The acoustic reflex threshold in aging ears

This study investigates the controversy regarding the influence of age on the acoustic reflex threshold for broadband noise, 500-, 1000-, 2000-, and 4000-Hz activators between Jerger et al. [Mono. Contemp. Audiol. 1 (1978)] and Jerger [J. Acoust. Soc. Am. 66 (1979)] on the one hand and Silman [J. Acoust. Soc. Am. 66 (1979)] and others on the other. The acoustic reflex thresholds for broadband noise, 500-, 1000-, 2000-, and 4000-Hz activators were evaluated under two measurement conditions. Seventy-two normal-hearing ears were drawn from 72 subjects ranging in age from 20-69 years. The results revealed that age was correlated with the acoustic reflex threshold for BBN activator but not for any of the tonal activators; the correlation was stronger under the 1-dB than under the 5-dB measurement condition. Also, the mean acoustic reflex thresholds for broadband noise activator were essentially similar to those reported by Jerger et al. (1978) but differed from those obtained in this study under the 1-dB measurement condition.     

A method to account for acoustic microstreaming when predicting bubble growth rates produced by rectified diffusion

A reinterpretation of existing theory for rectified diffusion, the process by which bubbles in a sound field may grow in radius, is presented in order to quantitate the effect of acoustic microstreaming on bubble growth rates. The 1/t term in the growth rate equation is defined as the "decay term" and t as the "decay time," the time required for the gas concentration in the liquid contacting the bubble to rise (or fall) from its initial to its final value. In the absence of microstreaming, t is the duration of sonification. In the presence of microstreaming, t may be calculated from the streaming velocity and the bubble radius. A comparison between theory and the experimental results of Eller [A. Eller, J. Acoust. Soc. Am. 46, 1246-1250 (1969)] and of Gould [R.K. Gould, J. Acoust. Soc. Am. 56, 1740-1746 (1974)] shows reasonable agreement in the low kHz range. Theoretical results in the frequency range of 1-10 MHz at 1 and 4 bar are also presented.     

A new formalism for time-dependent wave scattering from a bounded obstacle

A time-dependent three-dimensional acoustic scattering problem is considered. An incoming wave packet is scattered by a bounded, simply connected obstacle with locally Lipschitz boundary. The obstacle is assumed to have a constant boundary acoustic impedance. The limit cases of acoustically soft and acoustically hard obstacles are considered. The scattered acoustic field is the solution of an exterior problem for the wave equation. A new numerical method to compute the scattered acoustic field is proposed. This numerical method obtains the time-dependent scattered field as a superposition of time-harmonic acoustic waves and computes the time-harmonic acoustic waves by a new "operator expansion method." That is, the time-harmonic acoustic waves are solutions of an exterior boundary value problem for the Helmholtz equation. The method used to compute the time-harmonic waves improves on the method proposed by Misici, Pacelli, and Zirilli [J. Acoust. Soc. Am. 103, 106-113 (1998)] and is based on a "perturbative series" of the type of the one proposed in the operator expansion method by Milder [J. Acoust. Soc. Am. 89, 529-541 (1991)]. Computationally, the method is highly parallelizable with respect to time and space variables. Some numerical experiments on test problems obtained with a parallel implementation of the numerical method proposed are shown and discussed from the numerical and the physical point of view. The website: http://www.econ.unian.it/recchioni/w1 shows four animations relative to the numerical experiments.     

Optimal focusing by spatio-temporal inverse filter. I. Basic principles

A focusing technique based on the inversion of the propagation operator relating an array of transducers to a set of control points inside a medium was proposed in previous work [Tanter et al., J. Acoust. Soc. Am. 108, 223-234 (2000)] and is extended here to the time domain. As the inversion of the propagation operator is achieved both in space and time, this technique allows calculation of the set of temporal signals to be emitted by each element of the array in order to optimally focus on a chosen control point. This broadband inversion process takes advantage of the singular-value decomposition of the propagation operator in the Fourier domain. The physical meaning of this decomposition is explained in a homogeneous medium. In particular, a definition of the number of degrees of freedom necessary to define the acoustic field generated by an array of limited aperture in a focal plane of limited extent is given. This number corresponds to the number of independent signals that can be created in the focal area both in space and time. In this paper, this broadband inverse-focusing technique is compared in homogeneous media with the classical focusing achieved by simple geometrical considerations but also with time-reversal focusing. It is shown that, even in a simple medium, slight differences appear between these three focusing strategies. In the companion paper [Aubry et al., J. Acoust. Soc. Am. 110, 48-58 (2001)] the three focusing techniques are compared in heterogeneous, absorbing, or complex media where classical focusing is strongly degraded. The strong improvement achieved by the spatio-temporal inverse-filter technique emphasizes the great potential of multiple-channel systems having the ability to apply completely different signal waveforms on each transducer of the array. The application of this focusing technique could be of great interest in various ultrasonic fields such as medical imaging, nondestructive testing, and underwater acoustics.     

Simulating the response of multiple microphones to a single acoustic source in a reverberant room

A method is presented for simulating the impulse response between an acoustic source and multiple microphones in a reverberant room. The method is similar to the image method described by Allen and Berkley [J. Acoust. Soc. Am. 65, 943-950 (1979)] but includes modifications to simulate received echo arrival time accurately. The essential modification is to represent each received echo as a low-pass-filtered impulse at the correct arrival time. Using this "low-pass impulse" method, reverberant rooms can be simulated with sufficient accuracy to investigate multiple-microphone systems that are sensitive to interchannel phase.     

Analyzing reverse middle-ear transmission: noninvasive Gedankenexperiments.

The phenomenological framework outlined in the companion paper [C. A. Shera and G. Zweig, J. Acoust. Soc. Am. 92, 1356-1370 (1992)] characterizes both forward and reverse transmission through the middle ear. This paper illustrates its use in the analysis of noninvasive measurements of middle-ear and cochlear mechanics. A cochlear scattering framework is developed for the analysis of combination-tone and other experiments in which acoustic distortion products are used to drive the middle ear "in reverse." The framework is illustrated with a simple psychophysical Gedankenexperiment analogous to the neurophysiological experiments of P. F. Fahey and J. B. Allen [J. Acoust. Soc. Am. 77, 599-612 (1985)].     

Pitch and rhythm paradoxes: comments on "Auditory paradox based on fractal waveform"

Schroeder [J. Acoust. Soc. Am. 79, 186-189 (1986)] describes the paradox of acoustic waveforms that sound lower when reproduced at higher speeds. The author has also demonstrated this paradox [J.C. Risset, J. Acoust. Soc. Am. 46, 88 (A) (1969); see also Seventh ICA, Budapest, S10, 613-616 (1971)]; in addition he has recently synthesized a rhythmic analog of the paradox, namely rhythmical sequences that can sound slower when reproduced at higher speeds.     

Spectro-temporal modulation transfer functions and speech intelligibility

Detection thresholds for spectral and temporal modulations are measured using broadband spectra with sinusoidally rippled profiles that drift up or down the log-frequency axis at constant velocities. Spectro-temporal modulation transfer functions (MTFs) are derived as a function of ripple peak density (omega cycles/octave) and drifting velocity (omega Hz). The MTFs exhibit a low-pass function with respect to both dimensions, with 50% bandwidths of about 16 Hz and 2 cycles/octave. The data replicate (as special cases) previously measured purely temporal MTFs (omega = 0) [Viemeister, J. Acoust. Soc. Am. 66, 1364-1380 (1979)] and purely spectral MTFs (omega = 0) [Green, in Auditory Frequency Selectivity (Plenum, Cambridge, 1986), pp. 351-359]. A computational auditory model is presented that exhibits spectro-temporal MTFs consistent with the salient trends in the data. The model is used to demonstrate the potential relevance of these MTFs to the assessment of speech intelligibility in noise and reverberant conditions.     

Target detection and location with ambient noise

By placing a vertical array in an ambient noise field and forming an upward and a downward beam one obtains two time series which can be cross correlated to reveal a subbottom profile of the seabed [Siderius et al., J. Acoust. Soc. Am. 120, 1315-1323 (2006)]. Here the cross-correlation approach is applied to the location in range and bearing of a point target. An experiment was designed using floats and weights mounted (and dismounted) on the same cable as the vertical array. Careful measurements were made of the location of all likely floats, ballast weights, array terminations, and so on. After suitable coherent averaging, peaks were seen at delays (correlation offsets) agreeing with the reflector positions and were shown to be absent when reflectors were removed. A trivial extension of the theory developed in Harrison and Siderius [J. Acoust. Soc. Am. 123, 1282-1296 (2008)] is used to explain the rough amplitudes of the reflections. The approach differs from "acoustic daylight" principally in having a capability to determine a target range.     

Experimental validations of the HELS method for reconstructing acoustic radiation from a complex vibrating structure

This paper presents experimental validations of the Helmholtz Equation Least Squares (HELS) method [Wang and Wu, J. Acoust. Soc. Am. 102, 2020-2032 (1997); Wu and Wang, U.S. Patent Number 5712805 (1998); Wu, J. Acoust. Soc. Am. 107, 2511-2522 (2000)] on reconstruction of the radiated acoustic pressures from a complex vibrating structure. The structure under consideration has geometry and dimensions similar to those of a real passenger vehicle front end. To simulate noise radiation from a vehicle, a high fidelity loudspeaker installed inside the structure at the location of the engine is employed to generate both random and harmonic acoustic excitations. The radiated acoustic pressures are measured over a finite planar surface above the structure by a microphone. The measured data are taken as input to the HELS formulation to reconstruct the acoustic pressures on the top surface of the structure as well as in the field. The reconstructed acoustic pressures are then compared with measured ones at the same locations. Also shown are comparisons of the reconstructed and measured acoustic pressure spectra at various locations on the surface. Results show that satisfactory reconstruction can be obtained on the top surface of the structure subject to both random and harmonic excitations. Moreover, the more measurements and the closer their distances to the source surface, the more accurate the reconstruction. The efficiency of the HELS method may decrease with increasing of the excitation frequency. This high frequency difficulty is inherent in all expansion theories.     

Unification and extension of monolithic state space and iterative cochlear models

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.     

Illusory spectrotemporal ripples created with binaurally correlated noise

Binaural disparities are the primary acoustic cues employed in sound localization tasks. However, the degree of binaural correlation in a sound serves as a complementary cue for detecting competing sound sources [J. F. Culling, H. S. Colburn, and M. Spurchise, "Interaural correlation sensitivity," J. Acoust. Soc. Am. 110(2), 1020-1029 (2001) and L. R. Bernstein and C. Trahiotis, "On the use of the normalized correlation as an index of interaural envelope correlation," J. Acoust. Soc. Am. 100, 1754-1763 (1996)]. Here a random chord stereogram (RCS) sound is developed that produces a salient pop-out illusion of a slowly varying ripple sound [T. Chi et al., "Spectro-temporal modulation transfer functions and speech intelligibility," J. Acoust. Soc. Am. 106(5), 2719-2732 (1999)], even though the left and right ear sounds alone consist of noise-like random modulations. The quality and resolution of this percept is systematically controlled by adjusting the spectrotemporal correlation pattern between the left and right sounds. The prominence and limited time-frequency resolution for resolving the RCS suggests that envelope correlations are a dominant binaural cue for grouping acoustic objects.     

Acoustic radiation torque and the conservation of angular momentum (L)

This note concerns the evaluation of the static acoustic radiation torque exerted by an acoustic field on a scatterer immersed in a nonviscous fluid based on far-field scattering. The radiation torque is expressed as the integral of the time-averaged flux of angular momentum over a spherical surface far removed from the scattering object with its center at the centroid of the object. That result was given previously [G. Maidanik, J. Acoust. Soc. Am. 30, 620-623 (1956)]. Another expression given recently [Z. W. Fan et al., J. Acoust. Soc. Am. 124, 2727-2732 (2008)] is simplified to this formula. Comments are made on obtaining it directly from the general theorem of angular momentum conservation in the integral form.     

Theoretical and experimental examination of near-field acoustic levitation

A planar object can be levitated stably close to a piston sound source by making use of acoustic radiation pressure. This phenomenon is called near-field acoustic levitation [Y. Hashimoto et al., J. Acoust. Soc. Am. 100, 2057-2061 (1996)]. In the present article, the levitation distance is predicted theoretically by numerically solving basic equations in a compressible viscous fluid subject to the appropriate initial and boundary conditions. Additionally, experiments are carried out using a 19.5-kHz piston source with a 40-mm aperture and various aluminum disks of different sizes. The measured levitation distance agrees well with the theory, which is different from a conventional theory, and the levitation distance is not inversely proportional to the square root of the surface density of the levitated disk in a strict sense.     

Matched-field processing using measured replica fields

An approach for avoiding the problem of environmental uncertainty is tested using data from the TESPEX experiments. Acoustic data basing is an alternative to the difficult task of characterizing the environment by performing direct measurements and solving inverse problems. A source is towed throughout the region of interest to obtain a database of the acoustic field on an array of receivers. With this approach, there is no need to determine environmental parameters or solve the wave equation. Replica fields from an acoustic database are used to perform environmental source tracking [J. Acoust. Soc. Am. 94, 3335-3341 (1993)], which exploits environmental complexity and source motion.     

Acoustic fields in binary gas mixtures: mutual diffusion effects throughout and beyond the boundary layers

The acoustic behavior in thermo-viscous gas mixtures, both in proximity of walls and far from them (outside the boundary layers), involves deviations from the adiabatic and laminar movements in pure gases, which result from the influence of several diffusive fields, namely, shear, entropic, and concentration variation fields (their energy being provided by the acoustic field itself). Owing to the boundary conditions, that are slip condition, isothermal condition and concentration flux vanishing on the walls, a strong coupling between these fields occurs inside the boundary layers while their effects appear to be simple additive processes in the bulk of the medium. Although recent literature on this subject leads to interesting results, opening the way to several new issues [R. Raspet et al., J. Acoust. Soc. Am. 105, 65-73 (1999); R. Raspet et al., J. Acoust. Soc. Am. 112, 1414-1422 (2002); G. W. Swift and P. S. Spoor, J. Acoust. Soc. Am. 106, 1794-1800 (1999); D. A. Geller and G. W. Swift, J. Acoust. Soc. Am. 111, 1675-1684 (2002)], the results available still have limitations because they do not provide complete solutions for the propagative and diffusive fields throughout and beyond the boundary layers. The present work aims at providing these solutions in the whole domains considered. The results allow interpreting analytically the behavior of the fields above mentioned in closed cavities and ducts, and particularly in spherical cavities which are best suited to develop metrological applications.     

Broadband matched-field processing: coherent and incoherent approaches

Matched-field based methods always involve the comparison of the output of a physical model and the actual data. The method of comparison and the nature of the data varies according to the problem at hand, but the result becomes always largely conditioned by the accurateness of the physical model and the amount of data available. The usage of broadband methods has become a widely used approach to increase the amount of data and to stabilize the estimation process. Due to the difficulties to accurately predict the phase of the acoustic field the problem whether the information should be coherently or incoherently combined across frequency has been an open debate in the last years. This paper provides a data consistent model for the observed signal, formed by a deterministic channel structure multiplied by a perturbation random factor plus noise. The cross-frequency channel structure and the decorrelation of the perturbation random factor are shown to be the main causes of processor performance degradation. Different Bartlett processors, such as the incoherent processor [Baggeroer et al., J. Acoust. Soc. Am. 80, 571-587 (1988)], the coherent normalized processor [Z.-H. Michalopoulou, IEEE J. Ocean Eng. 21, 384-392 (1996)] and the matched-phase processor [Orris et al., J. Acoust. Soc. Am. 107, 2563-2375 (2000)], are reviewed and compared to the proposed cross-frequency incoherent processor. It is analytically shown that the proposed processor has the same performance as the matched-phase processor at the maximum of the ambiguity surface, without the need for estimating the phase terms and thus having an extremely low computational cost.     

Ultrasound field estimation method using a secondary source-array numerically constructed from a limited number of pressure measurements

A new and faster method for the accurate estimation of acoustic fields of underwater ultrasonic transducers was developed, tested experimentally, and compared to previously reported methods. Using a limited number of pressure measurements close to the transducer's face, the method numerically constructs a virtual secondary source-array whose acoustic field is similar to the field generated by the actual transducer (primary source). The measured data are used to obtain the normal particle velocity on the surface of the virtual secondary source-array, which in turn permits the calculation of the forward propagating field using the Rayleigh-Sommerfeld diffraction integral. The method is novel in that it constructs a virtual secondary source-array, thus eliminating the problems associated with obtaining the excitation source of a real transducer; and it is faster because it uses finite differences instead of a matrix inversion to obtain the excitation source. Results showed that predicted ultrasound fields agreed quantitatively and qualitatively with measured fields for three commonly used transducer types: two planar radiators (one circular, 0.5 MHz, 1.9-cm diam.; and one square, 1 MHz, 1.2 cm on a side), and a sharply focused radiator (1.5 MHz, 10-cm diam., 10-cm radius of curvature). The agreements suggest that the secondary source-array method (SSAM) is applicable to a wide range of radiator sizes, shapes, and operating frequencies. The SSAM was also compared to these authors' previous equivalent phased array methods (EPAM) [J. Acoust. Soc. Am. 102, 2734-2741 (1997); and Concentric ring equivalent phased array method (CREPAM), UFFC 46, 830-841 (1999)] which require matrix inversions. The SSAM proved to be much faster and equally or more nearly accurate than the previous methods.     

On the interpretability of speech/nonspeech comparisons: a reply to Fowler

Fowler [J. Acoust. Soc. Am. 88, 1236-1249 (1990)] makes a set of claims on the basis of which she denies the general interpretability of experiments that compare the perception of speech sounds to the perception of acoustically analogous nonspeech sound. She also challenges a specific auditory hypothesis offered by Diehl and Walsh [J. Acoust. Soc. Am. 85, 2154-2164 (1989)] to explain the stimulus-length effect in the perception of stops and glides. It will be argued that her conclusions are unwarranted.     

Examining the use of a time-varying loudness algorithm for quantifying characteristics of nonlinearly propagated noise (L)

A previous letter by Gee et al. [J. Acoust. Soc. Am. 121, EL1-EL7 (2007)] revealed likely shortcomings in using common, stationary (long-term) spectrum-based measures to quantify the perception of nonlinearly propagated noise. Here, the Glasberg and Moore [J. Audio Eng. Soc. 50, 331-342 (2002)] algorithm for time-varying loudness is investigated. Their short-term loudness, when applied to a shock-containing broadband signal and a phase-randomized signal with equivalent long-term spectrum, does not show a significant difference in loudness between the signals. Further analysis and discussion focus on the possible utility of the instantaneous loudness and the need for additional investigation in this area.