bluuebunny/crossref_metadata_2025 huggingface dataset:
Coupled hydrodynamic ship wake and PE-based acoustic propagation model
Authors: Balani Aditya , Culver R. Lee , Bradley David L. , Paterson Eric G. , Di Xiao , Kunz Robert F.
Published: October 2013

A full-wave, one-way, 2D parabolic equation (PE) code is used to predict sound propagation through a complex hydrodynamic bubble field. The normalized velocity component obtained from a hydrodynamic ship wake model is the basis for constructing the bubble distribution. A three-dimensional field is used to represent the complex environment: down wake (toward or away from the ship), cross wake (perpendicular to the wake axis), and with depth. Although the hydrodynamic wake model generates three-dimensional wake-bubble fields, the PE program is a 2D model. The problem is assumed to be two-dimensional by taking various two-dimensional cuts of the wake field, which are then used in the PE simulation. The wake of a ship is a complicated and challenging environment in which to model acoustic propagation. Predictions are made for the representation of the wake at various distances from the source ship. From this study a few results are seen: (1) The sound level corresponding to the strength of a spherically spreading sound field can be obtained. (2) Sound propagation through the wake bubble field is a three-dimensional problem. (3) Small changes in void fraction can produce noticeable differences in the propagation. [Work supported by ONR under Award No. N00014-02-1-0156.]
Propagation modeling issues for macro/micro scale environments
Authors: McCoy John
Published: October 2013

The propagations of acoustic signals through a porous solid and through a bubble cloud are both representative of a class of experiments for environments that are observable on macro and micro length scales. Intuitive models for these propagations are described by theories for poroelastic solids and for frothy fluids, respectively. These theories reflect a different intuitive understanding of the underlying physics that applies to the micro scale, and accommodate different phenomenology that can be observed on the macro scale. A more precise understanding of the linkage between micro and macro scale observations is a fundamental problem that has attracted a number of different approaches that are usually left unrelated. Moreover, there are other phenomenology that, while observable on a macro scale, are not modeled by either theory, and other models that are developed for these other phenomenology. An example of another phenomenology is a continuous backscattering occasioned by micro scale heterogeneity, which results in a kind of localization in one dimension. An example of another model is a radiative transport model. These issues are discussed in the light of still another approach to accommodating macro/micro scale variation, this one based on a phase space, spatial filtering. [Work supported by ONR.]
A modal approach to the prediction of the sound reduction index
Authors: Tisseyre Alain , Courné Cécile , Buzzy Thomas , Moulinier André
Published: October 2013

The calculation of the sound reduction index in modal analysis is presented in a general way; different possible approaches are described. These calculations are done in two steps: a vibratory study to determine the transverse displacement of the plate and a study of radiation. The specificity of orthotropic plates is presented. This study led to programming a calculation algorithm. Initial hypotheses are indicated, as well as results obtained for various plates or partitions. Modal analysis calculation results are then compared to the Cremer–Sewell approach results.
Experimental determination of porous media properties for predicting absorption of bulk-reacting surfaces
Authors: Rouse Jerry W. , Franzoni Linda P. , Jaworski Justin W.
Published: October 2013

Porous materials are commonly used and located along enclosure boundaries for the purpose of sound attenuation. An accurate representation of the boundary condition is needed for the analytical modeling of these enclosures. The proper characterization of porous materials requires the determination of the effective complex phase speed and effective complex density of the fluid during propagation through the porous material. This work presents a two-step process in which these characteristic properties are determined using a standing wave tube device. The normal incidence impedance of a porous material is obtained for two cases: a zero velocity condition, and a zero pressure condition, behind the material sample. The effective phase speed and density are then obtained from formulas containing the normal incidence impedances. For verification, standing waves are predicted theoretically using the previously obtained parameters as inputs for the case of a two-dimensional slender duct with the porous material lining a side wall. These predictions are compared to experimental measurements of sound pressure in the two-dimensional duct and good agreement is observed for the open-cell foams considered in this work.
Comparison of echo criteria for a large fan-shaped auditorium
Authors: Locey Lance L. , Leishman Timothy W. , Sommerfeldt Scott D. , Anderson Brian E.
Published: October 2013

Complaints of perceived echoes at specific locations in a 21,000 seat fan-shaped auditorium have prompted the measurement and analysis of numerous impulse responses. The responses were first processed using the echo criterion of Niese, then using the criterion of Dietsch and Kraak. This paper compares the two criteria and explores their abilities to assess whether peaks and anomalies of the measured responses were likely to produce audible echoes in the hall. The Niese criterion was found to better predict the perception of echoes produced by broad irregular decay trends. The Dietsch and Kraak criterion was able to better predict echoes produced by sufficiently strong specular reflections. Neither criterion alone was able to fully characterize these perceptions. Results obtained for both criteria at various seat locations will be presented and compared with subjective findings.
Using measured acoustic data to obtain estimates of seabed physical parameters
Authors: Knobles D. P.
Published: October 2013

Inversion for seabed parameters is part of a broader analysis that seeks to characterize the nature of and to quantify the predictability of sound propagation in complex shallow seas. This talk examines results obtained from numerous inversions using real data and attempts to place inversion techniques in perspective as a tool for understanding the physics of the interaction of sound with various types of seabeds. Data sets include those obtained in the Gulf of Mexico, off the eastern coast of the United States, and off the Korean Peninsula. Acoustic data include multifrequency measurements of transmission loss versus range and time series generated by both impulsive sources and surface ships received on both HLA and VLA. The issue of uniqueness is addressed by a combination of methods that include the comparison of inverted results to geophysical measurements, cost function sensitivity, and a comparison of modeled results to independent data not used in the inversion and localization. Of particular interest is the attenuation structure in both hard and soft sediments and the acoustic data required to extract this information. [Work supported by ONR.]
Optimal search strategies for nonlinear, multi-parameter geoacoustic inverse problems
Authors: Potty Gopu R. , Miller James H. , Lazauski Colin J. , Chen Chuen-Song
Published: October 2013

A mode travel time based technique for geoacoustic inversions has been developed using broadband explosive sources [Potty et al., J. Acoust. Soc. Am. 108, 973–986 (2000)]. In this inversion scheme we search for various parameters including water depth, sound speed in the water column represented by Empirical Orthogonal Function (EOFs), compressional wave speed in the sediment and sub-base, thickness of the sediment and range. The sensitivity of the objective function to perturbations in these parameters is different, i.e., different parameters have different sensitivities. The most sensitive parameters can be estimated with higher accuracy compared to parameters with low sensitivities. In this study we aim to link the search process to parameter sensitivities by matching the fineness of search to sensitivities. The highly sensitive parameters will be searched with a fine scale sampling of the model space with a coarser sampling adopted for less sensitive model parameters. This condition will be incorporated into the search tool (Genetic Algorithm) used for the inversion scheme. The objective of this approach is to utilize the computational resources in an optimum manner by giving more emphasis to parameters which are more sensitive as opposed to a uniform weighting for all parameters. This approach will be applied to data from the ASIAEX-2001 and Shelf Break Primer data sets. [Work supported by ONR.]
Geoacoustic inversion techniques (GAIT) Version 1.0 global search (GS)
Authors: Neumann Peter , Muncill Gregory
Published: October 2013

Geoacoustic Inversion Techniques (GAIT) Version 1.0 is a PEO (C4I and Space) PMW 155 funded product that accepts measured acoustic data and produces an optimized estimate of the bottom environment that produced the observed acoustic data. The Global Search (GS) segment of GAIT pairs the Adaptive Simulated Annealing (ASA) algorithm with a variety of Navy standard propagation loss models (PE, ASTRAL and Nautilus) and an active sonar performance prediction model (ASPM). The goal of the GS segment of GAIT is to provide a best estimate of the geoacoustic properties of the ocean bottom that, when paired with a selected model, result in the observed acoustic data. An overview of the GS segment of GAIT 1.0 will be presented with details on the ASA algorithm, component models, cost functions and geoacoustic parametrizations. Inversion results will be shown for synthetic test cases from the Inversion Technique Workshop (ITW) held in May 2001 and from both narrowband and broadband measured data test cases. [Work supported by PEO (C4I and Space) PMW 155 and uses the products of a Phase I and II SBIR from the ONR (Code 321US).]
Active rapid geo-acoustic characterization
Authors: Heaney Kevin D.
Published: October 2013

There is a growing body of evidence that the use of archival database values to estimate acoustic propagation and reverberation leads to substantial errors. To address this issue, ideas have been proposed to use mono-static reverberation to perform simultaneous geo-acoustic and scattering strength inversions. The difficulty with this technique has to do with the underlying physics of propagation and reverberation. The current techniques of using mono-static reverberation to invert for scattering strength and the geo-acoustic parameters are fundamentally limited by the fact that all reverberant energy must propagate from the source to the bottom and then back to the receiver. Thus all reverberant energy contains twice the transmission loss (TL). This ambiguity is un-resolvable from only a direct mono-static reverberation measurement. This ambiguity represents itself as a 1 dB error in estimation of the TL yields a 2 dB error in the scattering strength (SS). Further, an accurate sediment characterization and scattering strength determination, handling the range-dependence of these variables, is required. To address this problem, and to break the inherent degeneracy, a combined reverb direct blast inversion is proposed. This approach is applied to several sets of data from the Harsh Environmental Program (HEP).
Estimation of error bounds in geoacoustic inversions
Authors: Sanders William
Published: October 2013

Geoacoustic inversion has been shown to yield accurate estimates of effective ocean bottom parameters for simple environments when high fidelity models are used for both environmental deconstruction and signal propagation. But as the environment becomes more complex, some parameters are estimated with less certainty than others. Uncertainty, as represented by errors in inverse problems stem from measurement inaccuracies, model imperfections and prior assumptions. Whereas the propagation of errors in inverse problems is not generally possible, the problem becomes tractable if the forward equation can be linearized about some (preferably the true) set of values and, if further, all errors are assumed Gaussian. Under these assumptions, the covariances of the a posteriori errors can be formulated, thus providing bounds on the uncertainty resulting from the inverse process. These are ultimately expressed in terms of bounds on measurement errors, modeling errors and the linearization of the forward model. This effort analyzes the errors involved in some well understood and benchmarked cases and compares results to other published analyses. This is done by utilizing a parabolic equation (PE) propagation model. Derivatives of the field with respect to the environmental variables are derived in order to calculate the error bounds.
Two methods for solving a 3D acoustic inverse scattering problem
Authors: Seydou Fadoulourahmane , Gumerov Nail A. , Duraiswami Ramani
Published: October 2013

We consider the problem of finding the refractive index of a buried object by using far-field measurements in an inhomogeneous medium. We describe two methods for solving the inverse problem. Both methods are implemented in two steps in order to better deal with the ill-posedness of the problem. In the first method an integral equation of the first kind is derived for the far-field operator which is solved via least-squares and Tikhonov regularization. We then use the solution of the integral equation to derive an over-posed boundary value problem, i.e., the Helmholtz equation in a bounded domain with Cauchy data on the boundary. The index that satisfies this over-posed problem most closely is obtained via the Levenberg–Marquardt algorithm. The second method is an iterative method and is based on the Lippmann–Schwinger equation. It is implemented via the Newton method. The first step consists, as for the other method. Here we use a Fourier integral approach and regularization via discretization. The second step is to obtain the index by iterating the Lippmann–Schwinger equation starting with the Born approximation.
Acoustic cavitation movies
Authors: Crum Lawrence A.
Published: October 2013

Acoustic cavitation is a phenomenon that occurs on microsecond time scales and micron length scales, yet, it has many macroscopic manifestations. Accordingly, it is often difficult, at least for the author, to form realistic physical descriptions of the specific mechanisms through which it expresses itself in our macroscopic world. For example, there are still many who believe that cavitation erosion is due to the shock wave that is emitted by bubble implosion, rather than the liquid jet created on asymmetric collapse...and they may be right. Over the years, the author has accumulated a number of movies and high-speed photographs of cavitation activity, which he uses to form his own visual references. In the time allotted, he will show a number of these movies and photographs and discuss their relevance to existing technological problems. A limited number of CDs containing the presented materials will be available to interested individuals. [Work supported in part by the NIH, USAMRMC, and the ONR.]
Microbubble dynamics in the ultrasound field
Authors: Matsumoto Yoichiro
Published: October 2013

Ultrasonic biomedical applications such as ultrasound imaging with microbubble contrast agents, sonodynamic therapy, and so on, have attracted much attention in recent years. These phenomena have a close relationship to the behavior of microbubbles, so that it is essential to understanding the microbubble dynamics. A microbubble’s motion in ultrasound field is simulated in detail by the full equations for mass, momentum, and energy in both phases. It is revealed that the bubble motion is much influenced by internal phenomena such as thermal diffusion, mist formation, mass diffusion, heat and mass transfer through the bubble wall. As the driving acoustic amplitude is increased, steep increases of the oscillation amplitude occur with the largest radius observed around the resonant bubble radius. Subharmonic and higher harmonic oscillations become obvious. The acoustic and thermal radiation from the microbubbles are analyzed and it is found that considerably stronger higher harmonic emissions are obtained from smaller bubbles than for the resonant bubbles.
What are the limits of energy focusing in sonoluminescence?
Authors: Putterman Seth , Camara C. , Kappus B. , Su C. K. , Kirilov E.
Published: October 2013

Sonoluminescence [SL] is amazing for the extraordinary degree by which ultrasonic energy can be focused by a cavitating bubble. Local energy dissipation exceeds Kirkhoff’s law by 1E15 and the acoustic energy density concentrates by 12 orders of magnitude to create picosecond flashes of broadband ultraviolet light. At the minimum bubble radius, the acceleration exceeds 1E11 g and a megabar level shock wave is emitted into the surrounding fluid. For single bubbles driven at 30 KHz, SL is nature’s smallest blackbody. This implies that the bubble’s interior is such a dense plasma that the photon-matter mean free path is shorter than the wavelength of light, and suggests that SL originates in an unusual state of matter. Excitation of a vertical column of fluid [∼10 Hz] so as to create a water hammer leads to the upscaling of SL and generation of flashes of light with 3E8 photons and peak powers approaching 1 W. At 1 MHz, the spectrum resembles bremsstrahlung from a transparent plasma with a temperature ∼1 MK. At 10 MHz the collapsed size of the SL bubble approaches 10 nm, which raises the possibility that the SL parameter space may extend to the domain of quantum mechanics. [Research supported by DARPA and DOE.]
Use of intensity interferometry to determine the size of a cavitation hot-spot
Authors: Camara Carlos , Putterman Seth , Weninger Keith , Evans Paul
Published: October 2013

Although it is generally accepted that a sonoluminescing bubble collapses to a volume determined by the van Der Waals hard core of its contents, the size of the light-emitting region remains undetermined. Photos of the hot-spot indicate that the resolution of its size is diffraction limited. Intensity interferometry can resolve smaller sizes as well as the flash width if the light is incoherent. An experiment aimed at exploiting this is underway.
Nonlinear bubble dynamics in viscoelastic media
Authors: Emelianov Stanislav Y. , Hamilton Mark F. , Ilinskii Yurii A. , Zabolotskaya Evgenia A.
Published: October 2013

Microbubbles are already widely used for biomedical and clinical purposes, and are further being investigated for innovative medical and industrial applications. In most cases, the Rayleigh–Plesset equation for a gas bubble in liquid is used. In this paper we present a more general equation for gas bubble dynamics in viscoelastic media such as soft tissue. The equation, describing oscillations of a spherical bubble in an isotropic, incompressible medium, contains an integral over an elastic energy density function that depends on two invariants of the strain tensor. For moderate bubble oscillations one may use Landau’s cubic expansion of energy density, which contains five elastic moduli—shear and bulk moduli corresponding to quadratic terms, and three moduli (A, B, and C) corresponding to cubic terms. With energy density expressed through invariants of the strain tensor in spherical coordinates, the total elastic energy for an incompressible medium becomes an explicit function of bubble radius and contains only two of the five moduli, the shear modulus and modulus A. Compared to the Rayleigh–Plesset equation, the resulting equation has additional terms associated with shear force and elastic nonlinearity. Further modifications of this equation account for weak compressibility, viscosity, surface tension, radiation loss, and elastic shells.
Dan Johnson—Mr. Standards
Authors: Schomer Paul
Published: October 2013

Dan Johnson had a long history of involvement in the standards program of ASA. He has chaired and/or participated in many working groups, primarily in the areas of hearing conservation and instrumentation. He was one of a rare set that was an individual expert to all four S-committees, being very familiar with instrument design from his days at Larson-Davis, and being familiar with physiological and psychological effects, noise, and vibratory effects from his work with the Air Force at Wright–Patterson AFB. Dan was the chair of S-12, the vice-chair of ASACOS, and, most recently, the chair of ASACOS and Standards Director. As Standards Director, Dan guided us through many turbulent times including establishment of the Standards Store, the change of the standards manager, the budgetary trials of 2000 and 2001, and the appeal of ANSI S12.60-2002 (Classroom Acoustics) by ARI before Accredited Standards Committee S12. He is very missed.
A CHAT with Dan Johnson
Authors: Bertrand Robert A.
Published: October 2013

Dan’s preoccupation with preventing NIHL is well known. For those of us who had the occasion to CHAT with him, we often heard him say ‘‘if ears would bleed when exposed to noise, people would pay more attention to the harmful effect of noise upon hearing.’’ His objective, as he often stated, was to eradicate NIHL so that in a few decades, it would become a historical footnote. Among his many preoccupations in his illustrious career, approaches and techniques to prevent NIHL were of primordial importance. One approach he advocated is the use of TTS instead of the STS in HCP’s. His intention was to use the identification of TTS as an easy approach to introduce appropriate measures to prevent NIHL at a stage of reversibility rather than waiting for a confirmed permanent hearing loss, as noted with a confirmed STS. One of his last projects was developing the CHAT (Change of Hearing Audio Test) for easy use both in industrial and environmental settings to identify subjects with a TTS. Several groups are interested in pursuing his aim of using the TTS in HCP’s, hoping to fulfill his comment that in a few decades, NIHL will become a historical footnote.
Dan Johnson the mentor
Authors: McKinley Richard
Published: October 2013

I first met Dan Johnson in early 1975 as I was interviewing for an engineering job with Henning von Gierke’s bioengineering and bionics laboratory at Wright–Patterson Air Force Base. From the very beginning Dan was always direct and forthright. Over the ensuing next 27 years my knowledge and respect of Dan constantly grew. This presentation will review Dan’s technical and personal contributions while at the laboratory at Wright–Patterson Air Force Base. He was instrumental in the development of a national noise exposure criteria with the equal-energy-rule, an accurate single number hearing protector attenuation measure based on ‘‘C-A,’’ an impulse noise exposure criteria, a longitudinal study of hearing loss in children, development of noise dosimeters, and description of hearing damage risk from nonoccupational noise exposures such as disco’s, bowling alleys, lawn mowers, and school buses. Dan has had a significant effect on my career. I and the many people who knew him at the laboratory miss him greatly.
Personal noise exposure assessment from small firearms
Authors: Kardous Chucri A. , Murphy William J. , Willson Robert D.
Published: October 2013

The National Institute for Occupational Safety and Health conducted noise exposure evaluations of law-enforcement personnel during firearms training at indoor and outdoor firing ranges. A representative cross section of weapons used by officers was measured. Shooters participated in live-fire exercise at an indoor firing range using three different weapons: a Beretta .400 caliber pistol, a Remington 12-gauge shotgun, and an M4 .223-caliber assault rifle. Indoor and outdoor measurements were obtained for the Smith and Wesson .357 pistol and Colt .450 and 9-mm pistols, the Glock .400 pistol, and the Heckler and Koch and Colt AR15 .223 rifles. Impulses were measured using a Bruel and Kjaer 4136 1/4-in. microphone and TASCAM digital audio tape recorder. Relevant impulse noise metrics were calculated. Peak levels ranged from 155 to 168 dB SPL. A-weighted equivalent levels ranged from 124 to 128 dBA. The contributions of the secondary weapon firings were approximately 1 to 9 dBA. Other parameters such as A/B durations, number and mixture of impulses, spectral content, energy, kurtosis, temporal spacing, and hearing protectors’ effectiveness were examined. Comparisons of applicable damage risk criteria are presented. Further studies are needed to establish an occupational impulse noise damage risk criterion.
Analysis of impact/impulse noise for predicting noise induced hearing loss
Authors: Vipperman Jeffrey S. , Prince Mary M. , Flamm Angela M.
Published: October 2013

Studies indicate that the statistical properties and temporal structure of the sound signal are important in determining the extent of hearing hazard. As part of a pilot study to examine hearing conservation program effectiveness, NIOSH collected noise samples of impact noise sources in an automobile stamping plant, focusing on jobs with peak sound levels (Lpk) of greater than 120 dB. Digital tape recordings of sounds were collected using a Type I Precision Sound Level Meter and microphone connected to a DAT tape recorder. The events were archived and processed as .wav files to extract single events of interest on CD-R media and CD audio media. A preliminary analysis of sample wavelet files was conducted to characterize each event using metrics such as the number of impulses per unit time, the repetition rate or temporal pattern of these impulses, index of peakedness, crest factor, kurtosis, coefficient of kurtosis, rise time, fall time, and peak time. The spectrum, duration, and inverse of duration for each waveform were also computed. Finally, the data were evaluated with the Auditory Hazard Assessment Algorithm (AHAAH). Improvements to data collection for a future study examining different strategies for evaluating industrial noise exposure will be discussed.
Estimating the similarity of temporal discharge patterns of auditory neurons
Authors: Shofner William
Published: October 2013

Behavioral data obtained from stimulus generalization experiments are often interpreted to imply how similar animal perceptions of test stimuli are to a training stimulus. An appropriate decision variable would be useful for evaluating physiological responses in the context of stimulus generalization tasks. The decision variable P(A) obtained from ROC analysis of spike count distributions is useful for evaluating rate coding schemes [e.g., Relkin and Pelli, J. Acoust. Soc. Am. 82, 1679–1691 (1987)], but it ignores temporal information which may exist in responses to periodic sounds. Neural autocorrelograms show temporal firing patterns following a spike and were generated for cochlear nucleus unit responses to aperiodic, quasiperiodic, and periodic stimuli. Variance from average rate was computed as a function of time for each autocorrelogram; similarity between two autocorrelograms was estimated as the ratio of these variances. A value of 1 indicates that the autocorrelograms are identical, whereas larger values indicate less similarity. Ratios were around 1 if both autocorrelograms were from responses to wideband noise and were greater than 1 if one autocorrelogram was from responses to iterated rippled noises or harmonic complex tones. This ratio may be a useful decision variable for evaluating temporal coding schemes. [Work supported by NIDCD P01 DC00293.]
Monaural and diotic detection of tones in wideband and narrow-band reproducible noise maskers
Authors: Davidson Sean A. , Carney Laurel H.
Published: October 2013

In order to test monaural and binaural processing components of binaural detection, diotic and monaural detection were studied using reproducible noises. Correlations across masker bandwidth and interaural configuration were computed. Preliminary results reveal high correlations for probability of detection across N0S0 and NmSm configurations for both wideband and narrow-band maskers, suggesting that the binaural system does not influence results across reproducible noises for diotic detection. As reported previously [Evilsizer et al., J. Acoust. Soc. Am. 111, 336–345 (2001)], significant differences in correlations between wideband and narrow-band results were observed for the diotic case, suggesting cross-frequency interactions. Preliminary findings indicate that the same result holds for the monaural case. These data will be used to test models that incorporate monaural processing based on the phase-opponency model [Carney et al., Acta Acustica United with Acustica 88, 334–347 (2002)], and binaural processing based on the Jeffress model and on the two-channel model [McAlpine et al., Nature Neurosci. 4, 396–401 (2001)]. Predictions will be made for NmSm, N0S0, and N0Sπ interaural configurations, and for wideband and narrow-band maskers. [Work supported by NIDCD.]
Biologically inspired robust onset detection
Authors: Smith Leslie S.
Published: October 2013

Onsets are rapid increases in signal strength. The common onset time in different frequency bands provides an important cue for dynamically grouping sound energy, and hence for sound streaming. Onsets are important for segmenting sounds [Smith, J. New Music Res. 23, 11–23 (1994)] and for determining where to measure IIDs and ITDs for the sound direction finding [Smith, J. Acoust. Soc. Am. 111, 2467 (2002)]. Effective onset detection requires low latency and the capacity to cope with wide variation in the dynamic range. Many neurons in the auditory brainstem are sensitive to onsets. The system filters sound into cochlea-like bands (using a Gammatone filterbank), then spike codes positive-going zero-crossings. A wide dynamic range is achieved by using multiple spike trains per filter band, each with different sensitivity. The spike trains from each band innervate a leaky integrate-and-fire neuron. The excitatory synapses from the spike trains are fast and depressing: the shunting inhibitory synapses are facilitating and slower. The combined effect is that the neuron produces a single spike for each onset over a wide dynamic range with very low latency. The use of both inhibitory and excitatory synapses improves onset detection over purely excitatory synapses, leading to a better sound direction finding than previously reported.
Otoacoustic emissions and the ‘‘set of the center:’’ Covariances linking ear, brainstem, and cortex, documented with the AXS Test Battery
Authors: Lauter Judith L.
Published: October 2013

Although otoacoustic emissions (OAEs) are believed to reflect afferent processes connecting the central auditory system with the ear, such relations have not been demonstrated in humans, for central components rostral to the lower pons. The Auditory Cross-Section (AXS) Test Battery (Lauter, 2000, 2002) provides a straightforward way to do this, with the result that one can observe covariances between any of a number of OAE variables (TEOAE amplitude, DPOAE stability, etc.), and measures of the physiological status of the auditory nerve, brainstem centers from caudal pons to midbrain, and auditory cortex. The testing methodologies employed are relatively inexpensive and should be within the means of many more clinics and laboratories than can afford access to brain-imaging technologies such as fMRI and PET. In addition, a new statistical means of testing covariation in multivariate, within-subject data sets (Ninness and Lauter, 2003) will be described that offers a means of mathematically testing the results. Data illustrating this approach will be presented, documenting covariances observed during spontaneous day-to-day systemic fluctuations, as well as responses to medications.
Speech enhancement using dynamic synapse neural networks
Authors: Namarvar Hassan H. , Berger Theodore W.
Published: October 2013

An idea of speech enhancement using a Dynamic Synapse Neural Network (DSNN) with an extended Kalman filtering (EKF) training method is described. The goal of this study is to introduce a new methodology in better speech enhancement in the presence of continuous environment background noise, such as fans and air-conditioning units. The efficiency of this method is shown by applying it to noisy speech signals to remove recorded laboratory noise from signals at different signal-to-noise ratio levels. The preliminary results have been encouraging enough to justify our idea. To provide more noise robustness, this could be used as a pre-processing level in automatic speech recognition (ASR) systems. The proposed method would have a profound impact on the performance of ASR systems. [Work supported by DARPA CBS, NASA, and ONR.]
Text-to-speech from concatenation of articulatory units derived from natural speech
Authors: Sinder Daniel J. , Sondhi M. Mohan
Published: October 2013

It has been conjectured that articulatory synthesis possesses the greatest potential for generating high quality synthetic speech. However, for text-to-speech (TTS), waveform concatenation techniques have proven more practical due in part to the challenge of generating appropriate trajectories of articulatory parameters. A waveform generation method for TTS that combines the practical success of concatenative methods with the quality potential of articulatory synthesis is under development. The system concatenates articulatory units derived from natural speech using an articulatory voice mimic. The mimic estimates articulatory parameters by minimizing a cost function that includes a spectral distance between natural and synthetic speech and a geometric distance that penalizes rapid or discontinuous changes in articulator positions. A database of articulatory trajectories representing phonetic units is constructed from the estimated parameters. For TTS, phonetic units generated by text analysis are used to select the corresponding articulatory units from the database. Duration modification, concatenation, and smoothing across units are performed in the articulatory domain resulting in a single articulatory trajectory for the complete utterance. Speech is synthesized from the trajectory using a two mass model for voicing, achieving a high degree of acoustic continuity across unit boundaries while also allowing for source–tract interaction.
Speaker identification by difference sum and correlation coefficients of narrow-band spectrum
Authors: Yang Byunggon , Kang SunMee
Published: October 2013

Some problems in speaker identification procedures were examined: transformation of acoustic parameters into auditory scales, invalid measurement values, and comparability of spectral energy values across the frequency range. To resolve those problems, the acoustic spectral energy of three Korean numbers produced by ten female students from narrow-band spectrograms at 19 proportional time points of each voiced segment were analyzed. Then, cells of the first five spectral matrices were averaged to form a matrix model for each speaker. The correlation coefficients and sum of the absolute amplitude difference in each pair of the spectral models of the ten subjects were obtained. Also, some individual matrix models were compared to those of the same subject or the other subject with a similar spectral model. Results showed that in numbers ‘‘2’’ and ‘‘9’’ subjects could not be clearly distinguished from the others but in number ‘‘4’’ it shed some possibility of setting threshold values for speaker identification if the coefficients and the sum of absolute difference were employed. Further studies would be desirable on various combinations of the range of long-term average spectra and the degree of signal pre-emphasis. [Work supported by grant No. R01-1999-000-00229-0 from the Korea Science & Engineering Foundation.]
Systems for stereophonic sound reinforcement
Authors: Brown Jim
Published: October 2013

Although stereophonic loudspeaker systems for large spaces were developed in well-documented work at Bell Labs nearly 70 years ago, most contemporary designers appear to be ignorant of the critically important basic principles as applied to modern reinforcement systems. In fact, many designers don’t believe stereo can even work in a room larger than a living room. This paper draws on both the literature and the author’s experience over nearly 20 years with both permanent and portable systems using two and three front-referenced channels. Design criteria and examples of both good and problematic design practices are presented.
Real-time control of sound diffusion parameters
Authors: Leider Colby
Published: October 2013

Much electronic and computer music relies extensively on real-time diffusion of electronic sound in surround loudspeaker configurations. The most common method of sound diffusion in practice today is to use standard sound-reinforcement mixers whereby each fader controls the playback volume of a corresponding loudspeaker in a concert hall. Using the mixer this way, however, presents problems when one attempts to create sound trajectories, because complex, precise, and repeatable control of the individual faders is required. To address these interface issues, two different handheld controllers were created for real-time sound diffusion. The first controller is equipped accelerometers, force-sensing resistors, and joysticks and maps using input into diffusion parameters. The second controller mimics the operation of a standard mixer but allows more rapid movement of fader positions by replacing each fader with a bend sensor. To test the validity of each controller, twenty undergraduate music engineering students were asked to repeatedly perform various diffusion tasks (such as moving a monaural sound in a circle around the audience) using a standard mixer and each of the controllers. The accuracy and speed of their performances were tracked. The study concludes with lessons learned from the statistics gathered. [Research supported by the University of Miami.]
The virtual microphone technique in active sound field control systems
Authors: Lampropoulos Iraklis E. , Shimizu Yasushi
Published: October 2013

Active Sound Field Control (AFC) has been proven very useful in reverberation enhancement applications in large rooms. However, feedback control is required in order to eliminate peaks in the frequency response of the system. The present research closely follows the studies of Shimizu in AFC, in which smoothing of the rooms transfer function is achieved by averaging the impulse responses of multiple microphones. ‘‘The virtual or rotating microphone technique’’ reduces the number of microphones in the aforementioned AFC technology, while still achieving the same acoustical effects in the room. After the impulse responses at previously specified pairs of microphone positions are measured, the ratio of transfer functions for every pair is calculated, thus yielding a constant K. Next, microphones are removed and their impulse responses are reproduced by processing the incoming signal of each pair through a convolver, where the computed K constants have been previously stored. Band limiting, windowing and time variance effects are critical factors, in order to reduce incoherence effects and yield reliable approximations of inverse filters and consequently calculations of K. The project is implemented in a church lacking low frequency reverberation for music and makes use of 2 physical and 2 virtual microphones.
Analysis of multiple listener equalization performance due to displacement effects
Authors: Bharitkar Sunil , Hilmes Philip , Kyriakakis Chris
Published: October 2013

Traditionally, room response equalization is performed to enhance sound quality, by reducing the effects of reverberation, at a given listener. However, room responses vary with source and listener positions. Hence, in a multiple listener environment, equalization may be performed through averaging the room responses measured at multiple listener locations. However, the performance of averaging based equalization, at the listeners, may be affected when listener positions change or due to microphone-listener position mismatch (i.e., displacement effects). In this paper, we present a statistical approach to map variations in listener positions to equalization performance of spatial average methods. The analysis is done at frequencies above the Schroeder frequency where the direct and the reverberant sound fields are uncorrelated, and the results are presented in a 3-D plot to clearly show the changes in equalization performance (in dB’s) versus mismatch and frequencies for various listener configurations relative to a fixed loudspeaker source. The results indicate that, for the analyzed listener configurations, the zone of equalization depends on distance of microphones/listeners from the source and the frequencies in the sound. [Work supported by the U.S. Dept. of Army.]
Open-loop dereverberation of multichannel room impulse responses
Authors: Lee Bowon , Hasegawa-Johnson Mark A. , Goudeseune Camille
Published: October 2013

We are developing the audio display for a CAVE-type virtual reality theater, a 3-m cube with displays covering all six rigid faces. The user’s headgear continuously reports ear positions so headphones would be possible, but we nevertheless prefer loudspeakers because this enhances the sense of total immersion. Because sounds produced at the loudspeakers are distorted by the room impulse responses, we therefore face the problem of controlling the sound at the listener’s two ears. Our proposed solution consists of open-loop acoustic point control, i.e., dereverberation. The room impulse responses from each loudspeaker to each ear of the listener are inverted using multichannel inversion methods, to create exactly the desired sound field at the listener’s ears. Because the actual room impulse responses cannot be measured in real time (as the listener walks around), instead the impulse responses simulated by the image-source method is used. A new evaluation criterion is proposed to quantitatively evaluate both the simulation and the open-loop dereverberation. The actual impulse responses used for this evaluation are measured with a starter pistol, since this best approximates the point source assumed by the image-source method.
Multichannel sound reinforcement systems at work in a learning environment
Authors: Malek John , Campbell Colin
Published: October 2013

Many people have experienced the entertaining benefits of a surround sound system, either in their own home or in a movie theater, but another application exists for multichannel sound that has for the most part gone unused. This is the application of multichannel sound systems to the learning environment. By incorporating a 7.1 surround processor and a touch panel interface programmable control system, the main lecture hall at the University of Michigan Taubman College of Architecture and Urban Planning has been converted from an ordinary lecture hall to a working audiovisual laboratory. The multichannel sound system is used in a wide variety of experiments, including exposure to sounds to test listeners’ aural perception of the tonal characteristics of varying pitch, reverberation, speech transmission index, and sound-pressure level. The touch panel’s custom interface allows a variety of user groups to control different parts of the AV system and provides preset capability that allows for numerous system configurations.
Estimating the low-frequency (0.1–1 kHz) sound speed in marine sediments using the harmonics from the propeller of a light aircraft
Authors: Buckingham Michael J. , Giddens Eric M.
Published: October 2013

During ONR’s Sediment Acoustics Experiment 1999 (SAX99) in the northeastern Gulf of Mexico, several research groups made high-precision, in situ measurements of dispersion in the medium-sand sediment at frequencies greater than 20 kHz. Comparable precision at lower frequencies is difficult to achieve with in situ time-of-flight techniques because of wavelength issues which, inter alia, dictate an inconveniently large and costly acoustic source. Yet low-frequency (1 kHz) sound speed measurements are sorely needed to distinguish between competing theoretical predictions. An alternative to the traditional travel-time approach employs a single hydrophone buried in the sediment and, instead of an in situ sound source, the low-frequency harmonics from the propeller of a light aircraft. Essentially, the airborne-source technique relies on the difference between the Doppler-shifted frequencies on aircraft approach and departure, as detected on the buried hydrophone, to yield a direct measure of the local sound speed at the detector. Experiments recently conducted about 1.5 km north of Scripps pier, using a single-engine Tobago TB10 aircraft with a two-blade propeller, will be described and the resultant estimates of the low-frequency sound speed in the fine-sand sediment at the site will be presented. [Work supported by ONR.]
Source depth and array tilt effects on seabed inversion of ambient noise
Authors: Desharnais Francine , Thomson David J. , Gillard Chris A.
Published: October 2013

Ambient noise coherence between two vertically separated sensors in shallow water relates to the directionality of the noise field and is sensitive to the reflective properties of the surficial sediments. In earlier work, an energy-flux model was developed to calculate the noise coherence over a multilayered seabed; in this model the surface sources were assigned a frequency-independent dipole directivity pattern. The model was subsequently combined with a hybrid nonlinear inversion procedure to effectively search the space of geoacoustic properties that parametrize a multilayered seabed. In order to benchmark the energy-flux model against a wave-theoretical formulation, the issue of the unknown but finite source depth of the noise sources must be addressed. These near-surface sources result in a frequency-dependent beam pattern that is not dipolar above a few hundred hertz [M. J. Buckingham and N. M. Carbone, J. Acoust. Soc. Am. 102, 2637–2644 (1997)]. In this paper, we extend the energy-flux model to include the effect of finite source depth as well as sensor tilt on the observed directivity of the ambient noise field. These extra degrees of freedom are added to the search space of the inversion procedure and their influence on estimating seabed properties from noise coherence data is investigated.
Depth partitioning of modal energy of guided acoustic waves in shallow water as an additional input for geoacoustic inversion
Authors: Pierce Allan D. , Frisk George V. , Souza Luiz L.
Published: October 2013

Modal mapping experiments (MOMAX) previously described by Souza, Frisk, and Becker [J. Acoust. Soc. Am. 112, 2281 (2001)] employ a CW-source at a fixed water depth and a receiver also at a fixed depth, with the range between the two varying systematically. The received amplitude versus range over a limited range interval yields a k-space spectrum, where the individual peaks correspond to the eigen-wavenumbers of the individual guided modes. This extraction in conjunction with a knowledge of the sound speed profile within the water column yields discrete values of the impedance at the water-bottom interface for the corresponding frequency and horizontal wavenumber, and these values in turn can be used in an approximate geoacoustic inversion for the bottom properties. The areas under these peaks are shown to be especially robust and can be given a theoretical interpretation with reference to the modal sum solution for a point source in shallow water with weak range dependence. This yields the ratio of the modal energies in the water column to those in the bottom. Both types of information are combined into a theory that yields an improved geoacoustic inversion.