cochlear partition and (2) the fast-wave hypothesis proposing transmission via longitudinal compression waves. Ren with co-workers have addressed this topic experimentally by measuring the spatial vibration pattern of the basilar membrane (BM) in response to two tones of frequency f1 and f2. They interpreted the observed negative phase slopes of the stationary BM vibrations at the cubic distortion frequency fDP = 2f1 − f2 as evidence for the fast-wave hypothesis. Here, using a physically based model, it is shown that their phase data is actually in accordance with the slow-wave hypothesis. The analysis is based on a frequency-domain formulation of the two-dimensional motion equation of a nonlinear hydrodynamic cochlea model. Application of the analysis to their experimental data suggests that the measurement sites of negative phase slope were located at or apical to the DP generation sites. Therefore, current experimental and theoretical evidence supports the slow-wave hypothesis. Nevertheless, the analysis does not allow rejection of the fast-wave hypothesis.
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43.64.Jb Otoacoustic emissions
43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system

Unification and extension of monolithic state space and iterative cochlear models

Michael J. Rapson, Jonathan C. Tapson, and David Karpul

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3935-3952 (2012); (18 pages)

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Time domain cochlear models have primarily followed a method introduced by Allen and Sondhi [J. Acoust. Soc. Am. 66, 123–132 (1979)]. Recently the “state space formalism” proposed by Elliott et al. [J. Acoust. Soc. Am. 122, 2759–2771 (2007)] has been used to simulate a wide range of nonlinear cochlear models. It used a one-dimensional approach that is extended to two dimensions in this paper, using the finite element method. The recently developed “state space formalism” in fact shares a close relationship to the earlier approach. Working from Diependaal et al. [J. Acoust. Soc. Am. 82, 1655–1666 (1987)] the two approaches are compared and the relationship formalized. Understanding this relationship allows models to be converted from one to the other in order to utilize each of their strengths. A second method to derive the state space matrices required for the “state space formalism” is also presented. This method offers improved numerical properties because it uses the information available about the model more effectively. Numerical results support the claims regarding fluid dimension and the underlying similarity of the two approaches. Finally, the recent advances in the state space formalism [Bertaccini and Sisto, J. Comp. Phys. 230, 2575–2587 (2011)] are discussed in terms of this relationship.
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43.64.Kc Cochlear mechanics
43.64.Bt Models and theories of the auditory system
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Sparse regularized regression identifies behaviorally-relevant stimulus features from psychophysical data

Vinzenz H. Schönfelder and Felix A. Wichmann

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3953-3969 (2012); (17 pages)

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As a prerequisite to quantitative psychophysical models of sensory processing it is necessary to learn to what extent decisions in behavioral tasks depend on specific stimulus features, the perceptual cues. Based on relative linear combination weights, this study demonstrates how stimulus-response data can be analyzed in this regard relying on an L1-regularized multiple logistic regression, a modern statistical procedure developed in machine learning. This method prevents complex models from over-fitting to noisy data. In addition, it enforces “sparse” solutions, a computational approximation to the postulate that a good model should contain the minimal set of predictors necessary to explain the data. In simulations, behavioral data from a classical auditory tone-in-noise detection task were generated. The proposed method is shown to precisely identify observer cues from a large set of covarying, interdependent stimulus features—a setting where standard correlational and regression methods fail. The proposed method succeeds for a wide range of signal-to-noise ratios and for deterministic as well as probabilistic observers. Furthermore, the detailed decision rules of the simulated observers were reconstructed from the estimated linear model weights allowing predictions of responses on the basis of individual stimuli.
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43.66.Ba Models and theories of auditory processes
43.66.Dc Masking

Comparing models of the combined-stimulation advantage for speech recognition

Christophe Micheyl and Andrew J. Oxenham

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3970-3980 (2012); (11 pages) | Cited 1 time

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The “combined-stimulation advantage” refers to an improvement in speech recognition when cochlear-implant or vocoded stimulation is supplemented by low-frequency acoustic information. Previous studies have been interpreted as evidence for “super-additive” or “synergistic” effects in the combination of low-frequency and electric or vocoded speech information by human listeners. However, this conclusion was based on predictions of performance obtained using a suboptimal high-threshold model of information combination. The present study shows that a different model, based on Gaussian signal detection theory, can predict surprisingly large combined-stimulation advantages, even when performance with either information source alone is close to chance, without involving any synergistic interaction. A reanalysis of published data using this model reveals that previous results, which have been interpreted as evidence for super-additive effects in perception of combined speech stimuli, are actually consistent with a more parsimonious explanation, according to which the combined-stimulation advantage reflects an optimal combination of two independent sources of information. The present results do not rule out the possible existence of synergistic effects in combined stimulation; however, they emphasize the possibility that the combined-stimulation advantages observed in some studies can be explained simply by non-interactive combination of two information sources.
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43.66.Ba Models and theories of auditory processes
43.66.Ts Auditory prostheses, hearing aids

Binaural loudness summation for speech presented via earphones and loudspeaker with and without visual cues

Michael Epstein and Mary Florentine

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3981-3988 (2012); (8 pages)

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Preliminary data [M. Epstein and M. Florentine, Ear. Hear. 30, 234–237 (2009)] obtained using speech stimuli from a visually present talker heard via loudspeakers in a sound-attenuating chamber indicate little difference in loudness when listening with one or two ears (i.e., significantly reduced binaural loudness summation, BLS), which is known as “binaural loudness constancy.” These data challenge current understanding drawn from laboratory measurements that indicate a tone presented binaurally is louder than the same tone presented monaurally. Twelve normal listeners were presented recorded spondees, monaurally and binaurally across a wide range of levels via earphones and a loudspeaker with and without visual cues. Statistical analyses of binaural-to-monaural ratios of magnitude estimates indicate that the amount of BLS is significantly less for speech presented via a loudspeaker with visual cues than for stimuli with any other combination of test parameters (i.e., speech presented via earphones or a loudspeaker without visual cues, and speech presented via earphones with visual cues). These results indicate that the loudness of a visually present talker in daily environments is little affected by switching between binaural and monaural listening. This supports the phenomenon of binaural loudness constancy and underscores the importance of ecological validity in loudness research.
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43.66.Cb Loudness, absolute threshold
43.66.Pn Binaural hearing
43.66.Ba Models and theories of auditory processes
43.66.Lj Perceptual effects of sound

Further evidence that fundamental-frequency difference limens measure pitch discrimination

Christophe Micheyl, Claire M. Ryan, and Andrew J. Oxenham

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 3989-4001 (2012); (13 pages) | Cited 2 times

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Difference limens for complex tones (DLCs) that differ in F0 are widely regarded as a measure of periodicity-pitch discrimination. However, because F0 changes are inevitably accompanied by changes in the frequencies of the harmonics, DLCs may actually reflect the discriminability of individual components. To test this hypothesis, DLCs were measured for complex tones, the component frequencies of which were shifted coherently upward or downward by ΔF = 0%, 25%, 37.5%, or 50% of the F0, yielding fully harmonic (ΔF = 0%), strongly inharmonic (ΔF = 25%, 37.5%), or odd-harmonic (ΔF = 50%) tones. If DLCs truly reflect periodicity-pitch discriminability, they should be larger (worse) for inharmonic tones than for harmonic and odd harmonic tones because inharmonic tones have a weaker pitch. Consistent with this prediction, the results of two experiments showed a non-monotonic dependence of DLCs on ΔF, with larger DLCs for ΔF’s of ±25% or ±37.5% than for ΔF’s of 0 or ±50% of F0. These findings are consistent with models of pitch perception that involve harmonic templates or with an autocorrelation-based model provided that more than just the highest peak in the summary autocorrelogram is taken into account.
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43.66.Hg Pitch
43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes

Identification of walked-upon materials in auditory, kinesthetic, haptic, and audio-haptic conditions

Bruno L. Giordano, Yon Visell, Hsin-Yun Yao, Vincent Hayward, Jeremy R. Cooperstock, and Stephen McAdams

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 4002-4012 (2012); (11 pages) | Cited 1 time

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Locomotion generates multisensory information about walked-upon objects. How perceptual systems use such information to get to know the environment remains unexplored. The ability to identify solid (e.g., marble) and aggregate (e.g., gravel) walked-upon materials was investigated in auditory, haptic or audio-haptic conditions, and in a kinesthetic condition where tactile information was perturbed with a vibromechanical noise. Overall, identification performance was better than chance in all experimental conditions and for both solids and the better identified aggregates. Despite large mechanical differences between the response of solids and aggregates to locomotion, for both material categories discrimination was at its worst in the auditory and kinesthetic conditions and at its best in the haptic and audio-haptic conditions. An analysis of the dominance of sensory information in the audio-haptic context supported a focus on the most accurate modality, haptics, but only for the identification of solid materials. When identifying aggregates, response biases appeared to produce a focus on the least accurate modality—kinesthesia. When walking on loose materials such as gravel, individuals do not perceive surfaces by focusing on the most accurate modality, but by focusing on the modality that would most promptly signal postural instabilities.
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43.66.Lj Perceptual effects of sound
43.66.Wv Vibration and tactile senses
43.66.Jh Timbre, timbre in musical acoustics
43.66.Ba Models and theories of auditory processes

Temporal predictions based on a gradual change in tempo

Thomas E. Cope, Manon Grube, and Timothy D. Griffiths

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 4013-4022 (2012); (10 pages) | Cited 1 time

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Previous studies investigating sensitivity to step changes in tempo and prediction of tone onset time have generally utilized isochronous sequences. This study investigates subjects’ ability to detect deviations from a gradual change in the tempo of a tone sequence (experiment 1) and their judgment of the perceptually optimal timing of this tone (experiment 2). In experiment 1, inter-onset-intervals within pairs of eight-tone sequences followed a geometric progression to create a gradual tempo change. In one sequence, the final tone was presented either earlier or later than specified by the progression. Subjects performed well at detecting deviations that exaggerated the tempo progression but poorly when it was counteracted. Experiment 2 used similar pairs except that the final tone was always presented earlier in one sequence than the other. Final interval length was adaptively adjusted to subjects’ judgments; it was adjudged in best agreement with the progression when its length was roughly half way between the mathematically correct value and the length of the penultimate interval. The data support “multiple-look” and entrainment models of tempo sensitivity and suggest that temporal prediction is based less on the tempo contour of a whole sequence than on the duration of the preceding interval.
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43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.75.Cd Music perception and cognition

The three-channel model of sound localization mechanisms: Interaural level differences

Rachel N. Dingle, Susan E. Hall, and Dennis P. Phillips

J. Acoust. Soc. Am. Volume 131, Issue 5, pp. 4023-4029 (2012); (7 pages) | Cited 2 times

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The current understanding of mammalian sound localization is that azimuthal (horizontal) position assignments are dependent upon the relative activation of two populations of broadly-tuned hemifield neurons with overlapping medial borders. Recent psychophysical work has provided evidence for a third channel of low-frequency interaural time difference (ITD)-sensitive neurons tuned to the azimuthal midline. However, the neurophysiological data on free-field azimuth receptive fields, especially of cortical neurons, has primarily studied high-frequency cells whose receptive fields are more likely to have been shaped by interaural level differences (ILDs) than ITDs. In four experiments, a selective adaptation paradigm was used to probe for the existence of a midline channel in the domain of ILDs. If no midline channel exists, symmetrical adaptation of the lateral channels should not result in a shift in the perceived intracranial location of subsequent test tones away from the adaptors because the relative activation of the two channels will remain unchanged. Instead, results indicate a shift in perceived test tone location away from the adaptors, which supports the existence of a midline channel in the domain of ILDs. Interestingly, this shift occurs not only at high frequencies, traditionally associated with ILDs in natural settings, but at low frequencies as well.
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43.66.Qp Localization of sound sources
43.66.Pn Binaural hearing
43.66.Ba Models and theories of auditory processes
43.66.Ed Auditory fatigue, temporary threshold shift
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