bluuebunny/crossref_metadata_2025 huggingface dataset:
Envelope and temporal fine structure perceptual weights for sentences: Effect of age, hearing loss, and amplification
Authors: Fogerty Daniel , Humes Larry E.
Published: October 2011

The speech signal may be divided into spectral frequency bands, each band containing temporal properties of the envelope and fine structure. This study measured the perceptual weighting strategies for the envelope and fine structure in each of three frequency bands for sentence materials in young normal-hearing listeners, older normal-hearing listeners, aided older hearing-impaired listeners, and noise-matched young normal-hearing listeners. A novel processing method was designed to vary the availability of each acoustic property independently through noisy signal extraction. Thus, the full speech stimulus was presented with noise used to mask the different auditory channels. Perceptual weights were determined by correlating a listener's performance with the signal-to-noise ratio of each acoustic property on a trial-by-trial basis. Preliminary results demonstrate that fine structure perceptual weights remain stable across the four listener groups. However, a different weighting typography was observed between the listener groups for envelope cues. Preliminary results suggest that spectral shaping used to preserve the audibility of the speech stimulus may alter the allocation of perceptual resources. The relative perceptual weighting of envelope cues may also change with age. These effects were largely limited to envelope weights. [Work supported by NIA R01 AG008293.]
Living sound identification system using smartphone for persons with hearing loss
Authors: Sarudate Ashita , Itoh Kenzo
Published: October 2011

In today's advanced information society that overflows with a variety of sounds, people with hearing loss find it very difficult to obtain information on sounds within the home. Although there are many systems to aid handicapped persons, support systems for the hearing impaired do not provide adequate performance. In light of this, the various living sounds identification system for persons with hearing loss was proposed. This system adopts the method of pre-storing signal characteristics so that it can discriminate important living sounds in the home with a high degree of accuracy. In order to construct a real-time processing system, basic signal processing was subjected to frame-by-frame analysis. In addition, so as to be able to detect signals precisely in a noisy environment, signal part detection—based on auditory perception—was adopted. In an experiment in a simulated life space, the system was able to discriminate ten living sounds with almost 100% accuracy. Therefore, the proposed method was very applicable. In addition, the method used by us to present the sound identification results using a mobile phone is considered to be of value. This paper proposed the living sound identification system for use with a smartphone and discussed the utility of this system.
Double multiple signal characterization active large band (D-MUSICAL) for extraction of observable in acoustic tomography
Authors: Grégoire Le Touzé , Barbara Nicolas , Philippe Roux , Jérôme Mars
Published: October 2011

In the ocean, the acoustic propagation is characterized by different arrivals. Each arrival is characterized by its observables: amplitude, arrival time, emission, and reception angles. To achieve ocean acoustic tomography, a method to separate arrivals and extract observables has been developed. Two vertical arrays are used: a source and a receive array. Until now, the double-beamforming algorithm has been used to extract observables. It allows to go from the initial (time–source depth–receiver depth) domain to the beamformed domain (time–source angle–receive angle). This algorithm is efficient thanks to the 3-D domain of estimation but shows resolution limitation in the angle domain. To improve wave separation, we develop and apply a high-resolution algorithm based on MUSIC principle, in the same 3-D domain. The proposed method, called Double-MUSICAL, is the extension on the emission angle of the 2-D MUSIC Acitve Large band method (MUSICAL). Its principle is to separate the recording space onto the two orthogonal signal and noise spaces. We present results on real data recorded in an ultrasonic tank and show that the developed method better detects and separates the first arrivals than double beamforming.
Multiple sources localization with microphone array: A subarray approach
Authors: Tam Kai-Chung , Lau Siu-Kit , Tang Shiu-Keung
Published: October 2011

The problems of location and distance estimations with planar microphone array for multiple point sources are addressed in the present study. Though traditional array processing methods show the reliability in estimation of direction-of arrival (DOA) of the sources, such techniques are incapable to provide source distances which requires for three dimensional (3-D) target coordinates. With planar array processing, azimuth and elevation angles are always obtainable but no longer to extend into more details in 3-D analysis. A new approach to deal with this problem is proposed by estimating multiple sources' DOA with multiple planar sub-arrays. A least square solution based on the estimated DOA of the sources is derived to give the source distances with minimum two sub-arrays in arbitrary microphone configurations. The benefits of the proposed method include a extension of the capability of planar array processing to a higher dimensional analysis. Moreover, its adaptability with any DOA estimation methods makes it convenient to use. The new approach is examined with conventional delay and sum beamforming and MUSIC methods under noisy conditions by both simulations and experiments. [K.C.T. is supported by a studentship from the Hong Kong Polytechnic University.]
Acoustically sensing the presence of methane and carbon dioxide in underground coal mine
Authors: Singh A. , Radhakrishna M.
Published: October 2011

A WSN can be used to continuously sense, monitor, and transmit data to a centralized control station in a under ground coal mine. A fact limiting the possibility is the presence of highly humid condition in UG coal mines. Current sensors cannot work continuously over prolonged period in UG coal mine environment. This paper describes a multi-aspect data fusion approach for acoustical sensors, which make it possible to measure the build up of methane and carbon dioxide in UG coal mine environment. Suggested approach takes time of flight, phase, and attenuation of sonic pulses to determine the build up of methane and carbon dioxide. Suggested approach is more power efficient in comparison to existing sensors. A temperature sensor is used to accommodate change in characteristics of sonic pulses.
Sequential geoacoustic inversion using frequency coherent processing
Authors: Yardim Caglar , Gerstoft Peter , Hodgkiss William S.
Published: October 2011

Insufficient sampling in frequency domain results in range aliasing that affect geoacoustic inversion. Range aliasing and its effects on source localization and environmental parameter inversion are demonstrated on data collected during the MAPEX2000 experiment. Using a source and geoacoustic particle filter mitigates range aliasing caused by using just a few frequencies. Particle filters are sequential Bayesian techniques that provide dynamic estimation of both the geoacoustic parameters and their evolving uncertainties. This approach allows for side lobe suppression due to prior information supplied by the particle filter and hence improved source localization that results in better geoacoustic inversion.
Examining meso-scale variability: Sequential geoacoustic posterior probability estimation for autonomous underwater vehicle (AUV) towed array data
Authors: Dettmer Jan , Holland Charles W. , Dosso Stan E.
Published: October 2011

This paper applies a sequential trans-dimensional Monte Carlo method to data recorded on an autonomous underwater vehicle towed array. The seismo-acoustic recordings are processed in terms of reflection coefficients as a function of frequency and angle, resulting in a sequence of data sets with small seabed footprints. The sequential particle filter applies advanced Markov chain Monte Carlo steps in combination with importance sampling techniques to carry out consecutive geoacoustic inversions along the track. In particular, the typically high information content of reflection-coefficient data is addressed by heuristic interpolating distributions implemented using annealed importance sampling to bridge between adjacent data sets with distant/disjoint high-likelihood regions. The geoacoustic parametrization, including changes in the number of sediment layers, is estimated along the track with trans-dimensional partition modeling. This trans-dimensional approach intrinsically estimates the amount of structure consistent with the data-information content and accounts for the limited knowledge about the parametrization in the geoacoustic posterior probability density estimates. The algorithm is applied to a sequence of data sets collected on the Malta Plateau with small seabed footprints (~10 m), which facilitates examination of meso-scale seabed variability.
Broadband geoacoustic inversion on a horizontal line array
Authors: Tolstoy A.
Published: October 2011

Shallow water bottom properties can be very difficult to estimate and are usually done so by means of vertical line arrays over a range of frequencies. This work will examine the behavior of the minimization broadband processor at low frequencies (25–100 Hz) for a variety of horizontal line arrays at endfire in simulated Shallow Water 2006 environments. Do we ever have enough sensitivity and resolution to estimate any of the bottom properties when using a horizontal array?
Remote-sensing of hydrodynamics in Tokyo Bay and Bakan Strait
Authors: Yamamoto Tokuo
Published: October 2011

In the inertia range, turbulent energy is concentrated in eddies. The eddies interact, dissipate the kinetic energy and reduces the eddy size. The turbulent velocity fields are measured by inverting the acoustic wave attenuation by volume scattering. The eddy viscosity ε is the rate of kinetic energy dissipation of the turbulence. Acoustic wave dissipation is roughly proportional to (velocity intensity fluctuations) ε, 2/3. The Reynolds stresses are also induced by the velocity intensity fluctuations ε, 2/3. The Reynolds stresses take various forms, such as Rip currents, wave set-ups, etc. The acoustic wave volume scattering, transmission loss by volume scattering, turbulent fields, Reynolds stresses, and eddy viscosity fields are inverted from eight point transmission lines in two completely different turbulent flow fields. They are Tokyo Bay (ε ∼ 0.02) and Bakan Strait (ε ∼ 1.2). The Reynolds stresses are in Tokyo Bay (∼33 kPa) and in Bakan Strait (∼1200 kPa). The side by side comparisons of the two data sets reviles the quite complicated differences in their responses to turbulent current field which will be presented in this paper.
Estimates of the water column sound-speed field in three dimensions using a linearized perturbative technique
Authors: Bender Christopher M. , Ballard Megan S. , Wilson Preston S.
Published: October 2011

A perturbative inversion scheme [S. D. Rajan etal., J. Acoust. Soc. Am. 82, 998–1017 (1987)] is applied to estimate the water-column sound-speed field in a three-dimensional volume of the shallow ocean. The input data to the inversion are estimates of modal travel time data calculated from measurements from a distributed network of sources and receivers. The differences in the data are used as the basis for updating, or perturbing, a background model through the inversion algorithm to arrive at a solution. According to this technique, the horizontal plane is divided into a grid of range-independent regions and an estimate of the depth-dependent sound-speed profile within each region is obtained. The linearized perturbative technique is particularly well-suited for the three-dimensional inverse problem as the solution typically obtained in less than 20 iterations. During this talk, the method will be validated using synthetic data which are representative of oceanographic range-dependence in shallow water environments. The effect of the density and distribution of sources and receivers on the resolution of the inversion results will be discussed. [Work supported by ARL:UT IRD.]
A new broadband matched field processor?
Authors: Tolstoy A.
Published: October 2011

It is well known that the use of multiple frequencies helps to reduce uncertainty and to improve resolution for matched field processing. This can be extremely important for the estimation of geoacoustic (bottom) parameters and usually means averaging ambiguity surfaces either incoherently or coherently (if the source spectrum is known) for the linear or minimum variance processors. However, a new approach requires that only the minimum processor values be retained over all the frequencies considered. This presentation will show how this method improves resolution and sidelobe reduction (for the simple incoherent method) both for the linear and for the minimum variance processing. Of course, one never gets something for nothing. This improvement comes at the cost of using only those frequencies which show high-matched field values, usually the low frequencies (25–75 Hz), for geoacoustic inversion. If some frequency component is strongly degraded, then the method will not work. The behavior of this new processor (to be called the “minimization broadband processor”) is demonstrated on simulated shallow water data.
Changes in on-stage and in-house listening conditions with adjustments to over-stage canopy height and tilt: Field investigations in two concert halls
Authors: Dohn Bill
Published: October 2011

On-stage and in-house measurements were made in two concert halls with adjustable over-stage canopies to assist with future designs and verify recommended canopy settings for facility users. Results of measurements with varying canopy heights and tilt angles will be shared, along with descriptions of the two (very different) canopy designs and the two (very similar) concert hall designs. User preferences for canopy settings in both halls will also be discussed.
Working with student musicians on developing standard settings for concert stages with adjustable acoustics
Authors: Kahn David W.
Published: October 2011

The acoustician's work on concert stages with adjustable acoustics usually includes a so called “tuning” process. This tuning process is unique and different for students attending music school as compared with, for example, a professional orchestra. We describe a tuning process, working with student musicians, at a new concert hall in Texas. Adjustable elements include movable ceiling elements, movable doors on the upper wall surfaces surrounding the concert platform, and retractable curtains that cover both the upper and lower wall surfaces surrounding the platform.
Implications of vocalizations during giant panda breeding
Authors: Keating Jennifer L. , Brown Tracey K. , Caine Nancy G. , Bowles Ann E. , Charlton Benjamin D. , Swaisgood Ronald R.
Published: October 2011

Silent most of the year, giant pandas (Ailuropoda melanoleuca) engage in sustained and diverse vocal behavior during their brief annual reproductive window. These vocalizations suggest a complex communication repertoire and play an important role in facilitation of mating. To increase our understanding of their vocal communication, we analyzed male and female vocalizations during breeding interactions. Digital audio recordings were collected during breeding seasons of 2008, 2009, and 2011 at zoological facilities in San Diego, California, and China, and were processed using Soundtrack Pro 2.0 and Raven Pro 1.4 software. Seven types of vocalizations were identified (“bark,” “moan,” “growl,” “squeal,” “chirp,” “bleat,” and “copulation call”). We compared vocalizations from 32 confirmed copulations and 37 non-copulatory breeding sessions of 31 individuals (24 females, 7 males). They revealed differences in vocalizations produced before, during, and after intromission. During intromission, the male produces a copulation call that distinguishes successful breeding encounters from unsuccessful pairings acoustically. This call has not been described previously. The results shed light on the motivational and functional significance of panda vocalizations during mating encounters. They can be used in captive breeding programs to promote reproduction in this endangered species. [Work supported by ZSSD, CSUSM, and CCRCGP.]
Taking ASA/ANSI Standards to ISO
Authors: Bahtiarian Michael
Published: October 2011

The author served as the chairman of the working group that completed the first voluntary consensus standard for measurement of underwater noise from ships (ANSI/ASA S12.64-2009/Part 1, Quantities and Procedures for Description and Measurement of Underwater Sound from Ships, Part 1; General Requirements). Within the last year, an effort to bring this standard to the international arena has started. The S12.64 standard is an entirely new document without any prior written standards to use as a template, however, the methodology was previously used by the U.S. Navy and NOAA. This session will address the process of organizing and developing a new ASA/ANSI standard and then taking such a standard to International Standards Organization (ISO). It will address organizational challenges of working with experts from industry, government, and academia, both from the United States. and overseas. It will cover issues related to planning and executing physical and web enabled meetings. Lastly, it will address the similarities and differences of the standards development process under ISO.
Architectures of sound: A new material in socio-spatial formation
Authors: Beretta Lorenzo
Published: October 2011

In architecture, sound is only haphazardly used as an accessory to pre-defined, pre-constituted architectural blocks. The paper examines those sonic properties that are able to alter cognition, behavior, and human interaction leading toward the definition of sound as a material. Over the centuries, buildings have changed their constituent materials ranging from ceramic bricks to concrete, from wood to glass, from light to vapor. Sound is not just a floating, ephemeral presence that exists over time but a solid material that defines and changes the space around us. As a result, spaces are not just lived through their solid elements but through their transposition in the mind of each user. It is for this that by acting on the properties of sound, it is possible to redefine the way people navigate and inhabit spaces. Sound should no longer be used as an accessory or content of architectural spaces but, instead, as a material able to give form, volume, and shape to what is generally referred to as architecture—an inhabitable entity of social relevance in the urban context.
Three-dimensional dynamical equations of interacting bubbles
Authors: Kurihara Eru
Published: October 2011

It is known that bubble cavitation plays important role in kidney stone fragmentation in the shock wave lithotripsy and other medical applications of shock wave. The behavior of such bubbles, however, considerably complicated because of its nonlinearity and mutual interactions among the bubbles. For weakly nonlinear oscillations, dynamics of interacting bubble can be approximately expressed by the method of multi-pole expansion with spherical harmonics. In the previous study, the author derived a set of dynamical equations of two interacting aspherical bubbles with Lagrangian mechanics. The axisymmetrical system of two interacting bubbles can be described in two-dimensional coordinate system, and then shape oscillation of the bubbles is expressed with Legendre polynomials. The bubble behavior described by the derived equations qualitatively agreed with experimental results by high-speed photographs. In the dynamics of three or more bubbles, however, the behavior of bubbles is essentially three dimensional, and thus the system of these bubbles should be represented by three-dimensional spherical harmonics (associated Legendre functions). In this study dynamical equations for three interacting aspherical bubbles are derived by multi-pole expansion in the framework of Lagrangian formalism. [Work supported by Grants-in-Aid for Scientific Research 23760142 and a research grant from The Mazda Foundation.]
A compact Terfenol-D vector projector
Authors: Slaughter Julie C.
Published: October 2011

In active sonar applications, it is desirable to have a directional, steerable sound source to accelerate the target localization process. A narrow cardioid beam pattern with approximately 78° beam width and rear side lobes 25 dB down can be generated by combining monopole, dipole, and quadrupole beam patterns with appropriate amplitudes and phases. A cylindrical array composed of eight Terfenol-D Tonpilz transducers arranged in two rings with each transducer pointing radially outward from a common center mass has been developed to generate narrow cardioid beam patterns. Focus during the design process was on minimizing size, maximizing the bandwidth, and maximizing the duty cycle. The diameter of the source is 0.19 wavelengths at the lowest operating frequency and 0.76 wavelengths at the highest operating frequency. Bandwidth of the source is greater than two octaves. Lumped parameter modeling and finite element modeling are used to demonstrate the monopole, dipole, quadrupole, and narrow cardioid beam patterns. The effects of extraneous vibration modes on the beam patterns and sound output are discussed. Maximum continuous wave sound output and duty cycle at maximum output are estimated from thermal models and test data from a single Tonpilz element. [Work supported by the Office of Naval Research.]
Numerical simulation of the sound of organ pipes
Authors: Steppat Michael
Published: October 2011

Numerical simulations of musical instrument sounds are useful in studies, when questions about the influence of physical parameters stand in first place. Differential equations can be used to solve the given problem. The results given as a discrete sequence can be stored in audiofiles and made audible. In the current project, the influence of the attack of organ pipes with respect to the reverb time of the acoustical environment in the church building is studied by different numerical calculation methods such as finite element method and computational fluid dynamics. The resulting sound pressure and the velocity in a vortex street of the jet emerging from a pipe can be calculated at different listening positions. The computation allows also visualization of the jet in a three dimensional view. In the tests, the same parameters used by organ builders to voice organs are used for the simulation. Main point is the strength of the resulting starting transient and its influence to the acoustical environment. The calculation process which includes the synthesizing and adding a reverb and the results with different voicing parameters will be presented.
Video analysis and modeling of the kalimba tine
Authors: Ludwigsen Daniel O.
Published: October 2011

The kalimba, like the more traditional mbira, uses plucked metal tines mounted to a wood resonator. The mounting is similar to a three point flexural test and provides an initial strain to part of the tine. The plucked end of the tine is free once released, but modeling the other end is less obvious. The motion of the longest tine of the treble kalimba (B3, 247 Hz) was captured via high speed video (1200 f.p.s.). Analysis of tine displacement informed the boundary conditions of an Euler–Bernoulli model for this thin beam vibration, which in turn predicted mode shapes and frequencies. The two lowest mode frequencies can be compared to prominent features in the spectrogram of recorded tones. Rapidly decaying harmonic content observed in the spectrogram is not suggested by this simple model; a more sophisticated approach is required to fully understand the behavior of the kalimba tine.
Validation of a descriptor, the “sum function", related to “quality” and derived from the input impedance of wind instruments
Authors: Causse R. E. , Eveno P. , Kieffer B. , Gilbert J. , Dalmont J. P. , Petiot J. F.
Published: October 2011

The quality of a musical instrument embraces many aspects such as tuning, ease of play, tone, etc. This study aims to validate the use of the sum function (SF) proposed by Wogram from the measurement of input impedance as a descriptor of quality. This work is part of a wider project, PAFI (Aid Platform for Instrumental Making), supported by the French National Agency of Research. To validate the choice of the SF, we created a family of trumpets made from a basic instrument for which the leadpipe will be slightly modified for each model. The SF was calculated for a range of selected frequencies from the measurement of the input impedance of these different trumpets. The next step was to ask musicians experts to play these instruments, to measure the playing frequencies, and to note their feedback about quality. These tests were supplemented by comparative tests carried out this time with the help of a robotic artificial mouth. The final step involved was to try to identify correlations between the SF and the results of various tests and propose correction factors to be made to the formula of the SF, related to the nuance or range for example.
Modal response and sound radiation from a hammered dulcimer
Authors: Christensen Benjamin Y. , Gee Kent L. , Anderson Brian E. , Wall Alan T.
Published: October 2011

The sound radiation of the hammered dulcimer has been investigated. The dulcimer studied is a 16/15 fixed-top instrument of typical size with Baltic birch (laminated plywood) soundboard and back. To determine the instrument body resonances, the dulcimer was driven at the treble and bass bridges with a shaker. Accelerometers were used to obtain the resonance frequencies of the soundboard and back, and a microphone was placed inside the instrument to obtain the cavity resonances. The individual resonance peaks found were further investigated using scanning laser Doppler vibrometry and near-field acoustical holography. Preliminary results show that there is little modal response of the instrument at the fundamental frequencies of the lowest notes of the dulcimer. In addition, the vibration coupling to the back plate through the internal bracing causes it to serve as a second soundboard. Lastly, the holography results indicate significant radiation from the sound holes at some frequencies, which may contradict the commonly held notion that the dulcimer sound holes are largely decorative.
Large-Eddy-Simulations of over-expanded supersonic jet noise for launcher applications
Authors: Dargaud Jean-Baptiste , Troyes Julien , Vuillot François
Published: October 2011

During the lift-off phase of a space launcher, powerful rocket motors generate harsh acoustic environment on the launch pad. Following the blast waves created at ignition, jet noise is a major contributor to the acoustic loads received by the launcher and its payload. This paper describes recent simulations performed at ONERA to compute the noise emitted by solid rocket motors at lift-off conditions. Far-field noise prediction is achieved by associating a LES solution of the jet flow with an acoustics surface integral method. The computations are carried out with in-house codes CEDRE for the LES solution and KIM for Ffowcs Williams and Hawkings porous surface integration method. This work has been conducted in the framework of the cooperation on launcher acoustics between CNES (French National Space Agency) and JAXA (Japan Aerospace Exploration Agency) involving the French AEID research group. The test case is that of a reduced scale solid rocket motor, fired vertically and has been provided by JAXA. Computations were run for varied numerical conditions, and the final paper will detail results and discuss comparisons with experimental acoustic measurements.
Photoacoustic detection of thin layers of explosives
Authors: Marcus Logan , Raspet Richard , Aranchuk Slava
Published: October 2011

Explosives may be identified by their infrared absorption spectra. Photoacoustic measurements, which measure the acoustic output generated by the explosive sample absorbing radiation from a modulated laser beam, are efficient at detecting and identifying explosives. We are investigating the standoff photoacoustic detection of surface traces of explosives on a variety of substrates using a laser Doppler vibrometer. This talk describes the detailed modeling of the measurement. A modulated infrared Gaussian profile laser source is absorbed in the layer of explosive and in the substrate. The heat absorbed raises the temperature of the explosive and the substrate leading to expansion and surface displacement. The heat absorbed by the gas adjacent to the solid generates an outward propagating acoustic wave. A laser Doppler vibrometer measures the phase shift due to the surface displacement and the compression and refraction of the acoustic wave. A cylindrically symmetric calculation of each physical step will be described. Calculations demonstrating how substrate parameters such as coefficient of thermal expansion, IR absorptivity, and Poisson's ratio effect the detection threshold of the measurements will be presented.
Sound propagation in a pipe with dynamically rough boundary
Authors: Romanova A. , Horoshenkov K. V. , Tait S. J.
Published: October 2011

The surface pattern of the water flow in partly filled circular pipe contains information on some key characteristics which are important for a better understanding of the hydraulic energy losses and turbulent processes. Surface pattern variation is a dynamic and nonstationary process which is difficult to measure directly. In this sense, airborne sound waves provide an attractive statistical mean which describes the apparent boundary roughness, its spatial correlation function, and frequency spectrum. These parameters can then be linked to key hydraulic characteristics of the flow. This work presents new experimental setup, which is used to study these characteristics under controlled laboratory conditions and allows for simultaneous measurements of the acoustic field in the pipe and water surface roughness. The acoustic technique makes use of Gaussian pulses which are transmitted in air above the turbulent flow of water over a carefully instrumented section and recorded on an intensity probe. The results obtained for a range of flow regimes illustrate that it is possible to relate unambiguously the variation in the recorded acoustic field to a short-term variance in the water surface roughness and its spectrum. A suitable theoretical foundation based on small perturbation theory is proposed to interpret the obtained data.
Utilization of an acoustic tomography array as a large sonic anemometer/thermometer
Authors: Vecherin Sergey N. , Ostashev Vladimir E. , Wilson D. Keith
Published: October 2011

The temperature, wind velocity, and vertical and horizontal kinematic heat fluxes are important characteristics of the atmospheric surface layer. Point measurements of these meteorological parameters are often not representative due to their horizontal variations. For remote sensing of the area-averaged values of these parameters, we suggest using the acoustic tomography array at the Boulder Atmospheric Observatory (BAO). In this approach, the tomography array (with the horizontal size of 80 m by 80 m) is used, in essence, as a large sonic anemometer/thermometer for measurements of the area-averaged, instantaneous values of temperature and wind velocity. The area-averaged horizontal heat flux is then calculated from a time series of the area-averaged temperature and wind velocity. Feasibility of this approach is studied in numerical simulations of the BAO tomography array with the use of LES fields of temperature and wind velocity. The results obtained show that the area-averaged values of temperature, wind velocity, and horizontal heat flux are reliably reconstructed. Numerical analysis of the LES fields indicates that the area-averaged vertical heat flux might be inferred from the horizontal flux. Preliminary experimental results obtained with the BAO acoustic tomography array show that this remote sensing technique is feasible.
Computational model for the dynamic stabilization of the Rayleigh-Bénard instability in rectangular enclosures
Authors: Carbo Randy M. , Smith Robert W. M. , Poese Matthew E.
Published: October 2011

When fluid within a container is heated from the bottom, onset of convection occurs when Rayleigh number, gβΔTℓ3νχ, exceeds some critical value. If an acoustic field is imposed on the fluid in the container, the critical Rayleigh number is a strong function of the frequency and amplitude of that acoustic field as noted by Swift and Backhaus [J. Acoust. Soc. Am. 126(5), 2009]. Results will be reported for a linear model constructed to predict the modified critical Rayleigh number, based on a full field solution of the hydrodynamic equations using the approach of Gelfgat [J. Comp. Phys. 156, 1999]. The spatial portion of the differential equations was solved using the Galerkin method, and the dynamic stability was determined using Floquet analysis. One of the benefits of the approach compared to the averaging methods used by Gershuni and Lyubimov, [Thermal Vibration Convection (Wiley, New York, 1998)] is that the parametric stability boundary can also be recovered. This study includes a variety of container aspect ratios, boundary conditions, and Rayleigh numbers ranging from 103 to 108. [Work supported by the Office of Naval Research and ARL Exploratory and Foundational Research Program.]
Acoustics cavitation in microfluidics for sonoluminescence and sonochemistry
Authors: Ohl S. W. , Tandiono T. , Ow D. , Klaseboer E. , Wong V. , Ohl C. D.
Published: October 2011

Strong ultrasound is applied to a microfluidic channel to generate nonlinear surface waves which entrap bubbles at the gas–liquid interface to form oscillating bubbles. The ultrasound is generated by the piezoelectric transducer on the side of polydimethylsiloxane microchannel. The microchannel is attached to a glass slide through plasma bonding, while the transducer is glued by epoxy for strong coupling. The high speed photography shows that continuous cavitation clusters are formed within the microchannel as gas is injected. As they collapse rapidly, they are able to produce very intense concentration of energy that is able to emit light. This phenomenon is known as sonoluminescence. Previously, sonoluminescence is achieved via a single bubble or multiple bubbles in a bulk liquid. The authors report a realization of sonoluminescence in a microfluidic device. The same oscillating bubbles can also be used as micro-labs. They can trigger chemical reactions that require high temperature and pressure. We achieve the formation of OH radicals in a lab-on-a-chip device. In conclusion, nonequilibrium microbubbles can be induced in a microfluidic system. They oscillate and collapse, and in the process provide a source of energy concentration for the emission of light and the activation of chemical reactions.
Psychophysical weights estimated for interaural cues in envelope slopes of amplitude-modulated waveforms
Authors: Hsieh I-Hui , Saberi Kourosh
Published: October 2011

Current binaural theory contends that the auditory system encodes interaural delays in highpass-filtered complex sounds by phase locking to their slowly modulating envelopes. Spectrotemporal analysis of interaurally time-delayed highpass waveforms reveals the presence of a concomitant interaural level cue which could contribute to lateralization judgments. The current study systematically investigated the relative contribution of time and concomitant level cues carried by positive and negative envelope slopes of modified sinusoidally amplitude-modulated (SAM) high-frequency carriers. Psychophysical thresholds and observer decision weights were measured independently for the positive and negative modulation slopes of the acoustic signal. Decision weights were also measured to determine whether or not interaural delays are uniformly weighted at different temporal cycles of a SAM waveform. We found that lateralization of interaurally delayed SAM waveforms is influenced equally by ITDs in the rise and decay envelope slopes, and not by concomitant ILD cues, and that ITD cues are more heavily weighted in the initial few cycles of the SAM envelope.
Auditory training effects on auditory steady state responses
Authors: Hinojosa Vidal I. , Jin Su-Hyun
Published: October 2011

Human auditory steady-state responses (ASSRs) have been linked to recognition scores in normal hearing and hearing impaired adults (Dimitrijevic etal. 2004). By taking this into account ASSRs should improve over time when speech perception improves. Computerized aural rehabilitation programs such as the LACE program by Neurotone have claims of being able to improve speech in noise perception, such as a 2.2 dB improvement using the QuickSin Test as an outcome measure. The main purpose of this study is to answer the question “Can improvements in speech perception be tracked by electrophysiological methods such as ASSR?”. For this study, ASSRs and HINT scores will be tracked before and after auditory training in two groups of subjects. The first group will be hearing impaired people who have had hearing aid experience before. The second group consists of hearing impaired people who have just been fit with hearing aids for the first time. ASSR stimulus will be modeled after speech as in the Dimitrijevic 2004 study. By tracking these improvements in speech perception objectively using the ASSR and subjectively using HINT scores, the study will hopefully add validation to objectively testing speech perception using ASSR techniques.
A modified model for predicting breathiness judgments using partial and noise loudness measures
Authors: Skowronski Mark D. , Shrivastav Rahul , Eddins David A.
Published: October 2011

Breathiness is one of three classifications, along with roughness and strain, used to characterize disordered voice quality. Breathiness has been measured using a magnitude estimation task and modeled with a power function of perceptual measures [Shrivastav etal., J. Acoust. Soc. Am. 129, 1605–1615 (2011)]. The experiment was repeated using a matching task which required listeners to match the perceived breathiness of a vowel to a sawtooth-in-noise comparison by adjusting the signal-to-noise ratio of the comparison. The matching task provides a more accurate measure of perception and can potentially accommodate all three dimensions of dysphonic voice quality (breathiness, roughness, and strain) in a single experimental paradigm. Breathiness was modeled with a power function of noise-to-partial loudness ratio. Breathiness judgments (in decibels) varied over about 4 orders of magnitude and exhibited a linear relationship with the log ratio of noise-to-partial loudness. This data was modeled with a logistic function which accounted for breathiness saturation at low levels of noise loudness. Accuracy of the models is discussed as well as sources of variation in the experiment (talker, listener) and strategies for mitigating their effects on modeling accuracy.
Comparison of a 3 dimension model versus a 2 dimension-axisymmetric finite element model of an occluded ear canal
Authors: Viallet Guilhem , Sgard Pr Franck , Laville Frédéric
Published: October 2011

Due to low cost and simplicity, earplugs are a widespread solution to prevent the problem of hearing loss in the workplace environment. In practice, earplugs often perceived are being uncomfortable and/or do not always perform as desired. The attenuation, based on a laboratory measurement, is often overestimated compared to in situ measurements. The use of a model of an occluded ear canal can help to build an individual measurement system of the attenuation and to develop an earplug with optimized attenuation. It is established that the unoccluded auditory canal can be approximated by a cylinder to predict the interior pressure field up to 6 kHz. A remaining question is whether this approximation holds true for an occluded ear. First, a simplified 2-D-axisymmetric finite element model of an ear canal coupled to a cylindrical earplug is developed. Special emphasis is on the coupling between the earplug and the lateral walls of the auditory canal. Second, a 3-D model of a real ear canal coupled to a cylindrical earplug is developed to examine the limits of the simplified model. Several assumptions about the deformation of the ear canal/earplug system are tested when comparing the sound attenuation provided by both models.
Legal standards for the admissibility of expert testimony: Implications of the 2009 National Research Council report on forensic sciences
Authors: Thompson William C.
Published: October 2011

This talk will review formal legal standards for the admissibility of expert testimony in the United States, focusing on the Daubert standard that is applied in Federal courts and the Frye standard that is applied in several major states (including California and New York). It will also discuss the National Research Council's 2009 critique of judicial “gatekeeping,” particularly the NRC's stunning claim that courts have violated their own purported standards by allowing forensic scientists to present scientifically dubious testimony based on inadequately validated methods. It will conclude by providing suggestions for researchers in emerging areas of forensic inquiry, like forensic acoustical science, who contemplate testifying in court.
An introduction to forensic gunshot acoustics
Authors: Beck Steven D. , Nakasone Hirotaka , Marr Kenneth W.
Published: October 2011

Due to the proliferation of audio recording devices in the military, law enforcement, and the civilian community, there has been an increase in the number of recorded gunshot sounds submitted for forensic analysis. A gunshot sound is composed of one or more discrete acoustic transient events. The two primary acoustic events are the muzzle blast (bang) and the ballistic shockwave (crack). The acoustic event characteristics depend on their source generating mechanisms and vary according to the firearm make, model, barrel length, and the specific ammunition characteristics. Forensic gunshot analysis deals with a single recorded shot lasting for a fraction of a second. These acoustic events are usually high intensity, often up to 160 dB SPL, are highly directional, and are often recorded in high distortion environments. Forensic gunshot analysis must take into account variations in the source generation characteristics and the sources of distortion for these recorded acoustic events in order to answer these fundamental forensic questions: Is this event a gunshot? Are two events from the same firearm? Who fired first? To illustrate the complex nature of the analysis, we present the gunshot data collected in a pristine controlled environment and the data collected in a forensic environment.
Intra- and inter-speaker variability in duration and spectral properties of English /s/
Authors: Kavanagh Colleen
Published: October 2011

This study investigates the speaker-specificity of acoustic characteristics of the English fricative /s/ and contributes background population statistics for use in forensic speaker comparison work. The intra- and inter-speaker variability in duration and spectral properties of /s/ was investigated in data from 30 young adult male speakers of Cambridge and Leeds English. Read speech was used in the present study to allow direct comparison across speakers. Segment duration was normalized for speaking rate. Spectra were filtered at 4 kHz in order to explore speaker discrimination performance at settings mimicking the bandpass filter effect of telephone transmission. Additional filters were applied at 8, 16, and 22.05 kHz to investigate discrimination with data from various frequency ranges. Spectral measures were calculated from a 40-ms window centered on the midpoint of each token. Although mean values display relatively little inter-speaker variation, the individuals at the extreme high and low ends of the distributions may be the best discriminated, particularly those at the extremes on more than one parameter. Discriminant analyses were conducted to determine the most speaker-specific predictors; relative performance was compared across the four filter conditions. The discriminatory ability of these parameters will also be presented using a likelihood ratio framework.
Effects of consonant context on vowel formant contours in spontaneous and read speech
Authors: Kiefte Michael , Nearey Terrance M.
Published: October 2011

A database of recordings from 163 speakers from Nova Scotia, Canada was collected with the aim of comparing formant contours between spontaneous and read speech. In the reading task, participants were asked to produce a number of real and nonsense words spanning the inventory of vowels in this dialect in a variety of consonant contexts. In the second part of the reading task, speakers were asked to read 20 sentences from the TIMIT database. These latter recordings were used to assist in the training of the force-alignment system which used to segment the recordings into phonemes from a text transcript. In addition to the reading task, speakers also provided a monologue on a topic of their choice. These recordings were screened for disfluencies, noise, and disruptions, manually segmented into breathgroups, and then transcribed. Stressed vowels in /CVC/ contexts, where C corresponds to plosives, were sampled from both the read and spontaneous speech. Formants were tracked and measured automatically, and an analysis similar to that of Broad and Clermont [J. Acoust. Soc. Am., 81, 155–165 1987] was performed in which consonant effects on formant transitions are treated as additive effects. [Work supported by SSHRC.]
Examining the voice bar
Authors: Fulop Sean A. , Disner Sandra
Published: October 2011

In a spectrogram of a human vowel sound, it is possible to observe the formant resonances which define the vowel auditorily. It is usually also possible to observe an emphasized frequency below F1, which has often been called the voice bar. Although recognition of the voice bar dates back to 19th century phonetics, it has never been the subject of a specific investigation. As a result, the nature and origin of the voice bar remain mysterious. Recent work on voice source synthesis [C. d'Alessandro etal., eNTERFACE 2005 Proceedings, pp. 52–61] has explained the appearance of an emphasized frequency in the neighborhood of 200 Hz–simply, it results from the frequency peak of the radiated source spectrum. Yet many speech scientists continue to ignore the voice bar, even to the point of denying its reality. Measurements of the voice bar in a number of different speakers and languages will be clearly shown in this paper, using reassigned spectrograms and linear prediction spectral estimates. The voice bar has confused phoneticians for over a century, and has often confounded efforts to measure F1. The proper recognition of the voice bar can begin with this preliminary study.