bluuebunny/crossref_metadata_2025 huggingface dataset:
Steady-state vibration absorption using undamped linear substructures
Authors: Koc Ilker Murat , Akay Adnan , Carcaterra Antonio
Published: October 2013

This study extends the previous applications of linear energy sinks to steady-state vibration absorption. First developed for the absorption of transient vibrations, linear energy sinks also act as effective vibration absorbers under steady-state excitation. However, their effectiveness depends on the value of excitation frequency with respect to the natural frequency of the primary structure around which the frequencies of the attached oscillators of energy sink are distributed. Vibratory energy is absorbed efficiently for excitation frequencies above the natural frequency of the primary structure, while absorption is not as effective at lower excitation frequencies. [Research carried out while AA served at NSF.]
The energy flow for guided waves at and inside the surface of an ensonified fluid loaded elastic cylindrical shell via the Poynting vector field: Preliminary results
Authors: Dean Cleon E. , Braselton James P.
Published: October 2013

The Poynting vector field is used to show preliminary results for the energy flow both at the surface and inside an ensonified fluid loaded elastic cylindrical shell for both forward and retrograde propagating waves. The present work uses a method adapted from a simpler technique due to Kaduchak and Marston [G. Kaduchak and P. L. Marston, ‘‘Traveling-wave decomposition of surface displacements associated with scattering by a cylindrical shell: Numerical evaluation displaying guided forward and backward wave properties,’’ J. Acoust. Soc. Am. 98, 3501–3507 (1995)] to isolate unidirectional energy flows.
Low-wave-number turbulent boundary layer wall pressure measurements from vibration data on a cylinder in pipe flow
Authors: Bonness William K. , Capone Dean E.
Published: October 2013

Low-order vibration modes of a cylinder subjected to a fully developed turbulent boundary layer (TBL) in a pipe are measured and used to determine the low-wave-number content of energy in the boundary layer. The experiment is conducted using a 6-in.-diameter, thin-walled, aluminum test section filled with water flowing at 20 ft/s. Coupling of certain cylinder modes in the axial and radial directions allows use of the inverse method to determine the wave-number levels of TBL wall normal pressure and wall shear stress in the flow direction. These measurements are used to evaluate commonly used empirical models of the TBL wave-number pressure spectrum attributed to Corcos (1963) and Chase (1987) at lower wave numbers than have previously been reported. Preliminary results from several higher order cylinder modes confirm prior experimental findings that the Corcos model overestimates the low-wave-number pressures for these conditions by nearly 20 dB. The Chase model provides estimates within 3 dB of the measured low-wave-number pressure spectrum. [Work supported by Navsea 073R.]
Limiting performance and optimal synthesis of elastic-inertial multi-degree-of-freedom vibration isolation systems
Authors: Ryaboy Vyacheslav
Published: October 2013

This paper is concerned with optimal problems that seek to find the best vibration isolation performance achievable by a linear mechanical system with arbitrary structure and number of degrees-of-freedom. The solution is presented in form of inequality relating isolation efficiency, frequency domain, total mass, and overall static or quasi-static stiffness. The inequality is valid for all mechanical systems of the class, however complicated these systems may be. Partial optimal problems, such as mass minimization of a system with prescribed transmissibility function, are solved as intermediate steps in deducing the limiting performance inequalities. Synthesis of a vibration isolation system with prescribed properties is a closely related problem. It has multiple solutions that are presented in a systematized way; the class of the simplest systems allowing for a unique solution is defined.
A comparison between one-sided and two-sided Arnoldi-based model order reduction (MORe) techniques for fully coupled structural-acoustic analysis
Authors: Puri R. Srinivasan , Morrey Denise , Cipolla Jeffrey L.
Published: October 2013

Reduced order models are developed for fully coupled structural-acoustic unsymmetric matrix models, resulting from Cragg’s displacement/pressure formulation, using Krylov subspace techniques. The reduced order model is obtained by applying a Galerkin and Petrov-Galerkin projection of the coupled system matrices, from a higher dimensional subspace to a lower dimensional subspace, while matching the moments of the coupled higher dimensional system. Two such techniques, based on the Arnoldi algorithm, focusing on one-sided and two-sided moment matching, are presented. To validate the numerical techniques, an ABAQUS coupled structural-acoustic Benchmark problem is chosen and solved using the direct approach. First, the physical problem is modeled using ANSYS FE package and compared with closed form solutions. Next, ANSYS results are compared with nodal velocities obtained by generating reduced order models via moment matching. The results show that the reduced order models give a very significant reduction in computational time, while preserving the desired accuracy of the solution. It is also shown that the accuracy of the one-sided method could be further improved by using two-sided methods, where the Arnoldi vectors are optimized for chosen outputs. The accuracy of the approaches, convergence models, and computational times are compared.
Complex peak time-frequency representations
Authors: Nelson Douglas
Published: October 2013

Complex representations encode twice as much information as representations encoding energy alone. Complex surfaces may be manipulated by conventional complex arithmetic operations to extract information about individual signal components. Presented are two representations. The first is an energy distribution representing energy and instantaneous frequency as the surface magnitude and phase, and the second is a signal distribution in which the instantaneous value of the signal at each time is distributed linearly in frequency. In each representation, the surface is first processed to concentrate the surface along curves representing the instantaneous frequencies of individual signal components, the surface at each time is then reduced to a short vector representing only the complex surface peaks with the largest magnitude. For the energy distribution model, the peak representation may be used to detect, track, and process individual signal components. For the signal distribution model, it is demonstrated that noise and interfering signals may be removed and an estimate of the clean signal may be recovered by a simple summation process. Moreover, the recovered clean signal is free of tonal artifacts produced by methods, such as spectral subtraction.
Efficient spectral measures for automatic speech recognition
Authors: O’Shaughnessy Douglas D.
Published: October 2013

It is well known that automatic speech recognition (ASR) requires good spectral analysis in order to have successful ASR accuracy. A wideband spectrogram seems to contain all the needed acoustic information to map any given speech signal into its corresponding sequence of phonemes. (For ASR, language models are often used to augment acoustics, but here we will limit ourselves to acoustic analysis.) Various methods beyond the basic Fourier transform have found success in ASR, e.g., linear predictive analysis, wavelets, and mel-frequency cepstra (MFCC). These have all been focussed on extracting an efficient set of spectral parameters to facilitate phonetic discrimination. Part of the difficulty is separating spectral envelope information from excitation parameters, as variations in pitch are largely viewed as orthogonal to phoneme recognition. Another complicating factor is that amplitude and frequency scales in speech production and perception are better modeled as nonlinear (unlike the linear, fixed-bandwidth approach of Fourier transforms). Modern ASR techniques are far from optimal, as the front-end data compression yielding MFCCs, for a basic 80-ms phoneme, typically has more than 100 parameters, to distinguish among approximately 32 phonemes (a 5-bit choice). We will investigate various ways to render ASR analysis more efficient. [Work supported by NSERC-Canada.]
Inverting reassignment
Authors: Magnasco Marcelo O.
Published: October 2013

The short time Fourier transform has an extremely useful feature: its integral on vertical slices is the original signal. Thus the transform itself is explicitly invertible: we can always get back the original from the transform. Its absolute value may, under some conditions on the windowing function, be inverted as a matter of principle, though an algorithm to do so is rather unwieldy. Similarly, spectral reassignment can be shown to be invertible, as a matter of principle, under certain conditions on the reassignment operator; for instance, instantaneous time-frequency and other complex analytic operators can be inverted, in the sense that all relevant information to permit inversion is preserved; this does not mean, however, that a practical way to do so suggests itself. We shall discuss recent progress on practical algorithms which preserve phase information in reassignment to permit inversion.
Dispersion curves and small-scale geophysics using noise cross-correlation techniques
Authors: Roux Philippe , Gouedard Pierre , Cornou Cecile
Published: October 2013

It has been demonstrated, both theoretically and experimentally, that the Green’s function between two receivers can be retrieved from cross-correlation of isotropic noise records. Since surface waves dominate noise records in geophysics, tomographic inversion using noise correlation techniques have been performed from Rayleigh waves so far. However, very few numerical studies implying surface waves have been conducted to confirm the extraction of the true dispersion curves from noise correlation in a complicated sedimentary ground model. In this work, synthetic noise has been generated in a small-scale (<1 km) numerical realistic environment and classical processing techniques are applied to retrieve the phase velocity dispersion curves in the medium, first step toward an inversion. We compare results obtained from SPAC (spatial auto-correlation method) and noise correlation techniques on a ten-element array. Two cases are presented in the (1–20 Hz) frequency bandwidth that corresponds to an isotropic or a directional wavefield noise.
Monitoring a volcano with passive image interferometry
Authors: Sens-Schönfelder Christoph , Wegler Ulrich
Published: October 2013

Ambient seismic noise has been used successfully as a source of information for structural investigations. Ballistic surface as well as body waves were reconstructed by correlation of noise and used in tomographic studies. In these cases it is of course assumed that the medium under study is stationary, i.e., that the reconstructed Green’s function does not change with time. In this contribution we show that medium changes can well be monitored by means of subtle changes in the Green’s function. Using an interferometric approach applied to the coda part of the Green’s function, we detect temporal changes of delay times on Merapi volcano (Indonesia). The changes of delay time depend on lapse time, which indicates that the velocity changes inside the volcano are spatially heterogeneous. We present a hydrological model that can explain the temporal changes of delay time as well as its lapse time dependence changes of the ground water level induced by precipitation. From this analysis we conclude that (a) seismic coda can be practically retrieved from noise correlations, (b) temporal changes can be monitored with noise correlations, (c) even spatial heterogeneity of the changes can be identified, and (d) the coda retrieved from noise correlations is composed body waves. a)Currently at LGIT Grenoble, France. b)Currently at SZGRF/BGR Erlangen, Germany.
Array limitations of coherent noise processing for geoacoustic inversion
Authors: Gerstoft Peter , Huang Chen-Fen , Hodgkiss William S. , Siderius Martin
Published: October 2013

Ocean acoustic noise can efficiently be processed to extract Greens function information from noise [Roux et al., JASA 2004, Siderius et al., JASA 2006]. By crosscorrelating the ambient noise field from two sensors, it is possible to extract the impulse response between the two sensors including bottom and sub-bottom bounces if the noise field is homogenous. Since the major part of the ocean-noise is horizontal and thus strongly inhomogeneous, this processing does not extract the vertical bouncing energy well. Using array processing it is possible extracting the vertical bouncing energy. However, this beamforming-like approach can just be done for frequencies less than the array design frequency. For higher frequencies the side lobes in the beam pattern will make the horizontal energy appear as vertical and can thus not be used. Vertical velocity sensors can filter out some of this horizontal energy and a vertical velocity sensor array can be used at higher frequencies than a pressure sensor array. When this noise processing is used on a vertical array, it gives valuable information about the subbottom near the array.
Cathedral acoustics and organ sound reinforcement
Authors: Mapp Peter
Published: October 2013

The acoustic characteristics of 4 UK cathedrals have been investigated. In three of the buildings it is reported that the organ requires additional support or reinforcement to cope with the large volume of the space involved. All 4 buildings were found to exhibit multiple coupled spaces making it difficult to accurately assess the active acoustic volume. In order help to overcome the perceived lack of ’volume’ from the organ, in two of the cathedrals, it is proposed to install a second rank of pipes some distance down the nave from the main organ. The acoustic and psychoacoustic implications if this approach are discussed. An alternative means of providing support for the organ via the installed sound reinforcement system has also been investigated and the electroacoustic and subjective requirements for this are discussed and reviewed.
Autonomous undersea vehicles and acoustical oceanography
Authors: Austin Thomas , Allen Ben
Published: October 2013

Autonomous undersea vehicles (AUVs) have been actively increasing in their capabilities over the past 10 years. Recent developments in the academic sector have transitioned to become commercially available technologies which are now available for general use in academic, commercial and military applications. The Oceanographic Systems Laboratory at Woods Hole Oceanographic Institution has been a leader in the development of robotic vehicles, with most development funding coming from the Office of Naval Research. The result of this development effort has been the REMUS family of AUV systems. Originally designed primarily for shallow water hydrographic reconnaissance and mine-hunting, these systems have since been adapted and utilized for many other applications including deep ocean search and survey, physical oceanography, and more recently acoustical oceanography. New, larger systems contain sufficient energy and payload capabilities to tow long hydrophone arrays, and can carry large projectors allowing active and passive techniques to be used in unmanned, un-tethered systems at any depth in the oceans. Recent developments in these unmanned vehicle systems will be discussed along with new payload capabilities now under development for acoustical oceanography.
Acoustic measurements in shallow water using an ocean glider
Authors: Dossot Georges A. , Miller James H. , Potty Gopu R. , Moore Kristy A. , Holmes Jason D. , Lynch James F.
Published: October 2013

Acoustic data was collected on a single hydrophone attached to a Webb Slocum glider deployed by Rutgers University during the Shallow Water Experiment (SW06) on the continental shelf off New Jersey. The geometry of the experiment provided for adequate recording of the 224 and 400 Hz WHOI tomography sources. The glider periodically surfaced for GPS fixes and data transfer via satellite phone. A synthetic aperture is created through coherent processing of the acoustic data as the glider traveled through the water. A number of issues including varying depth and nonsteady motion must be accounted for due to the trajectory of the glider. However, the glider provides a low-noise and low-speed platform, potentially enabling detection of low level signals. Initial results show that the phase of the tomography signals is linear as the glider traverses the sound field. An acoustic normal mode representation of the field provides the basis for processing of the data similar to Hankel transform approach of Frisk and Lynch [J. Acoust. Soc Am. 76, 205–216 (1984)] and Holmes, Carey, and Lynch [J. Acoust. Soc. Am. 119, 3346 (2006)]. Effects of spatial variations in sound speed are examined. [Work sponsored by the Office of Naval Research.]
Physical and historical bases of biomedical applications of radiation force
Authors: Sarvazyan Armen
Published: October 2013

Radiation force is a universal phenomenon in any wave motion, electromagnetic or acoustic. Radiation force is produced by a change in the density of energy of the propagating wave. The concept of radiation pressure follows from Maxwell’s equations (1871), while the idea of electromagnetic wave (light) pressure can be traced back to Johannes Kepler (1619). Acoustic radiation force was first experimentally demonstrated by August Kundt (1874) and its physical bases were analyzed in classical works of Lord Rayleigh (1902), L. Brillouin, and P. Langevin. Until recently, main biomedical application of acoustic radiation force was in measuring acoustic power of therapeutic devices, but during the last 15 years it became one of the hottest areas of biomedical ultrasonics with numerous new applications ranging from acoustical tweezers, targeted drug and gene delivery, manipulating of cells in suspension, increasing the sensitivity of biosensors and immunochemical tests, assessing viscoelastic properties of fluids and biological tissue, and monitoring lesions during therapy. The widest new area of applications of radiation force is related to medical imaging in general and elasticity imaging specifically. Remote excitation of shear stress in a given site in the body using radiation force of focused ultrasound offers numerous new possibilities for medical diagnostics.
A noninvasive resonant mode imaging
Authors: Zhang Xiaoming , Fatemi Mostafa , Kinnick Randall R. , Greenleaf James F.
Published: October 2013

The mechanical response of tissues to external forces has gained considerable interest in medical diagnosis. One approach of imaging the tissue elastic properties is to apply an external force on the surface of the body. Another approach is to generate a localized force inside the tissue with the radiation force of ultrasound. In this paper, a noninvasive method for imaging the vibration mode of an object is investigated. The radiation force of ultrasound is used to scan the object at a resonant frequency of the object. The vibration of the object is measured by laser and the resulting acoustic emission from the object is measured by a hydrophone. Experimental studies are carried out on a mechanical heart valve and arterial phantoms. The mode images on the valve are made by the hydrophone measurement and confirmed by FEM simulations. Compared with conventional B-scan imaging on arterial phantoms, the mode imaging can show not only the interface of the artery and the gelatin, but also the vibration modes of the artery. The images taken on the phantom surface suggest that an image of an interior artery can be made by vibration measurements on the surface of the body.
Histology-based simulation of ultrasonic scattering in cells and tissues
Authors: Doyle Timothy E. , Warnick Keith H. , Carruth Brent L.
Published: October 2013

Previously developed computer programs for simulating ultrasonic scattering in cells and tissues at the microscopic level [T. E. Doyle and K. H. Warnick, J. Acoust. Soc. Am. 120, 3283 (2006)] have been further refined and tested. The programs model the cells and nuclei in tissues as spherical particles with the use of vector multipole expansions and boundary condition solutions for the scattered fields. In addition, theorems, iteration, and matrix methods are used to solve for multiple scattering both inside and outside the cells. The latest refinements include the development of algorithms for simulating ultrasonic scattering from cells with nuclei arbitrarily located within the cell, and not centered as in previous models. Backscatter spectra have also been acquired for simulated cell clusters of up to several thousand cells. The effects of tissue structure, nucleus size, extracellular elastic properties, and intracellular elastic properties were investigated. The models provide a mechanistic link between measurable parameters in medical ultrasound and histological changes associated with various tissue pathologies. The application of these results to the improvement of ultrasonic tissue characterization and the specificity of various ultrasonic imaging modalities are discussed.
A perceptual space that can explain the robustness of bio-acoustic communication
Authors: Patterson Roy D. , van Dinther Ralph , Irino Toshio
Published: October 2013

The sounds that animals use to communicate, including the syllables of speech, have a very special ‘‘pulse resonance’’ form, which automatically distinguishes them from background noise. The parts of the body used to produce these sounds grow as the animal grows. Thus, there is ‘‘acoustic scale’’ variability in communication sounds, which poses a serious problem for the perception and recognition stages of communication. The success of bio-acoustic communication suggests that the auditory system has a special preprocessor that automatically normalizes for acoustic scale as it constructs our internal ‘‘auditory image’’ of a sound. In this paper, we propose that the normalization is performed in the early stages of the auditory pathway by adding an extra, rather special, dimension to the space of auditory perception. This paper is about the mathematics of the space, which has to be ‘‘scale-shift covariant’’ to support communication, and the discovery of a unitary operator that can construct the appropriate space. The mathematics makes it clear that there is no equivalent means of scale normalization available in the traditional time-frequency space of the spectrogram. [Research supported by the UK MRC (G0500221, G990369).]
Numerical simulation of rock acoustics with a multi-phase model
Authors: Song Limin , Zhang Yang , Deffenbaugh Max
Published: October 2013

Understanding how the effective acoustic properties of porous rock are related to its physical properties is very important in exploration and production of oil and gas, and remains an active research topic. We present a computational method to study the acoustics of fluid-filled porous rock over a range of frequencies. The method solves the elastic wave equation in the solid rock matrix and the viscous wave equation in the pore fluid. The coupled multi-phase wave equations are solved numerically over a digital representation of the rock sample where the material phase and properties at each volume cell are identified. The significance of the new method is that the viscous effects of pore fluid on the bulk acoustic response are modeled accurately with the linearized Navier-Stokes equation instead of with the generalized Maxwell body low frequency approximation that is commonly used to overcome the difficulty of modeling frequency-dependent fluid shear modulus in time domain. The rotated staggered grid finite difference scheme is used to solve the coupled multi-phase wave equations. We simulate P-wave propagation through an idealized rock model of alternating solid and fluid layers where an analytical solution is available and obtain excellent agreement between numerical and analytical results.
Sediment attenuation effects in sound transmission results from the New Jersey Shelf
Authors: Dediu Simona M. , Siegmann William L. , Carey William M. , Pierce Allan D.
Published: October 2013

Experiments were conducted near the site of AMCOR Borehole 6010 on the New Jersey Shelf to characterize propagation predictability. The importance of a non-linear power-law frequency dependence of the sediment volume attenuation in the uppermost sediment layer is demonstrated. One metric of transmission loss variation with range is an effective attenuation coefficient that can be extracted from measurements and calculated with the parabolic equation. Previously it was found [W. M. Carey and R. Evans, J. Oceanic Eng., 23] that a power exponent of 1.5 modeled the measurements. The present approach uses 1 kHz as an attenuation reference frequency and employs different parameter ranges and optimization criteria. For 400–1000 Hz, this procedure leads to a power exponent in the range 1.7–2.0, which is consistent with other sand-silt regions. The estimates are robust with respect to variations in the water and sediment sound-speed profiles and the sediment layer thickness. The influence of measured range dependence in the sound speed and bathymetry is examined. Estimates of signal time spread are calculated, which can also be obtained for recent experiments in the same ocean region. [Work partially supported by ONR]
Bayesian focalization and tracking
Authors: Dosso Stan E. , Wilmut Michael J.
Published: October 2013

This paper describes a Bayesian approach to source localization and tracking with uncertain ocean environmental parameters (water column and seabed). The inversion is formulated for both source location and environmental parameters, and solved using Gibbs sampling methods which sample directly from the posterior probability density. The information content for source localization is quantified in terms of probability ambiguity surfaces, consisting of joint marginal probability distributions for source range and depth integrated over unknown environmental parameters, similar to the optimum uncertain field processor (OUFP). The posterior environmental information content, including both prior information and information provided by the acoustic data, is also quantified in terms of marginal distributions. In source tracking, knowledge of the environmental parameters can be augmented as the tracking progresses, by explicitly passing posterior environmental information from one localization on, as improved prior information in the subsequent localization. Information on interparameter correlations is included by formulating marginal distributions in terms of principle components (empirical orthogonal functions) of the environmental parameters.
Trends over minutes to decades in oceanic ambient sound measured off the southern Californian coast
Authors: Andrew Rex K. , Leigh Charlotte V. , Howe Bruce M. , Mercer James A.
Published: October 2013

Ambient sound spectra have been collected since 1994 from a decommissioned deep water military acoustic receiver located southwest of San Nicolas Island (southern California). A new spectrum is collected every 6 minutes. The dataset can be used to study level variability on scales from minutes to years. The calibration curves for this system, however, were considered suspect. Recently, a calibrated acoustic recorder was deployed near the receiver (McDonald et al., JASA, 120(2), August 2006). Comparison of spectra for December 2003 provides a long-sought-after modern absolute calibration. With this new correction, the dataset can additionally be compared to levels measured in the 1960’s by the same receiver. Although the corrected measurements corroborate (albeit by construction) the 1960–2000 ambient noise increase reported by McDonald et al., there is scant evidence of an increase over the last decade. This study again highlights the need for on-going long-term well-calibrated observation programs. [Work sponsored by ONR.]
An evaluation of acoustical characteristics of classrooms in tropical environments
Authors: Alur Ramachandraiah , Subramaniam Nithya
Published: October 2013

Classrooms in India, especially those located in warm humid areas, are generally equipped with mechanical ventilation such as fans or air-conditioning systems which influence the background noise of these classrooms. These systems obviously lead to a high level of background noise even during unoccupied conditions. Ventilation requirements become acute in summer and rooms are provided with a lot of open windows. This will be a problem if there are high levels of external noise. An acoustical analysis has been carried out to assess the acoustical quality and conditions for speech communication. Speech intelligibility tests along with physical and acoustical measurements were made in various classrooms (IIT Madras campus). Octave band measurements of background noise levels and reverberation times were made. The measured values have been compared with a few existing standards such as ANSI. Preliminary surveys have been administered to students and instructors to get a subjective opinion of their experiences in these environments. The classrooms consisted of untreated, partially treated, and fully treated rooms. The inferences from this study have been drawn to suggest better design procedures for such classroom environments.
Utilizing variable acoustics at Snow College’s Jorgensen Concert Hall—A user’s perspective along with acoustical parameters measured in the hall
Authors: Larsen Vance E. , Dohn Bill
Published: October 2013

The Jorgensen Concert Hall is the centerpiece of the Horne School of Music at Snow College in Ephraim, Utah. It seats about 720 people and was completed in 2004. By adjusting reverberation and sound reflections on the stage and in the house, the Jorgensen accommodates a wide variety of college concerts, including jazz big band, chamber ensembles, choral groups, concert band, and symphony orchestra. The Jorgensen has also become popular with visiting groups such as the Utah Symphony, the Orchestra at Temple Square, and the Julliard Jazz Trio. Successes and challenges in using the Jorgensen will be discussed, along with a presentation of acoustical parameters measured in the hall.
Sound attenuation in ducts utilizing spatially periodic area changes with absorbing material
Authors: Burton Lisa , Bliss Donald , Franzoni Linda
Published: October 2013

The propagation of quasi-one-dimensional acoustic waves through a duct with spatially periodic area changes has been studied theoretically and experimentally in previous research. This configuration exhibits stop-band and passband behavior, with substantially reduced sound transmission in the stop bands, but little effect in passbands. In the present study the periodically spaced regions of the duct with larger cross-sectional area are partially filled with porous material, leaving a constant area passage for airflow. The effect of the porous material is to make the propagation wave number in the larger cross-sectional areas complex, thereby providing significant attenuation even in passbands. To determine this wave number, a fairly simple theory based on cross-sectional averaging and a higher order correction dependent on geometry is developed to achieve excellent accuracy. The dispersion relation for the duct and the transfer function for the overall system are analyzed using a transfer matrix approach. The relationship between area change parameters and the location and width of stop bands is discussed. Experimental measurements for ducts with periodic area changes having open cell foam as the porous material are compared with the theory. Application to a muffler designed to place target engine harmonics in stop bands is shown.
Prediction of porous media properties using a cross-sectional averaging theory for one-dimensional sound propagation in ducts having two propagation media
Authors: Burton Lisa , Bliss Donald , Franzoni Linda
Published: October 2013

An alternative method to measure the complex effective wave speed and density of bulk reacting porous materials is described. A hybrid duct partially filled with porous material, having constant cross-sectional properties, is studied theoretically and experimentally. A fairly simple quasi one-dimensional theory based on cross-sectional averaging is derived, compared to exact solutions, and found to work extremely well below the cut-on frequency for higher modes. The basic theory depends only on the ratio of cross-sectional areas and the properties of the individual propagation media, but not on the configuration of material in a cross-section. Accuracy can be improved at higher frequencies through a higher order correction that includes a shape coefficient that depends on the specific cross-sectional configuration. Once the propagation wavenumbers are known for two or more cross-sectional configurations having the same two media, e.g. air and porous material, it is shown that the porous material properties can be extracted relatively easily. Experimentally, porous material samples line the sides of a tube closed at one end, and the complex wavenumber is determined from standing wave measurements. Results are compared for various lining configurations to assess the accuracy of the method and the sensitivity to cross-sectional configuration in practice.
Methodology for reconstruction of vehicle accident acoustics for use in forensic visualization
Authors: Neale William , Terpstra Toby
Published: October 2013

This paper describes a methodology for incorporating physically accurate sound composition in forensic visualizations. The use of sound in forensic visualization provides the viewer a more realistic and comprehensive understanding of actual accident events. Forensic visualization can represent complex events such as car crashes through animation, making it easier to understand the accident. Without sound, however, a visual representation of an accident will lack important information. Sound adds a spatial dimension for the viewer since sound defines space through such properties as reflection and reverberation. In addition, sound provides an understanding of important details of the accident such as the duration and severity of an accident, and potential audible sounds heard by a witness. Sound also allows the viewer to experience events in the accident that are outside the viewer area or occluded from view. Currently, there is no methodology for compositing sound in an animation to follow the principles of sound and reflect the specifics of an accident. Acoustical principles define how sound attenuates, reflects, dampens, blends, and changes in pitch and sound level. The unique circumstances of an accident define what sounds are present and the timing and sequencing of these sounds. This paper provides a methodology for creating sound that both follows acoustical principles and reflects the unique circumstances of an accident.
Evidence that stimulus bandwidth affects temporal resolution and temporal integration in naive listeners
Authors: Mossbridge Julia A. , Wright Beverly A.
Published: October 2013

The separation of auditory events in time (temporal resolution) and the summation of these events over time (temporal integration) are both essential to normal hearing. Here, it is reported that both of these abilities are affected by stimulus bandwidth in naive listeners. In a single group of 30 naive listeners, amplitude-modulation detection thresholds were determined over a range of modulation rates, and signal-detection thresholds over a range of signal durations, using noises of two different bandwidths (0.25 kHz and 3.80 kHz), both with a 4.40-kHz upper cutoff frequency. Amplitude-modulation thresholds were adjusted mathematically to compensate for reductions in modulation depth introduced by filtering after modulation. The average time constants estimated from these data were significantly larger for the narrowband than the broadband noise for both temporal resolution (5.1 versus 1.3 ms; p<0.0001) and temporal integration (251 versus 13 ms; p<0.0001). Thus, naive listeners showed shorter temporal resolution and integration windows with a wider stimulus bandwidth, suggesting that, at least for this population, both of these temporal-processing abilities are influenced by the spectral parameters of the input. [Work supported by a grant from NIH/NIDCD.]
Spatial release from speech-on-speech masking in adult hearing aid users
Authors: Marrone Nicole L. , Mason Christine R. , Kidd Gerald
Published: October 2013

Although it is well documented that spatially separating speech maskers from a target speech source normally results in a release from masking, little is known about this effect in hearing aid users. The current study addressed the interaction between spatial release from masking, reverberation, age, and amplification in a multitalker soundfield experiment. The listeners had symmetric sensorineural hearing loss and were experienced users of bilateral hearing aids. The task was to identify the speech of a target talker located directly in front of the listener (0 deg azimuth) in the presence of two masker talkers co-located with the target or spatially separated at ±90 deg azimuth. In this symmetric spatial configuration of sound sources, there was not an overall difference in target-to-masker ratio (T/M) at either ear during stimulus presentation, thus eliminating any ‘‘better ear’’ advantage. Reverberation was varied by physically changing the sound absorption characteristics of the sound booth in which the experiments were conducted. Four listening conditions were tested: unaided, bilateral aided, unilateral aided-right, and unilateral aided-left. In general, the hearing aid users demonstrated less spatial release than normal-hearing listeners although there were large inter-subject differences across the different aided conditions. [Work supported by NIH/NIDCD and AFOSR.]
Verification of the assumption of frequency-independent recovery from forward masking
Authors: Wojtczak Magdalena , Oxenham Andrew J.
Published: October 2013

Psychophysical methods for deriving basilar-membrane (BM) input-output functions from forward-masking data assume that the recovery from forward masking at a given probe-frequency place is independent of the masker frequency. The validity of this assumption was tested here. The levels of an on-frequency (4-kHz) and off-frequency (2.4-kHz) forward masker were individually adjusted to yield the same masked threshold for the probe presented immediately after the masker offset. Once the masker levels had been established, probe thresholds were measured using these maskers for masker-probe delays between 0 and 115 ms. To prevent off-frequency listening, a low-level notched noise centered on the probe frequency was presented throughout each trial. Possible confusion effects were reduced by presenting a contralateral 7-kHz tone synchronous with the masker. Since the recovery functions started with equal probe levels, any differences between the recovery functions for the on- and off-frequency masker would imply masker-frequency or masker-level dependence of forward-masking recovery. Preliminary data suggest some listeners may exhibit a slower recovery for the high-level (92-dB SPL) off-frequency masker than for the lower-level on-frequency masker. Implications for BM compression estimates derived from psychophysical forward masking will be discussed. [Work supported by NIH grant R01DC03909.]
Temporal integration and compression near absolute threshold in normal and impaired ears
Authors: Plack Christopher J. , Skeels Vicki
Published: October 2013

The decrease in absolute threshold with increasing stimulus duration is greater for listeners with normal hearing than for listeners with sensorineural hearing loss. It has been suggested that the difference is related to reduced basilar-membrane (BM) compression in the impaired group. This hypothesis was tested by comparing temporal integration and BM compression near absolute threshold in normal and impaired ears. In the first phase, absolute thresholds were measured for 4-, 24-, and 44-ms pure-tone signals, with frequencies (fs) of 2 and 4 kHz. The difference between the absolute thresholds for the 4- and 24-ms signals was used as a measure of temporal integration. In the second phase, compression near threshold was estimated by measuring the level of a 200-ms off-frequency (0.45 fs) pure-tone forward masker required to mask a 44-ms pure-tone signal presented at sensation levels of 5 and 10 dB. There was a significant negative correlation between the amount of temporal integration and absolute threshold, but there was no correlation between absolute threshold and compression at low levels; both normal and impaired ears showed a nearly linear response. The results suggest that the differences in integration between normal and impaired ears cannot be explained by differences in BM compression.
Word recognition in dysarthric speech: Evidence for the time-course hypothesis
Authors: Lansford Kaitlin , Liss Julie , Spitzer Stephanie , Mattys Sven , Caviness John
Published: October 2013

The detrimental effect of speaker variability on word recognition has been reported using a wide variety of tasks and stimuli. In this study, the use of dysarthric speech alongside unimpaired speech allowed us to examine the effect of indexical variability from a new perspective. We compared voice-specificity effects in words produced by healthy speakers and the same words produced by dysarthric patients suffering from Huntington’s disease (HD). The speech of persons with HD is particularly suited to this issue because of its idiosyncratic and variable manifestation. Reaction time and accuracy results provided confirmatory evidence for Luce et al.’s (2003) time-course hypothesis: The slower responses occasioned by processing dysarthric stimuli were accompanied with greater reliance on surface details of the stimuli. Thus, specificity effects can be seen as resulting from a coping strategy called upon when the system is presented with an input that is highly dissimilar from canonical lexical representations. In addition, we found some evidence that the time-course hypothesis holds even when intelligibility is controlled. This finding confirms that time, rather than processing effort, is the cause of specificity effects, with activation confined to abstract representations in an early stage and spreading to episodic traces later on.
Modeling the effects of frequency shifts on vowel identification
Authors: Nearey Terrance M. , Assmann Peter F.
Published: October 2013

Previous experiments examining the effects of frequency shifts on vowel perception show that identification accuracy drops when the spectrum envelope is shifted up by more than about 150%, or shifted down by factors smaller than about 60% relative to adult male ranges. Such shifts produce formant patterns near the extreme limits found in human voices. But these effects interact with fundamental frequency (F0): in some conditions identification accuracy is improved by shifting the formant frequencies (FFs) and F0 in the same direction, compared to conditions where one is raised and the other is lowered. The results indicate the presence of perceptual mechanisms that are sensitive to the natural covariation of F0 and FFs in human voices. Initial modeling shows that including F0 and FFs predicts listeners’ behavior better than FFs alone. Specifically, posterior probabilities from linear discriminant function analysis are better correlated with listeners’ identification rates when F0 is included than when it is not. Further modeling suggests prediction of overall perceptual results generally improves (especially in mismatched conditions) for modified models that include a positive correlation between F0 and FF that is somewhat weaker than that observed in natural speech databases. [Work supported by NSF and SSHRC].
Laryngeal airflow modeling; an investigation based on computational aeroacoustics
Authors: Mihaescu Mihai , Gutmark Ephraim , Khosla Sid , Scherer Ronald
Published: October 2013

The paper presents an efficient method for computational aeroacoustics applied to simulate the flow and the acoustics for a static laryngeal model considering the vocal folds widely open. The work analyzes the whisperlike process defined as the sound generated by the turbulent glottal airflow in the absence of vocal fold vibration. A decomposition of the flow variables is used that allows separation of flow and acoustic computations. The approach consists in solving the low Mach number flow field by incompressible large eddy simulation. This provides the possibility to compute the acoustic sources. The inhomogeneous wave equation derived from the fundamental equations of motion for a compressible fluid is employed to compute the acoustic field. The purpose of the study is to provide realistic numerical predictions of the flow patterns and the generated acoustic field in the glottis and the vocal tract regions.
External factors impacting the performance of speaker identification: Multisession audio research project (MARP) corpus experiments
Authors: Wenndt Stanley , Lawson Aaron , Stauffer Allen
Published: October 2013

This study looks at the effect of data conditions on automated speech processing systems. The goal is to better understand the impact of acoustical features on accuracy and to develop more robust features. A speaker identification (SID) system was used for the experiments. To explore this issue, a new longitudinal database was collected involving 60 speakers over 18 months. This corpus allowed us to examine four data factors that impact SID: (1) intersession variability, (2) question intonation (3), text-dependency (identical phonetic content), and (4) whispered speech. First, we found that intersession SID suffered an average loss in accuracy of 17% independent of the time latency between sessions. Second, differing the intonation conditions in train and test hurt SID performance by 5%. Third, text-dependent data showed the most dramatic impact, where using phonetically-identical test and train sentences yielded 0% error. However, replacing the target speakers content with a random text-independent sentence, ceteris paribus, caused accuracy to plummet 94%. In all of the erroneous identifications, the top-ranked speaker was speaking the sentence used for the training model. Finally, when there was a mismatch between whispered speech being used in training and normally phonated speech in testing, the SID performance was severely impacted.
Flutter echo at Pigott Auditorium
Authors: Jurdy Basel
Published: October 2013

A flutter echo was easily apparent after completing the remodel of Pigott Auditorium at Seattle University in Seattle, Washington. Knowing that this was a possibility in the remodel design, walls were articulated in an attempt to avoid this phenomenon. The techniques used to investigate the problem, results of the investigation and the eventual remedy are presented.
Programming remediation to a newly complete performing arts center
Authors: Rosenberg Carl J. , Hinckley Jennifer , Sacks Jonah
Published: October 2013

A large new performing arts building at a university included an 800-seat recital hall, proscenium theater, black box theater, large instrumental rehearsal room, choral rehearsal room, and practice rooms. It was designed with attention to acoustics concerns throughout, but the final results left the owner with a myriad of problems. These problems included poor hearing conditions in the large performance space, intrusion of exterior environmental noise, excessive mechanical system noise and vibration, and inexplicable lack of separation between rehearsal rooms that were separated by an STC 70+ wall. The authors parsed through the complaints and field conditions to extract a range of reasons for the problems. The paper presents the analysis techniques for these challenges and shows options that were developed for practical solutions.