bluuebunny/crossref_metadata_2025 huggingface dataset:
Shock transmission and eigenvalue veering within a coupled system
Authors: Vijayan Kiran , Woodhouse Jim
Published: October 2011

The operation of dynamical systems in harsh environments requires continuous monitoring. Internal sensors may be used to monitor the conditions in real time. A typical example is the sensor and electronic components used in space structures which, especially during launch, are subject to huge g force. The paper will present an experimental and theoretical study on a simplified model used to analyze the possible cause of high acceleration on the enclosed sensors and equipments due to impulsive loading. The model system consists of two beams coupled using compliant connections. An impulse hammer excites one beam, and vibrations are transmitted to the indirectly driven beam. A theoretical model is developed using a Rayleigh–Ritz approach and validated using experimental results in both the frequency and time domains. Monto Carlo simulation was done with random masses positioned on the indirectly driven beam to determine the worst-case conditions for maximum peak acceleration. Highest acceleration levels were found when mode matching in the two beams led to veering behavior in the coupled modes. The results suggest guidelines for the detailed design of internal components of a structure exposed to shock loading from its environment. [The authors thank Schlumberger Cambridge Research for financial support.]
Sparse acoustic response function estimation with a mixture Gaussian model
Authors: Gendron Paul
Published: October 2011

Underwater acoustic response functions at high frequencies and large bandwidths exhibit significant spatio-temporal variability that depends greatly on volume, boundaries, and the source–receiver motion. Considered here is a mixture Gaussian assignment over Doppler, beam-angle, and channel bandwidth employed to describe the behavior of the sparse acoustic response function over received signal duration, aperture, and bandwidth. Accurate modeling of the dependence between the mixture components can be handled by considering dependence among neighboring indicator variables of the mixture assignment. This allows for a more accurate and adaptable description of the natural persistence that acoustic paths exhibit and improve channel estimation quality. This adaptive structure is applied to underwater M-ary spread spectrum acoustic communication transmissions during the MACE10 experiment of the coast of Martha's Vineyard at near 10 kHz of bandwidth and at ranges of 1 and 2 km. Posterior conditional expectations of the acoustic response are compared with least squares type estimates and performance is quantified in terms of the observed bit error rate (BER) as a function of received SNR. A BER = E-6 at rSNR = −16 dB for 12 element combining is demonstrated. [This work is supported by the Office of Naval Research and by NISE BAR.]
Multi-input multi-output multicarrier acoustic communications in shallow water
Authors: Kang Taehyuk , Song Heechun , Hodgkiss William
Published: October 2011

Recently multi-carrier orthogonal frequency division multiplexing (OFDM) communications, popularly used in wireless channels, has been introduced in underwater acoustic channels with a large delay spread, as an alternative to the typical single carrier approaches. In this paper, we investigate multiple-input/multiple-output (MIMO) OFDM communications which can effectively increase the data rate in band-limited underwater channels. The performance of MIMO OFDM communications will be illustrated using the data collected from the KAM11 experiment conducted in shallow water, west of Kauai, Hawaii, which involved multiple transmit and receive arrays with different bandwidths, inter-element spacings, and apertures at various ranges between them up to 10 km.
Estimation of acoustic communication channel capacity of an ocean waveguide disturbed by surface waves
Authors: Hayward Thomas J.
Published: October 2011

The ocean environment is known to present significant challenges to underwater acoustic communication, including noise-imposed bandwidth constraints and large-scale temporal dispersion due to macro-multipath propagation. In addition, acoustic interaction with the rough, moving ocean surface degrades the spatial and temporal coherence of acoustic communication signals, thereby diminishing the achievable data rates. In this work, the effects of the ocean surface on acoustic communication channel capacity are examined in a computational study for a shallow-water waveguide bounded by an ocean surface disturbed by gravity and capillary waves. Surface wave spectra are derived from empirical models [Donelan and Pierson, J. Geophys. Res. 92, 4971] and are used to construct realizations of the 2-D moving surface. Then, 2-D and 3-D propagation models are applied to derive the space-time correlation properties of the acoustic channel, from which stochastic channel models are constructed. The adequacy of N × 2-D propagation modeling for this construction is also assessed. The channel models are then used as a basis for the estimation of channel spectral efficiency (achievable data rate per unit of frequency increment). Extensions to estimates of channel capacity are then discussed. [Work supported by the Office of Naval Research.]
Importance of surface forward scattering on reverberation level
Authors: Thorsos Eric I. , Yang Jie , Elam W. T. , Henyey Frank S. , Hefner Brian T.
Published: October 2011

Transport theory has been developed for modeling shallow water propagation at mid frequencies (1–10 kHz) where forward scattering from a rough sea surface is taken into account in a computationally efficient manner. The method is based on a decomposition of the field in terms of unperturbed modes, and forward scattering at the sea surface leads to mode coupling that is treated with perturbation theory. Transport theory has recently been extended to model shallow water reverberation, and the effect of forward scattering from the sea surface is found to be very important in accurately modeling the reverberation level under winter-like (nearly isovelocity) sound speed conditions. At a frequency of 3 kHz and a wind speed of 15 knots (7.7 m/s) with a fully developed sea, the reverberation level from sea surface scattering is found to be up to 20 dB greater than when a coherent reflection loss is used to account for the effect of surface roughness during the propagation out to and back from the primary backscatter source of reverberation. Examples illustrating the importance of sea surface forward scattering on reverberation level are shown. [Work supported by ONR Ocean Acoustics.]
Predicting surface scattering from surface elevation time series
Authors: Deane Grant B. , Preisig James C.
Published: October 2011

Surface reverberation in mid-frequency bands (we are considering 3–15 kHz) can be an important determinant for the performance of underwater communications systems operating in surface scattering environments. At these frequencies and at relatively short ranges (order 10 water depths), gravity waves focus sound energy incident on the surface, creating intensifications, Doppler shifts and phase shifts in the reflected field, all of which impact the performance of underwater acoustic communications systems. Observations of the time-varying arrival intensity structure from an experiment conducted in the Martha's Vineyard Coastal Observatory will be presented along with model calculations made using the Kirchhoff approximation. The sensitivity of the reflected field to the distribution of energy in the surface wave field and short wavelength gravity waves will will be discussed. [Work supported by the ONR Ocean Acoustics Program.]
Methods for separating harmonic instruments from a monaural mix
Authors: Bay Mert , Beauchamp James W.
Published: October 2011

During the past decade, solutions to the problem of musical sound source separation have become more evident. Potential applications include sound editing, enhanced spatialization, music-minus-one, karaoke, music classification/identification, music transcription, and computational musicology. Our current approach is to restrict the input signal to a mix of a limited number of instruments, each comprised of harmonic partials, with known F0 contours. (These can be obtained either by audio-to-midi alignment or multiple-F0 estimation.) Since harmonic frequencies of known F0s are easily predicted, binary mask separation is robust except for frequency regions where harmonics of different instruments collide. Three methods for repairing collisions are compared: (1) F0-informed non-negative matrix inversion using instrument spectral libraries; (2) least-squares estimate of collision frequencies based on a multiple sinusoidal model; and (3) common amplitude modulation (CAM) [Li et al., IEEE Trans. Audio, Speech, Lang., Process. 17(7), 2009]. Separation examples using these methods will be demonstrated.
Motion simulation in the environment for auditory research
Authors: Boren Braxton B. , Ericson Mark A.
Published: October 2011

Virtual sound source motion has been implemented in the Army Research Laboratory's Environment for Auditory Research, which contains a 57-channel spherical loudspeaker array located in an anechoic chamber. Using the Psychophysics Toolbox Version 3 low-latency PortAudio API, 57 channels of streaming audio are dynamically updated in real-time using MATLAB for signal processing. Both distance-based amplitude panning (DBAP) and vector base amplitude panning (VBAP) have been implemented in MATLAB for controlling source motion. Sources are defined on a given path, such as a circle, ellipse, or the “dog bonex”; flight pattern often used in aviation. While DBAP works convincingly for virtual sources located on the sphere defined by the loudspeaker array, VBAP is needed to position sources outside the array. Source motion paths are defined parametrically with respect to time, and playback buffers update the panned position every 11.6 ms. Based on the source's instantaneous distance, diffuse-field or free-field amplitude attenuation is added in MATLAB as well air absorption filtering. This system will be used for a variety of audio simulations and auditory experiments.
New developments in the measurement and calibration of absorption, scattering, and diffusion coefficients
Authors: D'Antonio Peter , Rife Brian
Published: October 2011

Round robin experiments have confirmed that current methods to measure the random incidence absorption coefficient are inaccurate and not reproducible from lab to lab. As a result, various improvements to ISO 354 are being considered. Among them include measures to prevent the coefficient from exceeding unity and achieve reproducibility, using calibration. Since all measurement equipment is calibrated, it seems prudent to calibrate the rev room as well. We will describe approaches used at RPG to replace suspended clouds with boundary diffusors, the use of Eyring instead of Sabine and calibration, using a non-diaphragmatic reflective sample, with the same area, thickness and perimeter as the absorber, as a “zero” or low absorption reference. The new rotating table ISO 17497-1 scattering coefficient apparatus, using dual source/simultaneous impulse response measurements will be described. Since the technique utilizes subtraction of the reflective table as a reference, some of the problems associated with the rev room method are mitigated and reproducibility is achieved. Finally, the new diffusion coefficient goniometer, utilizing 32 simultaneous impulse response measurements, will be described. As with the scattering coefficient, the normalized diffusion coefficient benefits from the use of a reflective calibration reference, thus removing edge diffraction and other effects.
The sound absorption measurement according to ISO 354
Authors: Vercammen Martijn , Lautenbach Margriet
Published: October 2011

Sound absorption measurements of building materials such as sound absorbing ceilings and other products are performed in a reverberation chamber according to ISO 354. It is known that the interlaboratory reproducibility of these measurements is not very well. At this moment, the differences of results between laboratories are much larger than can be accepted, from a practical point of view for predictions as well as from a jurisdictional point of view. An ISO working group has started to investigate possibilities to improve the method. Due to the insufficient diffuse sound field in a reverberation chamber with test sample, the shape of the reverberation room and the placing of diffusers will influence the result. A round robin research containing ten laboratories is performed to get information on the spread and if it is possible to reduce this by correcting for the mean free path or by application of a reference material. Additional measurements are performed to improve the measurement conditions such as measurements with volume diffusers. Possible improvements of ISO 354 will be presented. These consist of a procedure to qualify laboratories based on the statistical variation of the reverberation time and based on the results of a reference absorber.
Bird's singing codification using quick response barcode
Authors: Jácome Andrade Hugo Fernando , Parra Puente David
Published: October 2011

This work responds to the necessity to objectively describe bird's songs in scientific publications and field guides. Nevertheless, it can also be extended to the study of other animal like marine mammals, bats, humans, and insects. The project consists on a computer application development in MAX/MSP language, that extracts coefficients from an audio signal, those coefficients will characterize the signal, and are codified by QR tool (Quick Response Barcode), created by the Japanese company Denso-Wave in 1994. The following stage of the project, that is in developing process, is the programming of the de-codifier application, that rebuild the audio signal from those numbers, to bring back the signal to the human hearing field, as final stage of the project, it will be developed a similar application with the same intentions for mobile phones with the built-in QR reading tool.
Depth and range tracking of sperm whales in the Gulf of Alaska using a two-element vertical array, satellite and bioacoustic tags
Authors: Mathias Delphine , Thode Aaron , Straley Jan , Andrews Russel
Published: October 2011

A two-element vertical array was deployed between August, 15 and 17 2010, on the continental slope off Southeast Alaska, in 1200 m water depth. The instruments were attached to a longline fishing anchorline, deployed at 300 m depth, close to the sound-speed minimum of the deep water profile. The anchorline also served as a decoy, attracting seven depredating sperm whales to the area. Three animals were tagged with a satellite tag and one of them was tagged with both a satellite and bioacoustic “BProbe” tag. Both tags recorded dive depth information. Relative arrival times of surface- and bottom-reflected paths are used to estimate animal range and depth on a single hydrophone, and compared with tagging results. The two-element array is then used to estimate vertical arrival angles of the direct and surface-reflected paths to determine whether range and depth localization can occur without the use of bottom multipath. This data will be useful in determining whether long-range tracking of sperm whales is possible using a single compact instrument deployment. Potential applications include observing what ranges whales are willing to travel to depredate. [Work conducted under the SEASWAP program, supported by the National Oceanic and Atmospheric Administration and the North Pacific Research Board.]
N3 call types produced long-term by a killer whale of the northern resident community under controlled conditions: Characteristics, variation, and behavioral context
Authors: Nash Juliette S. , Bowles Ann E.
Published: October 2011

A-Clan of the Northern Resident killer whale (Orcinus orca) community produces a dialect of discrete stereotyped vocalizations, including N3, a call found in all pods of the clan. Ford (1984) reported that N3 is produced almost exclusively in low-arousal states. A 45-yr old female (BC-F1) collected from the A23 matriline of A5 pod and held under controlled conditions has produced this call throughout her adult life. Her calls were collected in 1985 at Marineland of the Pacific (14.25 h of samples) and every few years from 1987 to 2010 at Sea World San Diego (252 h). Analysis of the data show that BC-F1 uses two stereotyped variations of the N3 call. While temporally and structurally similar, there is a positive and negative frequency inflection distinguishing them. BC-F1 delivers N3 with the positive-inflected call preceding the negative. The N3 call occurs in low-arousal states, but also in high-arousal states, including caller interruption, breaking-up of synchronous swimming, and percussive behavior. N3 is infrequently produced during affiliative synchronous swimming and call-matching bouts. Thus, the call is not simply an indication of the resting state but transitions from one state to another. [Research supported by the author's organizations with in-kind support from SeaWorld San Diego.]
Application of active-learning techniques to enhance student-based learning objectives
Authors: Neilsen Tracianne B. , Gee Kent L.
Published: October 2011

Research in physics education has indicated that the traditional lecture-style class is not the most efficient way to teach science courses at the university level. Current best-teaching practices focus on creating an active-learning environment and emphasize the role of the students in the learning process. Several of the recommended techniques have recently been applied to Brigham Young University's acoustics courses. Adjustments have been built on a foundation of establishing student-based learning outcomes and attempting to align these objectives with assessments and course activities. Improvements have been made to nearly every aspect of the courses including use of class time, assessment materials, and time the students spend out of the classroom. The progress made in bringing two of the courses, specifically an introductory, descriptive acoustics course for a general audience and a junior level introduction to acoustics course for majors, is described. Many of the principles can be similarly applied to acoustics education at other academic levels. Suggestions are made for those seeking to modernize courses at their institutions.
Can a statewide noise rule cover both urban and rural areas?
Authors: Braslau David
Published: October 2011

Minnesota adopted a statewide noise rule in 1974. Statistical descriptors L10 and L50 for daytime and nighttime periods were established, along with a state law that no local government can adopt more stringent standards. While these are land use receiving standards, the rule is applied to noise generated by individual sources. The rule has worked well in urban environments but control of existing and new sources in rural environments has been problematical, both for generators and receptors of noise. The intent of the rule is to provide criteria that potential new development can take into account when proposing and designing new facilities, but in rural areas with very low ambient levels, the rule allows very large increases in sound level against which receptors have no recourse. Local governments have taken other approaches to control new sound sources by proposing acceptable increases over ambient noise level at receiving land uses or simply denying permits to new noise sources. Sources such as wind turbines are particularly difficult to address, since these are commonly placed in quiet rural areas. The paper discusses some problems and attempted solutions to allow new noise sources in rural areas without compromising quiet rural environments.
Acoustic microscopy investigation of superconducting materials
Authors: Tahraoui T. , Debboub S. , Boumaïza Y. , Boudour A.
Published: October 2011

The present work has as subject the study of the ultrasonic attenuation in some superconductor materials. This study is based on the simulation of the acoustic signal obtained by the reflection acoustic microscope upon the exploration of a coated or uncoated material. The examination of the simulated signal has permitted the determination of the variation of the reflection coefficient with respect to the incidence angle of the exciting acoustic wave. We have also determined the elastic constants, the velocities of different modes of propagation, the acoustic attenuation of the Rayleigh mode as a function of temperature from the reflection coefficient, and the acoustic signature.
Green's function retrieval from noise by multidimensional deconvolution
Authors: Wapenaar Kees , van der Neut Joost , Slob Evert , Snieder Roel
Published: October 2011

The correlation of noise at two receivers is approximately proportional to the Green's function between these receivers. The approximation is accurate when the medium is lossless and the noise field is equipartitioned. These assumptions are in practice often violated: the medium of interest is often illuminated from one side only, the sources may be irregularly distributed and losses may be significant. For those situations, the correlation function is proportional to a Green's function with a blurred source. The source blurring is quantified by a so-called point-spread function which, like the correlation function, can be derived from the observed data (i.e., without the need to know the actual sources and the medium). The blurred source can be focused by multidimensionally deconvolving the correlation function for the point-spread function. We illustrate the correlation and deconvolution methods with several examples and discuss the advantages and limitations of both methods.
In defense of discrete features
Authors: Iskarous Khalil
Published: October 2011

A major challenge for discrete feature theories of speech perception is that the acoustic signal is continuously changing within and between segments at a high rate. These theories must explain how continuously changing spectral patterns are perceived as discrete and independent features. This problem does not disappear after auditory transformations of the acoustic signal, since temporal and frequency resolution, despite being coarse, still preserve most of the signal dynamics. In this work, it is shown that if articulatory features are extracted from the continuously changing acoustic signal, assuming that the signal is the output of a lossless vocal tract terminated in a unit resistance, i.e., a Darlington configuration (Iskarous, Journal of Phonetics, 2010), it is possible to obtain a discretely changing place of articulation feature from the continuous signal change. This will be illustrated using articulatory inversion of VV and CV transitions from the X-ray microbeam database for five male and five female participants. This work supports theories of discrete and independent perceptual features, as assumed for instance by Miller and Nicely (1955), if the acoustic signal is interpreted in terms of the vocal tract actions that cause it (Goldstein and Fowler, 2003).
Geophysical parameter estimation
Authors: Deffenbaugh Max
Published: October 2011

Seismic exploration for oil and gas is a parameter estimation problem. Geological properties and fluid content of subsurface reservoirs are sensed from the earth's surface. Seismic data are acquired by generating elastic waves at the surface and recording the reflections off subsurface targets using large receiver arrays. At present, it is not practical to determine earth parameters by iteratively refining an earth model until simulated seismic data match recorded field data. The computational demands of accurately simulating viscoelastic wave propagation in a 3-D heterogeneous earth are prohibitive. Instead, seismic data are processed to remove many complexities of wave propagation in the real environment, like multipath, attenuation, dispersion, shear waves, and interface waves. With these complexities removed, the processed data conform to simpler propagation models which can be solved in reasonable computer time. The cost of each simplification, however, is the loss of some information about the subsurface parameters of interest. This tradeoff between the tractability of the inversion and the accuracy of the result can be quantified by comparing computational time versus the Cramér-Rao bound on subsurface parameter estimates. Examples are discussed for some commonly used seismic signal processing algorithms.
A moment-based technique to palliate pervasive clutter while preserving objects of interest
Authors: Gauss Roger C. , Fialkowski Joseph M.
Published: October 2011

High false alarm rates are a persistent problem for ASW active sonars. This talk describes a recently developed sonar clutter characterization and control method, the “Poisson-Rayleigh Inspired Method” (PRIM). The closed-form PRIM is based on NRL's two-parameter Poisson-Rayleigh (P-R) model that, like the popular K-type model, provides a physical context for relating the characteristics of data distributions to scatterer attributes (density and relative strength). However, with its extra degree of freedom, the P-R model offers the potential to exploit more information through higher-order (4th and 6th) data moments, and thus do a better job of characterizing clutter. The technique is demonstrated on normalized, broadband sonar clutter data collected in two range-dependent shallow-water environments, one bottom dominated and one fish dominated. The data results suggest that in contrast to the popular K-distribution, the power of the discrete-scatterer component in the P-R distribution can provide feature information that is largely independent of the peak signal-to-noise ratio of an echo, implying a potential for improved rejection of clutter relative to target-like objects. [Work supported by ONR.]
Passive acoustic tracking of marine mammals and anthropogenic sound sources with autonomous three-dimensional small-aperture arrays
Authors: Gassmann Martin , Wiggins Sean M. , Hildebrand John A.
Published: October 2011

Marine mammals produce a wide range of frequency-modulated sounds at low and high frequencies as well as directional broadband echolocation sounds in a refractive ocean environment. This creates several challenges for passive acoustic long-term tracking of the various marine mammal species. To overcome these, three-dimensional small-aperture hydrophone arrays coupled to seafloor multi-channel recording packages were deployed in a large aperture array in the Southern California Bight. Taking advantage of the experimental setup in the oceanic waveguide, time and frequency-domain tracking methods will be presented and tracks of marine mammals as well as anthropogenic sources will be shown. This provides a tool to study over long timescales behavioral responses of tracked marine mammals to tracked anthropogenic sources.
Bearings-only target localization for unmanned underwater vehicles
Authors: Eickstedt Donald P.
Published: October 2011

This paper reports on target localization for autonomous underwater vehicles (AUVs) with acoustic bearing sensors where the bearing sensor provides detection and beamforming capabilities and reports target bearings with the associated measurement uncertainties. Estimating the state of a moving target using a bearing sensor on a moving platform is a difficult problem due to both the nonlinear nature of the measurement equations with respect to the unknown target parameters but also the observability conditions for the unknown parameters which require the observer to maneuver. This paper will report on a maximum likelihood (ML) method using Levenberg–Marquardt (LM) optimization for estimating the target state parameters. A known issue with ML solutions to this problem is that the algorithm may have difficulty reaching the global minimum unless the initial solution guess is close to the true solution. This paper will report on a genetic algorithm approach to making the initial solution guess. The performance of this approach, obtained from Matlab simulation, is presented. The implementation of the algorithms in the MOOS architecture is described using a very fast Lapack implementation for the LM parameter estimation to support the goal of optimal AUV maneuvering to improve the target state estimate.
Products focused on improving acoustic performance
Authors: Freiheit Ron
Published: October 2011

New product information with a focus on performance, practice, and rehearsal room acoustics will be presented. Performance-area acoustics are enhanced by full-stage acoustical shells, ceiling panels, and clouds for the audience area. These range from standard products to unique customizations. Portable shells that roll through standard doorways will also be shown. New modular sound-isolating practice rooms feature enhanced acoustic performance, improved aesthetic appearance, and upgraded optional floating floor. Active acoustics technology can be retrofit in an existing room, such as a teaching studio. Interesting applications of acoustical doors will also be highlighted. Other products include instrument storage cabinets with tunable absorption integrated into the design. An acoustic shield positioned behind an individual performer helps reduce potentially damaging sound energy from behind, as indicated by binaural measurements and recordings. Finally, an acoustician-specific, restricted-access website features third-party test data and other commonly requested technical information.
Sound rating of a new air handling product
Authors: Lind Stephen J.
Published: October 2011

It is important in building design to have an accurate acoustical description of equipment that will operate in the building. Noise from air conditioning systems is often a significant contributor to the building acoustic environment. In order to design the building correctly, the sound levels produced by the equipment must be known. Current theoretical or computer modeling methods are not able to accurately predict the resulting unit sound levels. The sound levels produced depend on many factors including equipment design, operating conditions, options chosen, and sound component. Empirical models are needed. The test method to develop the models follows Air Conditioning, Heating, and Refrigeration Institute Standard 260. Sound power levels for a range of fan sizes, types, and operating conditions are measured for the discharge, inlet, and casing sound components. Equipment options that affect the sound as it propagates through the equipment are also measured. The test results are used to create mathematical models to describe the unit sound power. The resulting mathematical models are then incorporated into a computer program to allow the user to accurately determine the sound in the product as configured for each job.
Use of damped drywall in architectural acoustics
Authors: Shafer Benjamin M. , Tinianov Brandon
Published: October 2011

Damped drywall, specifically QuietRock, has been tested both by various NVLAP-accredited laboratories and in situ in a variety of assemblies as an acoustic treatment for the transmission of airborne sound through building partitions. The application of damped drywall and its use for noise control through building partitions continue to expand as the body of test data grows and analytic models are developed to support this class of materials. However, the manufacturer of QuietRock has conducted several research studies that illustrate how damped drywall can be used to improve the transmission loss over a broad range of frequencies and in various assemblies. This presentation is a summary of these research studies and serves as a guide for the application of damped drywall in building construction.
Image reconstruction using singular-value decomposition of scattering operators
Authors: Jiang W. , Astheimer J. , Waag R.
Published: October 2011

Efficient inverse scattering algorithms for nonradial lossy objects are presented using singular-value decomposition to form reduced rank representations of the scattering operator. These algorithms extend eigenfunction methods that are not applicable to nonradial lossy scattering objects because the scattering operators for these objects are not normal. A method of local reconstruction by segregation of scattering contributions from different local regions is also presented. Scattering from each region is isolated by forming a reduced rank representation of the scattering operator that has domain and range spaces comprised of far-field patterns with retransmitted fields that focus on the local region. Accurate methods for the estimation of the boundary, average sound speed, and average attenuation slope of the scattering object are also given. These methods yield initial approximations of scattering objects that reduce the number of distorted Born iterations needed for high quality reconstruction. Calculated scattering from two lossy elliptical objects is used to evaluate the performance of the proposed methods. In both cases, the reconstruction procedures result in high-quality quantitative images of tissue parameters with sub-wavelength resolution.
Refraction of the cyclonic microbarom signal by the cyclonic winds
Authors: Blom P. , Waxler R. , Frazier W. G. , Talmadge C. , Hetzer C.
Published: October 2011

Non-linear interaction of the ocean surface and atmosphere is known to produce narrow-band, low frequency, continuous acoustic and seismic radiation termed microbaroms and microseisms, respectively. The microbarom signal typically has an amplitude of a few microbars and a peak at 0.2 Hz. The microbaroms are generated by counter-propagating surface waves of equal period. The microbarom source location associated with a hurricane is believed to be due to the interaction of the waves produced by the cyclonic winds with the background ocean wave field and is generally located many kilometers from the eye of the storm, along a perpendicular to the direction of the ambient winds. Following up on a suggestion of Bedard and coworkers, propagation of the microbarom signal through the storm wind structure has been investigated using geometric acoustics in an inhomogeneous moving medium. Strong refraction of the signal is predicted. To observe this refraction we have deployed infrasound arrays along the US eastern seaboard. Predicted and measured back azimuths for propagation through the wind structure have been compared to the data recorded during the 2010 and 2011 Atlantic hurricane seasons.
A linearized contrast source method for full-wave modeling of nonlinear acoustic wave fields in homogeneous media
Authors: Verweij Martin D. , Demi Libertario , van Dongen Koen W. A.
Published: October 2011

Many novel medical echography modalities, like super harmonic imaging (SHI), employ imaging of higher harmonics arising from nonlinear propagation. Design optimization of dedicated imaging probes for these modalities requires accurate computation of the higher harmonic beam profiles. The existing iterative nonlinear contrast source method can perform such simulations. With this method, the nonlinear term of the Westervelt equation is considered to represent a distributed contrast source in a linear background medium. The full transient nonlinear acoustic wave field follows from the Neumann iterative solution of the corresponding integral equation. Each iteration involves the spatiotemporal convolution of the background Green's function with an estimate of the contrast source, and appropriate filtering enables a discretization approaching two points per smallest wavelength or period. To further reduce the computational load and to anticipate extension of the method, in this presentation, the effect of linearizing the nonlinear contrast source is investigated. This enables the replacement of the Neumann scheme by more efficient schemes. Numerical simulations show that for design purposes the effect of linearization on the harmonic components involved with SHI remains sufficiently small. Moreover, it is shown that a significant decrease in computational costs may be achieved by using a Bi-CGSTAB scheme.
Attenuation of elastic waves due to wave-induced vorticity diffusion in porous media
Authors: Mueller Tobias M. , Sahay Pratap N.
Published: October 2011

A theory for attenuation of elastic waves due to wave-induced vorticity diffusion in the presence of randomly correlated pore-scale heterogeneities in porous media is developed. It is shown that the vorticity field is associated with a viscous wave in the pore space, the so-called slow shear wave. The latter is linked to the porous medium acoustics through incorporation of the fluid strain rate tensor of a Newtonian fluid in the poroelastic constitutive relations. The method of statistical smoothing in random media is used to derive dynamic-equivalent elastic wave numbers accounting for the conversion scattering process into the slow shear wave. The result is a model for wave attenuation and dispersion associated with the transition from viscosity- to inertia-dominated flow regime in porous media. It is also shown that the momentum flux transfer from the slow compressional into the slow shear wave is a proxy for the dynamic permeability in porous media. A dynamic permeability model is constructed that consists of an integral over the covariance function of the random pore-scale heterogeneities modulated by the slow shear wave. In a smooth pore-throat limit, the results reproduce the dynamic permeability model proposed by Johnson etal. [J. Fluid Mech. 176, 379 (1987).]
From high-speed imaging to perception: In search of a perceptually relevant voice source model
Authors: Alwan Abeer , Keating Patricia , Kreiman Jody
Published: October 2011

The goal of this project is to develop a new source model from high-speed recordings of vocal fold vibrations and simultaneous audio recordings. By analyzing area waveforms and acoustics and performing perception experiments, we will better parameterize the new source model and also uncover those aspects of the model that are perceptually salient, leading to improved TTS systems. We recorded acoustic and high-speed video signals from eight speakers producing the vowel /i/. Each speaker produced four phonation types (breathy, modal, pressed, and creaky) and three pitches (low, normal, and high). Glottal area waveforms were extracted from the high-speed recordings, semi-automatically, on a frame-by-frame basis. The acoustic results indicated that phonation types are differentiated by a number of spectral and noise measures, including H1*-H2* and the harmonics-to-noise ratio. In addition, the different pitch levels were responsible for changes in quality within each phonation type, but such changes were subject to inter-speaker variability. Perceptual results highlight the importance of lower-frequency harmonics in voice quality perception. These studies are critical to our computational modeling efforts, because they help us understand which aspects of the source model are perceptually important. [Work supported in part by NSF and NIH.]
The formation of flow separation vortices inside the glottis during vocal fold closing
Authors: Oren Liran , Khosla Sid , Gutmark Ephraim
Published: October 2011

Our previous work highly suggests that glottal airflow contains certain vortical structures, especially flow separation vortices (FSV) that significantly contribute to acoustics. We have previously identified these FSV directly above the folds but not between the folds. We have now developed a technique that allows us to see both intraglottal velocity fields and the medial wall of the folds during vocal fold closing. In this study, we used the new technique in an excised canine larynx to simultaneously measure intraglottal velocity fields (using high speed particle image velocimetry) and the medial surface of the folds. We observed the formation of FSV inside the glottis. Intraglottal pressures were calculated from measured intraglottal velocity fields using the Navier–Stokes equations. The maximum negative intraglottal pressure on the medial surface of one fold was −6 cm H2O. The circulation of the transglottal flow was highly correlated to the maximum closing speed (MCS) (r=0.98, p<0.001), and had good correlation to the spectral slope (r=-0.8, p<0.01) and HNR (Harmonic to Noise Ratio) (r=0.85, p<0.005). These results substantiate our hypothesis and our new technique is significant because it will allow us to experimentally characterize the intraglottal flow-structure relationship during vocal fold closing.
Acoustic variability affects asymmetry in infant speech discrimination
Authors: Archer Stephanie , Curtin Suzanne L.
Published: October 2011

From birth, infants are capable of discriminating speech sounds that occur cross-linguistically, but by their first year, infants have difficulty discriminating most non-native contrasts (Werker and Tees, 1984). Yet, language experience is not the only factor affecting discrimination. Further research into language-specific perception reveals asymmetries occur in the discrimination of vowels according to perceptual space (Polka and Bohn, 2003) and consonants based on frequency in the input (Anderson etal., 2003). It might be that acoustic variability also causes asymmetries in infant speech perception. Seventy-six 6- and 9-month olds participated in a discrimination task comparing bilabial and velar stop - /l/ onsets to unattested coronal stop - /l/ onsets (e.g. /kla/ - /tla/; /pla/ - /tla/) in both voiced and voiceless conditions. Infants successfully discriminated coronal from bilabial onsets (p<0.05), but not velar (p>0.05), with no effects of age or voicing. Adult productions of the attested clusters were subjected to an acoustic analysis. Of the four tested characteristics, higher standard deviations were found in velar onsets (/kla/ > /pla/: F2 onset and liquid duration; /gla/ > /bla/: F2 slope, VOT, and liquid duration). This suggests that acoustic variability in the input affects infants' speech discrimination.
All representations are not equal: Differential activation of words and phonological variants across accents
Authors: Sumner Meghan , Kataoka Reiko
Published: October 2011

Given the massive variation in natural speech, how listeners recognize words is a central issue for linguistic theory. Listener sensitivity to acoustic fluctuations in speech has provided us with one piece of the puzzle: detailed, specific representations. We provide another piece: differential activation of these representations. We conducted a semantic priming with General American listeners of positively viewed non-rhotic British English accents and negatively viewed non-rhotic New York City accents (slender'THIN; slend-UH(BE)'THIN; slend-UH(NYC)'THIN). Controlling participant experience and self-reported familiarity with both accents, only the words produced in a British accent primed semantically related targets. There is no a priori reason to expect this pattern, as the phonological variant and lexical items are controlled. We suggest that two variants equally experienced in number (raw frequency) are perceived differentially because positive and negative attitudes influence the activation lexical representations. Additionally, by examining particular participant groups from non-rhotic regions, we show an oscillation between effects of raw frequency and listener attitudes. This work (1) provides a broader understanding of factors influencing activation, (2) increases our understanding of how frequency effects are modulated by other factors (subjective listener attitudes), and (3) illuminates the interactive nature of linguistic and social factors in speech perception.
Free classification of dysarthric speech
Authors: Lansford Kaitlin L. , Norton Rebecca E. , Liss Julie M. , Utianski Rene L.
Published: October 2011

The “gold standard” for classification of motor speech disorders, based on the work of Darley etal. [J. Speech Lang. Hear. Res. 12, 246–269 (1969)], presumes that the underlying pathophysiology of each dysarthria subtype is responsible for the associated perceptual features. However, this approach to differential diagnosis suffers significant limitations in that (1) there is substantial overlap of perceptual characteristics associated with the dysarthria classes and (2) characteristics of a given dysarthria vary with severity. It is therefore difficult to perceptually classify speakers reliably without knowledge of the underlying etiology. In the present investigation, 20 speech pathology graduate students grouped 33 speakers with dysarthria into clusters according to the speakers' perceptual similarity using an unconstrained free classification task. The resulting similarity data underwent cluster analysis to identify subsets of perceptually similar speakers. Multidimensional scaling (MDS) revealed the dimensions salient to judgments of perceptual similarity. To interpret the dimensions derived by the MDS, correlation analyses comparing the speakers' spatial distances along the dimensions to obtained segmental and suprasegmental acoustic metrics was conducted. The results of these analyses and how they support the establishment of a non-etiology based, perceptually relevant taxonomy of dysarthric speech will be discussed. [This investigation was supported by 1 F31 DC010093-01A2 awarded to K.L.]
Spectral contrast analyses of vowels processed through a multichannel simulated hearing aid
Authors: Amlani Amyn M. , Bharadwaj Sneha V. , Jivani Shirin , Pogue Jody
Published: October 2011

Hearing aids are designed to separate the audible frequency response into independent channels so that gain can be modulated correspondingly, based on the degree and configuration of the hearing loss. However, increasing the number of channels and employing fast-acting compression might lead to the spectral flattening of speech segments. In this study, we performed spectral contrast analyses on three vowels /i, oo, u/ spoken by two male and two female talkers in a /CVC/ context. The initial consonant consisted of /p, b, t, d, k, g, s, sh/ and the final consonant consisted of /t/. Each /CVC/ word was spoken in the phrase, “I said /CVC/, again” Each phrase was then processed by a simulated hearing aid having 2, 4, 8, and 16 independent channels according to an amplification scheme (linear, compression [fast-fast, slow-slow, fast-slow]). Findings revealed that spectral contrast decreased (1) as the number of channels increased, (2) when a fast attack time was employed, and (3) when the talker was a male. Further, significant spectral flattening and shifts in vowel formants occurred for /i, oo/ compared to /u/. We discuss these findings relative to the speech-intelligibility performance of impaired listeners.
Tracking system developed using an optical fiber-based distributed acoustic sensor
Authors: Efstathopoulos Georgios , Finfer Daniel , Kamil Yousif , Shatalin Sergey , Parker Tom , Farhadiroushan Mahmoud
Published: October 2011

This work will present a system for acoustic source tracking and imaging developed using a novel distributed acoustic sensor. This sensor, the intelligent Distributed Acoustic Sensor (iDAS), uses standard off-the-shelf optical fiber as a transducer to capture the audio-frequency acoustic field (both phase and amplitude) simultaneously along the entire length of a fiber with a spatial resolution of about 0.5 m. iDAS was used to track the bearing of acoustic sources in a reverberant environment, with localization being performed in two stages. First, the received signal was preprocessed for the mitigating effects of calibration errors and multipaths. Then tracking was performed via particle filters. These filters can track multiple acoustical sources and are efficient enough to run in nearly real time. The output from this system is presented in the form of an acoustic camera-type video.