bluuebunny/crossref_metadata_2025 huggingface dataset:
Geoacoustic inversion using surface ship sources and HLA beam data
Authors: Koch Robert A. , Knobles David P. , Neilsen Tracianne B. , Haire Mark S.
Published: February 2014

The focus of this research is the synthesis of spatial and temporal processing to perform simultaneous inversion for environmental parameters and to localize moving broadband sources in shallow water environments. The processed data are a sample of broadband cross spectra from either individual phones or subaperture beams for one or more FFT sequences. An inversion using simulated annealing to estimate both environmental and source parameters is performed using a cost function that measures the correlation between the sample of data cross spectra and corresponding modeled cross spectra. Thus, the data are processed coherently in frequency and time. For each iteration of the simulated annealing search, the cost function is computed on a grid of initial source positions for a specified source speed and course and for specified seabed geoacoustic parameters. Horizontal line array data collected in February 1998 in the Gulf of Mexico using SWAMI are processed, and the results for the geoacoustic parameters from the inversion using the exercise ship as the source are compared with previous results obtained from light bulb implosions. [Work supported by ONR.]
Chirp sonar inversion of geoacoustic properties of finely layered seabed
Authors: Turgut Altan , Wolf Stephen N.
Published: February 2014

Chirp sonar reflection amplitude and phase data can be used for real-time inversion of sediment properties such as density, porosity, and sound-speed profiles by assuming homogeneous layering within the seabed [Turgut and Wolf, J. Acoust. Soc. Am. (1996)]. In case of fine layering, inversion results might be erroneous if the chirp pulse does not have enough bandwidth to resolve the layers. An alternative method, based on adjoint-state inversion techniques, is used to estimate sediment impedance fluctuations at wavelength scales. Both methods are tested using real and simulated reflection amplitude data. Simulated reflection amplitude data were obtained by using a poro-viscoacoustic sediment model based on Biot’s theory. Accuracy of the inversion can be improved when both methods are used in a complimentary manner. Limits of these two inversion techniques are also discussed and several recommendations are made to improve the data collection methods. [Work supported by ONR.]
Geoacoustic parameters inversion by an annealing-genetic algorithm
Authors: Lin Ying-Tsong , Chuang Yuan-Ming , Chen Chifang
Published: February 2014

A matched-filed processing (MFP) with the annealing genetic algorithm is used to invert the geoacoustic parameters by matching acoustic field data with numerical replica field. While considering the MFP as an optimization problem, it is unavoidable that a huge amount of replica field will be needed for matching with the measured field data. Furthermore, due to complicated geoacoustic parameters, the inversion becomes difficult to deal with. The simulated annealing algorithm has the property of better local search ability, while the genetic algorithm might jump over local optimal and have the better ability in global search. By merging simulated annealing algorithm into genetic algorithm, an annealing genetic algorithm is developed for efficiency and accuracy. Several scenarios of ocean model are used as test cases. The geoacoustic parameters are accurately estimated. [Work supported by National Science Council of Republic of China.]
Acoustics in medical ultrasound beamformation
Authors: Thomenius Kai E.
Published: February 2014

Today’s medical ultrasound scanners continue to push the envelope with efforts to improve imaging performance. Increasingly this demands more sophisticated application of acoustics in the design of the beamformers for these scanners. In this paper, limitations of present designs are reviewed by the use of acoustic field simulations. These design limitations include spatial sampling of the transducer aperture and the contributions of two-dimensional apertures and the impact of variations in the speed of sound in the various tissues. The role of beamformation in defining the likely scatterer populations that will be included in any beam will be discussed. The introduction of harmonic imaging into medical ultrasound has changed the landscape significantly especially in fields such as echocardiography. This has generated the need to understand the harmonic generation process of the propagating acoustic wave. Comparisons of the acoustic beams due to the fundamental frequency spectrum and the harmonic spectra are discussed along with means by which the scanners take advantage of them. From this base one can begin to identify areas of further improvement and speculate on their likely impact on the diagnostic process.
Ultrasonic imaging with aberration correction
Authors: Waag Robert C.
Published: February 2014

Measurement and simulation of ultrasonic wave front distortion produced by propagation through abdominal wall, breast, and chest wall show that the distortion varies with tissue type, scattering contributes significantly to distortion, and arrival time and energy level fluctuations can be large compared to precision in current ultrasonic imaging instrument beamformers. Investigation of focus degradation and compensation by measuring the transmit focus with a hydrophone and by reconstructing the time history of a virtual point source in each case without compensation and with various compensation techniques indicate that, while time-shift compensation in the aperture can improve the focus significantly, techniques capable of compensating amplitude and shape changes in waveforms can improve the focus even more. Pulse-echo studies of focusing that use estimates of aberration from random scattering show that both low transmit f-numbers and transmit compensation are important for accurate estimation of aberration. B-scan images formed through a water path, through tissue-mimicking aberration, and through the aberration using aberration correction demonstrate that images can be substantially improved by adaptive beamformation that employs estimation and correction of aberration on both transmit and receive.
Integrated matrix arrays
Authors: Erikson Ken , Stockwell Jason , McPhie Robert
Published: February 2014

Improved image quality requires the use of matrix (n×m) arrays with a thousand or more elements. As element numbers increase and their dimensions grow smaller, limitations to present fabrication technologies arise. Cost, ergonomics, produceability, and reliability are important issues. Signal loss due to the capacitance of interconnecting coax cables becomes a fundamental problem. Connecting an integrated circuit directly to the array elements alleviates all these problems. Each unit cell of such a custom transmitter/receiver integrated circuit (TRIC) may have high voltage switches for transmitting; a preamplifier which minimizes signal loss due to capacitance in coax cables and a multiplexer to send the array signals over fewer wires. Additional signal processing and beam forming may also be included. Issues with currently available arrays are reviewed. The new technology for direct connection of arrays to IC’s is described. The paper concludes with speculation about future possibilities of this approach.
Scanned synthesis
Authors: Verplank Bill , Mathews Max , Shaw Rob
Published: February 2014

‘‘Scanned synthesis’’ is done by scanning the slowly varying shape of an object and converting this shape to samples of a sound wave. The shape of the object is determined by the dynamic reactions of the object to forces applied by the performer. These forces vary at ‘‘haptic’’ rates (0–20 Hz). If the scanning path is closed, the sound wave is quasiperiodic and a fundamental pitch is perceived at the cycling frequency (20 Hz–20 kHz). Scanned synthesis provides direct dynamic control by the performer over the timbre of sounds as they are produced. The object can be real or simulated. With finite-element models, we have simulated the one-dimensional wave equation for a generalized slowly vibrating string. Timbres generated by manipulating the string at haptic rates are perceived as having a very pleasing live quality caused by the continually changing spectrum. To achieve additional richness, the performer can change the properties of the string in time and over the length of the string. [Work supported by Interval Research and CCRMA.]
Introduction to pulsar synthesis
Authors: Roads Curtis
Published: February 2014

Pulsar synthesis (PS) is a method of electronic music synthesis based on the generation of trains of sonic particles. PS can produce either rhythms or tones as it criss-crosses perceptual time spans. The basic method generates sounds similar to vintage electronic music sonorities, with several important enhancements. The advanced method combines multiple pulsar trains and convolution with sampled sounds. Together with Alberto de Campo, the author has designed a program for pulsar synthesis called PulsarGenerator. Applications of pulsar synthesis in compositions by the author are noted.
Noise analysis of a new flight procedure for regional jets at Logan Airport
Authors: Timmerman Nancy S.
Published: February 2014

During the second half of 2000, the FAA Boston Tower tested a new approach path for regional jets to Logan International Airport’s Runway 4L. The path was the same as that currently in use for turboprops. In this study, complaints, flight tracks, and noise levels were compared between the test period and previous time periods. Noise levels were reviewed at four of Massport’s permanent noise monitors for three types of regional jet and one type of turboprop. Maximum levels of regional jets were slightly higher than the turboprop investigated for comparable operations.
The effect of atmospheric gradients on aircraft noise contours
Authors: Plotkin Kenneth J.
Published: February 2014

Calculation of airport noise contours generally ignores atmospheric gradients. This is usually justified by the assumption that any effects will average out in the long run. An initial study has been conducted to evaluate this assumption. Noise contours have been computed for a single takeoff and landing of an F-16C under various atmospheric conditions. Atmospheric conditions included clear daytime, cloudy daytime, and nighttime temperature profiles, with winds at several speeds and directions. The heuristic propagation algorithms of L’Esperance et al. [AIAA Pap. 93-4405 (1993)] were incorporated into Wyle Laboratories NMSIM single-event noise simulation model. Calculated SEL footprints exhibited substantial changes in shape, particularly in sideline regions under crosswind conditions. Some trends of increased area under windy conditions were seen. Application of this analysis to a long-term set of actual weather data would validate (or invalidate) the ‘‘it all averages out’’ assumption. [Work supported by USAF Armstrong Laboratory. Atmospheric profiles and propagation routines provided by Andre L’Esperance.]
Noise control of a compressor station
Authors: Alabi Halimat I. , Krutz Tim , Jennische Mats , Goh Chiew Seng , Dominik Benjamin , Stricula Justin
Published: February 2014

The noise emissions from a compressor station, part of a wastewater treatment plant, were investigated. The noise was a source of annoyance to a nearby residential subdivision. Initial measurements made on the site revealed salient tonal components. Further measurements suggested that the two main sources were leakage around the pipe intakes, and flow noise from the exhaust mufflers. Modifications were made to both sources in order to mitigate the problem. The leaks were sealed using dense caulking materials, and spoilers were inserted in the mufflers. The impact of these noise control solutions was measured, and the benefits to the surrounding community were assessed.
Unusual experiences of two acoustical consultants
Authors: Peres Marco , Larson David
Published: February 2014

Some unusual projects worked on by two acoustical consultants will be shared with the audience along with the sound and vibration measure-ment techniques employed to solve the problems. A contrast will be drawn relating the paradox between the initial project objectives and high-tech instrumentation employed with the expectations of both the client and consultant. The presentation will be a great opportunity to share some of these unusual experiences with the audience and to reinforce the comic side of some of the situations we as acoustical consultants sometimes end up facing. Noisy ghosts, measurement of sound in the desert, an air horn that produces 125 dB(A) at 30 m (100 ft), and renting Lear jets for use as pink noise sources are among a few of the unusual projects that will be shared. A serious side will be presented for each project as well, primarily consisting of the application of modern noise measurement equipment (with sample data plots). The authors are sure the presentation will be both enlightening and entertaining.
Multichannel active noise control using numerically robust recursive least-squares algorithms
Authors: Bouchard Martin
Published: February 2014

Recursive least-squares (RLS) algorithms and fast-transversal-filters (FTF) algorithms were recently introduced for multichannel active noise control (ANC) systems. It was reported that these algorithms can greatly improve the convergence speed of ANC systems using adaptive FIR filters, compared to steepest descent algorithms or their variants. However, numerical instability of the algorithms was an issue that needed to be resolved. In this presentation, extensions of stable realizations of recursive least-squares algorithms such as the inverse QR-RLS and the QR decomposition least-squares-lattice (QRD-LSL) algorithms are first introduced for multichannel ANC. A first set of simulations will verify that these algorithms have indeed a better numerical stability than the previously published recursive least-squares ANC algorithms. The case of underdetermined ANC systems (i.e., systems with more actuators than error sensors) is then considered, to show that in these cases it may be required to use constrained algorithms in order to have numerical stability. Constrained least-squares algorithms for multichannel ANC systems are therefore introduced for two types of contraints: minimization of the actuator signals power and minimization of the adaptive filter coefficients squares. A second set of simulations will verify the stabilized behavior of the constrained algorithms.
New training algorithms for nonlinear active control of sound and vibration using neural networks
Authors: Bouchard Martin
Published: February 2014

In recent years, a few articles describing the use of neural networks for nonlinear active control of sound and vibration were published. Using a control structure with two multilayer feedforward neural networks (one as a nonlinear controller and one as a nonlinear plant model), steepest descent algorithms based on two distinct gradient approaches were introduced for the training of the controller network. The two gradient approaches were sometimes called the filtered-x approach and the adjoint approach. Some recursive-least-squares algorithms were also published. In this presentation, a heuristic procedure is introduced for the development of recursive-least-squares algorithms based on the filtered-x and the adjoint gradient approaches. This leads to the development of new recursive-least-squares algorithms for the training of the controller neural network in the two networks structure. These new algorithms produce a better convergence performance than previously published algorithms. Differences in the performance of algorithms using the filtered-x and the adjoint gradient approaches are discussed in the paper. The computational load of the algorithms discussed in the paper is evaluated for multichannel systems of nonlinear active control. Simulation results are presented to compare the convergence performance of the algorithms, showing the convergence gain provided by the new algorithms.
Linear arrays of powered resonance tube actuators for aeroacoustic control
Authors: Keidaisch John , Raman Ganesh , Kibens Valdis
Published: February 2014

Experiments were conducted to characterize the performance of a linear array of powered resonance tube actuators developed jointly by Boeing and IIT. The actuator is an adaptation of the Hartmann whistle and devices of this type have been shown to be effective in suppressing flow generated acoustic tones that commonly occur in aircraft applications, such as impingement tones and cavity resonances (Raman et al., AIAA Paper 2000-1930; Stanek et al., AIAA Paper 2000-1905). The goal of this characterization is to provide a database that can be used to compare the performance of these actuators with other actuator designs (such as piezoelectric actuators) and to optimize the design of these actuators. Optimization involves modifying the geometric parameters of the actuator in order to get the maximum acoustic control with minimum input mass flow rate. In order to accomplish these tasks, detailed measurements of the fluctuating velocity and pressure were made at numerous locations in the flowfield at the actuator exit. Results indicate that the actuation could produce frequencies ranging from 4352 to 11070 Hz (for the 1/2 inch depth tube chosen). Amplitudes were as high as 156 dB in the vicinity of actuation. [Work supported by Boeing/DARPA.]
On silencers, asymmetry, and group theory
Authors: Eriksson L. J.
Published: February 2014

One of the most important factors in the design of effect noise control structures is the use of asymmetry. The design of silencers and mufflers provides an excellent illustration of the use of asymmetry to improve acoustical performance. Symmetric silencer designs often concentrate performance on a narrow range of frequencies at the expense of broadband performance. Asymmetric designs enable more effect performance over a broad range of frequencies. This is particularly important for applications with wide variations in operating speed and high-harmonic content. This work examines the role of asymmetry in the design of passive and active silencers including both plane-wave and higher-order mode effects. Certain unifying principles are identified and their role in system design is discussed. It concludes with suggestions for future research on asymmetry in acoustics related to the theory of iterative structures, the Noether conservation theorem, and the mathematical theory of groups.
Acoustic waveform dependence on resonator shape in thermoacoustic engines
Authors: Hamilton Mark F. , Ilinskii Yurii A. , Zabolotskaya Evgenia A.
Published: February 2014

At high amplitudes, sound fields in thermoacousic refrigerators and prime movers are rich in harmonics and possess nonsinusoidal waveforms. In empty cylindrical resonators the natural frequencies are equidistant, the pressure waveforms are sawtooth in appearance, and they dissipate considerable energy at the shocks. Noncylindrical resonators possess nonequidistant natural frequency spectra that impede harmonic generation and reduce losses. The relative phases of the harmonics also depend on the natural frequency spectrum, mainly the relation of the second harmonic to the closest natural frequency, which usually corresponds to the second mode. Pressure waveforms are U shaped when the second harmonic is greater than this natural frequency, sawtooth when the second harmonic coincides with the natural frequency, and inverted-U shaped when the second harmonic is less than the natural frequency. These phenomena are illustrated by results of numerical calculations performed with a 2-D nonlinear model that approximates a thermoacoustic engine consisting of a stack in a resonator with varying cross section. Both pressure and particle velocity waveforms are presented at different locations in resonators of different shapes, possessing correspondingly different natural frequency spectra. Attention is devoted especially to waveforms near and inside the stack. [Work supported by ONR.]
Nonlinear properties of jet pumps
Authors: Petculescu Andi , Wilen Larry
Published: February 2014

Traveling-wave thermoacoustic devices employ jet pumps to cancel mass streaming which develops in toroidal geometries [G. W. Swift, D. L. Gardner, and S. Backhaus, J. Acoust. Soc. Am. 105, 711–724 (1999)]. We are using a simple nonresonant technique to measure nonlinear effects in jet pumps. The method involves a lumped-element analysis of a compliant region in parallel with the sample. The acoustic impedance of the sample is determined from the measured overall impedance and the impedance of the compliant region by a simple subtraction. At high amplitudes, (1) the resistance of the sample is amplitude dependent, and (2) a DC pressure develops across the sample. From these quantities, we determine the minor loss parameters Kin and Kout for a variety of jet pumps using an analysis similar to that described in Thurston [J. Acoust. Soc. Am. 30, 452–455 (1958)]. A physical picture will be presented which explains the dependence of these parameters on geometry. Time permitting, results for some exotic geometries will also be discussed. [Work supported by ONR.]
Sound propagation in the nighttime atmospheric boundary layer: The CASES-99 experiment
Authors: Noble John M. , Wilson D. Keith , Coleman Mark A.
Published: February 2014

A unique series of acoustical measurements was performed in calm, nighttime conditions in conjunction with the Cooperative Atmospheric-Surface Exchange Study (CASES-99). Conducted near Wichita, KS in October 1999, CASES-99 was one of the most ambitious experiments yet undertaken in boundary-layer meteorology. The purpose was to study processes in the nighttime boundary layer and the morning/evening transitions. Included among the instrumentation were a 60-m tower, free-flying and tethered balloons, and an FM-cw radar to image turbulence and wave motions. The concurrent sound-propagation experiment involved a series of five 6-m towers placed at distances between 300 and 1200 m from a loudspeaker. Each tower had microphones at heights of 0.5, 1, 2, and 3 m. Three transverse linear arrays were also placed on the ground at distances out to 1300 m from the speaker. Each data run consisted of broadcasting a 50-Hz square wave for a 2-h block of time. The acoustic and meteorological data are currently being analyzed. Preliminary results suggest that received sound levels are determined by both the gradual evolution of the wind and temperature gradients (due to cooling of the ground) and by discrete episodes associated with downward mixing of warm air.
The effects of turbulence anisotropy and inhomogeneity on line-of-sight sound propagation
Authors: Ostashev Vladimir E. , Wilson D. Keith
Published: February 2014

Line-of-sight wave propagation through isotropic, homogeneous turbulence has been studied by many previous researchers. A treatment for sound propagation, valid in the presence of both temperature and velocity fluctuations, was presented by Ostashev [Acoustics in Moving Inhomogeneous Media (E & FN SPON, London, 1997)]. Results for the statistical moments most frequently of interest were derived, in particular the correlation functions and variances of the log-amplitude and phase fluctuations, the mean sound field, and the coherence function of the sound field. In the present paper, these equations are generalized for the case of line-of-sight sound propagation through anisotropic, inhomogeneous turbulence. The obtained equations do coincide with those known in the literature in the limit when only temperature fluctuations are present or turbulence is isotropic and homogeneous. Using Mann’s spectrum of shear-produced atmospheric turbulence, the effects of turbulence anisotropy on the statistical moments of a sound field are studied. [This material is based upon work supported in part by the U.S. Army Research Office under Contract No. DAAG55-98-1-0463.]
Scattering of sound by a core vortex: Numerical simulations using a wide angle parabolic equation
Authors: Blanc-Benon Ph. , Dallois L.
Published: February 2014

The generation and scattering of sound by flow inhomogeneities such as vortices are basic problems which have received much attention in the efforts to develop methods to detect the strength and location of noise sources. Recent experimental studies have demonstrated that acoustic scattering can serve as an efficient (direct and nonintrusive) probe of the vorticity field for the characterization of turbulent flows. In 1978 Candel presented the results of a numerical study based on a standard parabolic equation in which the sound speed variations are included through an effective celerity ceff=c0+vx where the x axis give the direction of propagation of the acoustic wave. However, the effective celerity model does not include the effects of the perpendiculary components of the velocity field and the importance of this components increases with the angle of propagation. A wide angle parabolic equation has been recently proposed [Dallois et al., 7thAIAA/CEAS paper N 2001-2256] to take into account the mean velocity effects up to second order in Mach number. To estimate the validity of this new parabolic equation (MW–WAPE), we considered a configuration with both strong gradient and high Mach number. Our results are compared to a reference solution obtained by solving the linearized Euler equations. The MW–WAPE solutions are in good agreement with the reference solutions. This parabolic equation gives accurate results up to M≊0.5 whereas the standard equation using an effective celerity failed to give good results above M≊0.2.
Frequency selectivity estimated using stimulus-frequency otoacoustic emissions and psychophysical masking
Authors: Oxenham Andrew J. , Shera Christopher A.
Published: February 2014

Psychophysical studies of frequency selectivity have concluded that the relative bandwidths of auditory filters decrease only slightly with increasing center frequency above 1 kHz. In contrast, estimates based on measurements of stimulus-frequency otoacoustic emissions (SFOAEs) in humans and other mammals suggest that frequency selectivity improves significantly with increasing center frequency. Psychophysical measures have mostly used simultaneous masking, which can be influenced by suppression. Since suppressive effects are stronger at high frequencies, these techniques may systematically overestimate filter bandwidths at high center frequencies. This study investigates whether psychophysical and physiological estimates of filter bandwidth can be reconciled if suppressive effects are eliminated by using forward masking. Filter bandwidths at center frequencies of 1, 2, 4, 6, and 8 kHz were measured using the notch-noise technique in simultaneous and forward masking with the signal level fixed at 10 and 30 dB above threshold in quiet. Relative bandwidths estimated in the same subjects using measurements of SFOAE group delay at 40-dB SPL were consistent with earlier results. The psychophysical and physiological estimates of relative bandwidth are generally consistent, suggesting that the relationship between center frequency and filter bandwidth found in previous psychophysical studies may have been influenced by suppression. [Work supported by NIDCD.]
Accounting for manufacturing variability in interior noise computations
Authors: Allen Michael J. , Vlahopoulos Nickolas
Published: February 2014

A formulation that accounts for manufacturing variability in the analysis of structural/acoustic systems is presented. The methodology incorporates the concept of fast probability integration with finite element (FEA) and boundary element analysis (BEA) for producing the probabilistic acoustic response of a structural/acoustic system. The advanced mean value method is used for integrating the system probability density function. FEA and BEA are combined for producing the acoustic response that constitutes the performance function. The probabilistic acoustic response is calculated in terms of a cumulative distribution function. The new methodology is used to illustrate the difference between the results from a probabilistic analysis that accounts for manufacturing uncertainty, and an equivalent deterministic simulation through applications. The probabilistic computations are validated by comparison to Monte Carlo simulations. Significant computational time savings are achieved by the new probabilistic algorithm in comparison with the Monte Carlo simulations. Based on its computational efficiency and its accuracy, the new methodology is concluded to be a viable method of calculating numerically the probabilistic response of structural/acoustic systems due to manufacturing variability. [Research supported by the National Defense Science and Engineering Graduate Fellowship program.]
Can the spatial distribution of damping be measured?
Authors: Adhikari Sondipon , Woodhouse J. , Phani A. Srikanth
Published: February 2014

Our knowledge regarding the detailed description of damping in a vibrating structure is very primitive. The most common approach involves assuming proportional viscous damping, then determining damping factors for each mode. The next step, followed by many authors, is to obtain a full nonproportional viscous damping matrix. Both methods may misrepresent the true damping behavior, as viscous damping is assumed from the outset. In this study a new method for identification of the spatial distribution of damping is proposed. The approach is based on general linear damping models and is not confined to the traditional viscous damping model. Algorithms have been developed for taking measured frequencies, damping factors and complex mode shapes, and fitting the parameters of a damping model with an exponential relaxation function. The viscous damping model is a special case of this model. Simulation studies show that the method seems to perform quite robustly. Preliminary experimental results on a beam with a localized patch of damping are very encouraging and show that the method can indeed indicate the spatial distribution of damping correctly. The approach also gives some information on the question of whether the chosen exponential damping mechanism is correct or not.
Gear fatigue crack size estimation using embedded modeling
Authors: Li C. James , Lee Hyungdae , Choi Suk-Hwan
Published: February 2014

This paper describes an embedded modeling methodology for identifying gear meshing stiffness from measured gear angular displacement or transmission error. An embedded model integrating a physical based model of the gearbox and a parametric representation, in the form of truncated Fourier series, of meshing stiffness is established. A solution method is then used to find the meshing stiffness that minimizes the discrepancy between model output and the measured output. Furthermore, an algorithm is also developed to estimate the size of tooth crack from identified meshing stiffness. Both simulation and experimental studies were conducted to evaluate if identified tooth meshing stiffness can reveal a tooth crack more effectively, and if the crack size can be estimated with an adequate level of accuracy. [Work supported by ONR under the Multidisciplinary Research Program of the University Research Initiative.]
Structural assessment aided by low level response measurements
Authors: Duron Ziyad H.
Published: February 2014

The acquisition of low-level response measurements on large civil structures has been aided in recent years by significant advances in instrument and computer technologies. Measurement techniques applied to structures like dams and bridges have traditionally focused on known, measurable excitation forces (e.g., forced vibration) applied directly to the structure. An alternative way of thinking about the conduct of testing large structures is presented in which very low-level response measurements are used to evaluate dynamic characteristics, to evaluate system characteristics, and to evaluate changes in structural health or damage detection. Samples of actual field experiences are discussed and the ongoing development of a national database for large civil structural response characteristics is presented.
Computational aeroacoustics of phonation: Effect of subglottal pressure, glottal oscillation frequency, and ventricular folds
Authors: Zhang Cheng , Zhao Wei , Frankel Steven H. , Mongeau Luc
Published: February 2014

Numerical simulations of flow and acoustics in an idealized axisymmetric model of the human vocal tract have been conducted. The compressible Navier–Stokes equations were numerically integrated using highly accurate numerical treatments together with a forced glottal oscillation model and a moving grid. The effects of subglottal pressure and glottal oscillation frequency on velocity, vorticity, wall pressure, wall shear stress, and acoustic signal of the pulsating jet were investigated. An acoustic analogy was used to predict the far-field sound. Excellent agreement between the predicted and directly simulated far-field sound was obtained. The acoustic analogy was also used to decompose the acoustic source and identify monopole, dipole, and quadrupole contributions for analyses. The results show significant effects of subglottal pressure and glottal oscillation frequency on the jet vortical structure, wall forces, and acoustic radiation. The effect of including the ventricular (false) folds downstream of the oscillating glottal region was also investigated. Jet impingement on the ventricular folds introduces additional dipole sound sources. [Work supported by NIH DCO 3577-02, RO1 grant from NICDC.]
Acoustic correlates of breathy and clear phonation types
Authors: Wayland Ratree , Al-Khairy Mohamed
Published: February 2014

Only a decade ago were types of phonation shown to have a phonological function in a language. Several languages spoken in Southeast Asia were claimed to have contrastive phonation type (i.e., clear, breathy and creaky) in their vowel systems. Several different techniques were employed to investigate this phenomenon. These included examining the physical appearance of the acoustic wave form, inverse-filtering, measuring the difference between amplitude of the first and the second harmonics (H1–H2), or between the first harmonic and the most prominent harmonic in the F1 and F3 regions (H1–A1, H1–A3), harmonic-to-noise ratio (HNR) as well as some aerodynamic techniques. This present study proposes to investigate the breathy versus clear phonation contrast phenomenon in Chong, a Mon-Khmer language spoken in Thailand using several of the above-mentioned techniques. Preliminary analysis suggested that only the H1–H2 values obtained for breathy vowels were greater than for clear vowels. The nonsignificant difference between breathy and clear vowels in Chong in other dimensions will be discussed in relation to the question of whether or not Chong should be classified as a phonation-type language.
Phonation type controls the source rate
Authors: Park Hansang
Published: February 2014

The ratio of the source characteristic between the two harmonics in the relation of an octave was derived from the combined source and radiation characteristics, which shows that the source rate varies with frequency. A general term for harmonics was established from the ratio, where the higher harmonics are anchored to the fundamental component, such that the source and radiation characteristics of the higher harmonics can be represented in terms of those of the fundamental component. A new measure of phonation type was designed incorporating the varying source rate. Furthermore, it was claimed that phonation type controls the source rate. A phonation-type index k was proposed to account for variation in phonation type. The rationale for proposing the phonation-type index k is that phonation type is independent of the specific values of fundamental frequency but anchored to the fundamental component, and phonation type systematically contributes to the harmonics but not to the fundamental component. The phonation-type index k involves both open quotient and spectral tilt. The phonation-type index k may show unique distributions across speakers and contexts, and may account for modes of phonation type.
Methodology for straining engineered vocal fold tissues at audio frequencies
Authors: Klemuk Sarah A. , Titze Ingo R. , Gray Steven
Published: February 2014

Engineered vocal fold tissues will be grown under conditions of shear strain, the magnitude of which is determined by known strains during human vocal fold oscillation. A cup and plate system is used with the Bohlin CVO 120 rheometer to apply a shear strain of approximately 0.4 radians at frequencies of 20–60 Hz. Baseline viscoelastic measurements are recorded for the synthetic scaffolding material into which different cell lines will be injected and grown to form their own matrix. The methodology allows duration and repetitions of strain exposure to seeded scaffolding to be varied and for subsequent viscoelastic measurements to be compared. [Work supported by NIDCD.]
Measuring voice quality in dynamic speech
Authors: Epstein Melissa A.
Published: February 2014

Different voice qualities are used in normal speech to convey phonological contrasts (e.g., Zapotec, Gujarati, etc.), prosodic information, emotions, etc. When measuring changes in voice quality in dynamic speech, it is necessary to use a technique that can capture small, relative differences under quickly changing conditions. Although a number of techniques are available for assessing voice quality (e.g., perceptual assessment, qualitative assessment of the time amplitude waveform, qualitative assessment of the spectrogram, quantitative measurement of the spectrum, and quantitative measurement of the voicing source), they have, for the most part, been designed for capturing voice quality differences in sustained vowels or carefully matched speech conditions. This study develops a technique for relative voice quality measurement which ascertains voice quality differences across diverse words, pitches, and speakers. Specifically, inverse filtering is used to obtain the differentiated glottal flow for each vowel in the dynamic speech, which is theoretically robust across different vowel qualities and pitches. Then, for comparison, a baseline voice quality measurement is obtained for each word from each speaker to control for segmental and personal voice quality differences. Thus, the relative voice quality measurement will be resistant to the variations in dynamic speech. [Work supported by NIH and NIH/NIDCD.]
Effect of hydration on the dynamic mechanical properties of vocal fold tissues
Authors: Chan Roger W. , Tayama Niro
Published: February 2014

Previous research has shown that vocal fold hydration level is an important physiological variable affecting vocal fold vibration and vocal health. For instance, unbalanced laryngeal water transport has been associated with benign vocal fold lesions such as nodules and polyps, while hydration treatment is frequently used as a preventive or therapeutic strategy for the management of a variety of voice disorders. The theoretical basis for the role of hydration in voice production is that it may affect phonation threshold pressure or the ease of phonation, presumably by changing the viscous properties of vocal fold tissues. However, there is little empirical evidence showing how hydration might actually affect the biomechanical properties of vocal fold tissues. This study attempted to assess the effect of hydration on the viscoelastic shear properties of canine vocal fold mucosa. Osmotic changes in the hydration level of excised canine larynges were induced by incubation in hypertonic, isotonic, and hypotonic solutions. Viscoelastic shear properties of vocal fold mucosal tissues were measured with a rheometer. Significant changes in both issue elasticity and viscosity were observed with the induced changes in hydration. These findings supported the hypothesis that hydration affects voice production by altering tissue viscoelastic properties.
Pitch peak alignment as a function of lexical and pragmatic factors in two dialects of Serbian/Croatian
Authors: Smiljanic Rajka
Published: February 2014

This study examines the patterns of pitch peak retraction under focus in Serbian/Croatian (S/C). Acoustic measures of pitch peak alignment were used to identify the effect of pragmatic focus on pitch realization in two dialects of S/C with typologically distinct prosodic systems. In the Belgrade dialect there is a lexically determined contrast between two tonal alignments: early peak (‘‘falling accent’’) and late peak (‘‘rising accent’’). In the Zagreb dialect there is no such lexical distinction; instead, each word typically receives a pitch prominence on the stressed syllable. Subjects produced target words with neutral intonation and with narrow/contrastive focus. The peak alignment under these two pragmatic conditions was shown to vary in the two dialects. Zagreb shows peak retraction from late (neutral intonation) to early peak (narrow focus). In this way it is similar to other stress languages (e.g., Spanish). Belgrade, with lexically contrastive peak alignment, shows much less pragmatically conditioned variation in the position of accentual peaks. The two types of pragmatically conditioned alignment in Zagreb correspond closely to the two lexically determined alignments of Belgrade. This suggests that the existence of the lexical contrast limits the pitch variation due to the pragmatic influence, while in the absence of such contrast, pragmatic factors entirely determine peak alignment.
Acoustic/prosodic characteristics of repetitions and substitutions in medical dictations
Authors: Savova Guergana K.
Published: February 2014

This paper describes the results of a pilot study on the acoustic characteristics of two types of disfluencies exact repetitions and exact substitutions. The study was conducted at Linguistic Technologies, Inc., a company that employed automatic speech recognition (ASR) to medical dictations. Disfluencies present a significant problem to the current state-of-the-art ASR (Nakatani and Hirschberg, 1994; Lickley, 1994; Shriberg, 1994; Plauche and Shriberg, 1999). The acoustic/prosodic characteristics of disfluencies have been proposed as an additional knowledge source for improving recognition results (SRI studies, Price and Ostendorf, 1996). Disfluencies are of interest to psycholinguistics (Levelt, 1983; Clark and Wasaw, 1998). The study focuses on two talkers with a total of 67 disfluencies. The disfluency dissection is based on the Repair Interval Model (Nakatani and Hirschberg, 1994). Measurements of F0 were taken at the onset and offset peaks of the reparandum and the repair and the filled pause at the disfluency site. The duration of each word, silence, and filled pause was also measured. Entropic Cambridge Research Labs XWAVES was used for the hand-collected F0 measurements. The data gathered allowed the evaluation of several criteria: values of F0 variations, intonational contours, comparison of intonational contours, and comparisons of time durations.
The intonation of yes/no and focus constructions in Navajo
Authors: McDonough Joyce M.
Published: February 2014

This study is a preliminary investigation of tone and intonation in yes/no and focus questions in Navajo (Southern Athabaskan). Navajo is a tone language. The Navajo verb stands as complete proposition, with complex morphological and morpho-syntactic structure. The word final stem is a content morpheme preceded by functional morphemes filling out its propositional content. This stem is marked by acoustic properties commonly associated with stress: longer, louder segments and pitch range expansion, resulting in a striking end-prominent profile [McDonough, Anthro. Ling. 41, 503–539 (1999)]. Preliminary studies indicated no apparent boundary tones. Research has shown that tone languages may use intonational and/or stress-related strategies (boundary tones, pitch-range expansion) to mark focus and yes/no questions. To investigate the acoustic properties of these constructions in Navajo, 12 native speakers were recorded reciting short statements followed by either yes/no questions (three types) or focus constructions. Both type constructions in Navajo are marked by pro and enclitic-like particles on/surrounding the (nonutterance final) NPs. Results indicate no differences in the F0 contour between statements and the contrasting constructions. This fact is arguably related to the NPs nonargument status in the grammar, possibly indicating a consequential interaction of morphology, morpho-syntax, and intonation in these type languages. [Work supported by NSF.]
Prosody in spontaneous and read speech: Some preliminary findings from a tone language
Authors: Lee Chao-Yang
Published: February 2014

Intuitively it appears easy to identify whether someone is speaking spontaneously or reading out loud from a text. Previous research has shown that prosody plays a role for this perceptual distinction. The current study examines a number of acoustic prosodic parameters in the production of the two speaking styles in Mandarin Chinese. We ask whether the use of contrastive pitch in this tone language is implicated in the prosodic parameters related to the two speech modes. Spontaneous speech from three female speakers of Mandarin was recorded during interviews with the experimenter. For each speaker, ten declarative sentences were selected and transcribed. The speakers read the transcribed sentences in a second recording session 2 weeks later. Thus, ten pairs of lexically and syntactically identical sentences were obtained. Acoustic analyses were conducted of a number of intonational and temporal parameters (i.e., F0, F0 range, F0 standard deviation, F0 declination, and speaking rate). Preliminary analyses show that among the acoustic parameters, only the speaking rate consistently distinguishes between the two speaking styles. This result contrasts with previous research showing the contribution of F0-related parameters to the spontaneous-read distinction.
Intrinsic effects of obstruent consonants on F0
Authors: Hanson Helen M.
Published: February 2014

This study is part of a project to improve generation of F0 contours in rule-based speech synthesis using HLsyn. When a vowel follows an obstruent consonant, the fundamental frequency in the first few tens of milliseconds of the vowel is influenced by the voicing characteristics of the consonant. In an attempt to model this influence, pilot data were recorded by one subject. The alveolar stops /d,t,th/ and the nasal /m/ were paired with the vowels /i,a/ to form CVm syllables. The syllables mVm served as a baseline with which to compare the stop consonants. The target syllables were embedded in carrier sentences. Intonation was varied so that each target syllable was produced with either a H*, L*, or no pitch accent. F0 was increased relative to the baseline following all three obstruents, with the greatest increase for the voiceless aspirated stop and the least for the voiced stop. The F0 increase and the disparity among the three stops varied with pitch accent: H* showed the greatest influence, while L* showed the least. F0 may be increased for up to 80 ms following voicing onset. Data will be presented for additional subjects and obstruents. [Work supported by NIH Grant No. DC04331.]
Running autocorrelation method of F0 estimation
Authors: de Cheveigné Alain , Kawahara Hideki
Published: February 2014

A new method of fundamental frequency (F0) estimation is proposed that builds upon the standard autocorrelation method. Several principled steps are introduced that considerably reduce the number of errors. Formal evaluation on a database of speech recorded together with a laryngograph signal revealed an order of magnitude fewer gross errors than other standard methods. The method can be implemented efficiently, and can be extended to the task of estimating the periods of multiple concurrent sounds. A parallel may be drawn with time-domain models of auditory perception.