bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic-phonetic characteristics of naturally-elicited clear speech in British English.
Authors: Baker Rachel , Hazan Valerie
Published: October 2013

Clear speaking styles are characterized by enhancements of specific acoustic-phonetic aspects of the speech signal. However, most studies of casual and clear speaking styles have been based on read speech recorded in laboratory conditions, both normally and when instructed to speak clearly. In this study, casual and clear speech produced by 40 talkers of southern British English was elicited in unscripted dialogues between two talkers via a series of “spot the difference” picture tasks, based on the “diapix” task developed by Bradlow and collaborators. In the “casual speech” condition, the two talkers conversed normally while doing the task. In the “clear speech” condition, to simulate communication between a normal-hearing talker and a talker with a cochlear implant, one of the talkers heard the other talker’s speech passed through a three-channel vocoder. Additionally, talkers read sentences containing specific keywords that also occur in the diapix task, both normally and when instructed to speak clearly. Preliminary acoustic-phonetic analyses of the speech corpus will be presented. The clear speech elicited from the diapix and “read speech” conditions will be compared to ascertain whether clear read speech shows similar characteristics to clear speech elicited in more natural communicative situations. [Work supported by ESRC.]
Improved learning of a non-native phonetic contrast by combining active task performance with passive stimulus exposures.
Authors: Marrone Nicole , Wright Beverly A.
Published: October 2013

Adults can learn to hear the phonetic distinctions of another language with practice. It is commonly assumed that such improvements require active performance of the target task throughout training. However, we have seen that, for other perceptual tasks, similar performance gains can be achieved by replacing some active-performance trials with passive stimulus exposures. Here we examined the learning of a non-native phonetic contrast using several training protocols that varied in the proportion of active and passive trials during the practice period. Native, monolingual English speakers were trained to identify a Thai phonetic contrast along a voice-onset-time continuum. A group given half active training and half passive exposures showed as much improvement in the category boundary as a group given active training for the entire practice period. The active-passive group out-performed a group given half active training and half training on an unrelated task with no passive exposures. Finally, a group given only passive stimulus exposures showed the least improvement. Results suggest that learning a non-native phonetic contrast via active task performance can be reinforced by passive stimulus exposures. Therefore, active-passive training could be useful in real-world speech training applications. [Work supported by NIH/NIDCD.]
A tutorial on nonstationarity detection in acoustic signals: Parametric and nonparametric approaches.
Authors: Wolfe Patrick J.
Published: October 2013

This tutorial provides an overview of nonstationarity detection in acoustic signals, focusing on model-based parametric approaches as well as more flexible nonparametric ones. The techniques discussed are presented in the context of speech and audio waveforms, with several real-world examples given, but also apply more broadly to any class of acoustic signals that exhibits locally stationary behavior. Many such waveforms, in particular information-carrying natural sound signals, exhibit a degree of controlled nonstationarity, and are often well modeled as slowly time-varying systems. The tutorial first describes the basic concepts of such systems and their analysis via local Fourier methods. Parametric approaches appropriate for speech are then introduced by way of time-varying linear predictive models, along with nonparametric approaches based on variation of time-localized estimates of the power spectral density of an observed random process. [Work supported in part by DARPA, NGA, and NSF.]
Bayesian source tracking in an uncertain ocean environment.
Authors: Dosso Stan E. , Wilmut Michael J.
Published: October 2013

This paper considers matched-field tracking of a moving acoustic source in the ocean when acoustical properties of the environment (water column and seabed) are poorly known. The goal is not simply to estimate source locations, but to determine localization uncertainty distributions, thereby quantifying the information content of the tracking process, and to use this information to probabilistically predict future source locations. To this end, a Bayesian formulation is applied in which source and environmental parameters are considered random variables constrained by noisy acoustic data and by prior information on parameter values (e.g., physical limits for environmental properties) and on interparameter relationships (limits on source velocity). Source information is extracted from the posterior probability density (PPD) by integrating over unknown environmental parameters to obtain a time-ordered series of joint marginal probability surfaces over source range and depth. Given the strong nonlinearity of the localization problem, marginal PPDs are computed numerically using efficient Markov-chain Monte Carlo methods, including Metropolis-Hastings sampling over environmental parameters (rotated into principal components and applying linearized proposal distributions) and heat-bath Gibbs sampling over source locations. Several approaches to estimating optimal tracks, track uncertainties, and future source locations from the set of marginal-probability surfaces are considered.
Statistical signal characterization in ocean acoustic tomography and geoacoustic inversions.
Authors: Taroudakis Michael , Smaragdakis Kostas
Published: October 2013

A performance study of a statistical characterization of an underwater acoustic signal in relation to geoacoustic inversion or tomography problems is presented. The method of characterization has been presented by Taroudakis et al. [JASA 119, 1396–1405 (2006)] and is based on the use of an appropriate distribution law which describes the statistics of the sub-band coefficients of the signal wavelet transform. The method has been applied with synthetic data simulating tomographic experiments with low-frequency sources in shallow environments for the estimation of the water column and bottom properties. In this work the inversion procedure incorporating a genetic algorithm is applied in shallow water environments with simulated noisy data to assess the performance of the method and its limitations under realistic conditions.
Wavelet-based neural networks applied to automatic detection of road surface conditions using tire noise from vehicles.
Authors: Kongrattanaprasert Wuttiwat , Nomura Hideyuki , Kamakura Tomoo , Ueda Koji
Published: October 2013

The detection of road surface conditions is an important process in efficient road management. In particular, in snowy seasons, prior information about the road conditions such as an icy state, helps road users or automobile drivers to obviate serious traffic accidents. This paper proposes a novel approach for automatically detecting the states of the road surface from tire noises of vehicles. The method is based on a wavelet transform analysis, artificial neural networks, and the mathematical theory of evidence. The proposed method employs the wavelet transform using multiresolution signal decomposition techniques. The proposed classification is carried out in sets of multiple neural networks using learning vector quantization networks. The outcomes of the networks are then integrated using the voting decision-making scheme. It seems then feasible to detect passively and readily the states of the surface, i.e., as a rule of thumb, the dry, wet, snowy, and slushy state, automatically.
Audio enhancement of biomechanical impact data.
Authors: Guarino Joe , Orme Wes , Fischer Wayne
Published: October 2013

Analysis and interpretation of impact data from a force transducer or accelerometer can be augmented and enhanced using audio playback. Trends and differences which may be difficult to identify using data imagery can be elucidated and reinforced by converting digital data from a force plate to an audio signal, which can then be played through a high-quality speaker system. The audio stream can be processed using standard acoustical methods such as tempo and pitch shifting, which can emphasize frequencies and tone bursts for improved signal characterization. We apply audio enhancement to data from two separate biomechanical studies: (1) a drop landing experiment for the investigation of gender differences in impact upon landing and (2) an experiment for the investigation of impact differences between cleated and noncleated shoes on artificial turf. The continuous wavelet transform (CWT) is used to process data from a force plate in both studies. Results are compared using the semblance analysis approach described by Cooper and Cowan [“Comparing time series using wavelet-based semblance analysis,” Computers Geosciences 34, 95–102 (2008)]. Interpretation of results is enhanced using speaker-driven audio output synthesized from the CWT and semblance analysis.
Burial depth dependence of the bistatic scattering amplitude for cylinders illuminated by evanescent waves using two–dimensional finite elements.
Authors: Cerruti Nicholas R. , Thiessen David B. , Marston Philip L.
Published: October 2013

Prior research examined the dependence on simulated burial depth of the low-frequency scattering by small targets illuminated by evanescent waves [P. L. Marston, A. L. Espana, C. F. Osterhoudt, and D. B. Thiessen, J. Acoust. Soc. Am. 122, 3034 (2007)]. The backscattering amplitude from targets with localized coupling displayed a spatial decay rate approximately twice that of the evanescent wave. An extended reciprocity relation was proposed which accounts for the more general case of a bistatic observation in the water column above the sea floor. In the bistatic case the spatial decay rate may differ from the case of backscattering. The present research concerns the testing of generalized reciprocity for small circular cylinders using two-dimensional finite elements. The calculated spatial decay rate for low-frequency bistatic scattering follows the generalized reciprocity condition when the predicted decay rate (for the specified observation scattering angle) exceeds the spatial decay rate of the incident evanescent wave. This computational result includes agreement with the double decay-rate case of backscattering. The calculations indicate that bistatic observation can significantly reduce the spatial decay rate of the signal dependence on burial depth. [Work supported by ONR.]
Long-range low frequency shallow water acoustic propagation and bottom penetration experiments for underwater unexploded ordinance.
Authors: Simpson Harry J. , Saniga Mike L. , Houston Brian H. , Berdoz Alain R. , Frank Philip , Liskey Steve W. , Kraus Larry A.
Published: October 2013

Among the many issues now being explored in the acoustic detection underwater unexploded ordinance (UXO) in shallow water is the impact of propagation and bottom interaction. Here, we report on long range (1 km) propagation measurements in a littoral environment at the sediment-water interface in the 1 kHz–12 kHz frequency band. The water channel studied was an 8 m deep arm of St. Andrews Bay, Panama City, Florida. The medium grain sandy bottom had a compressional waves peed of 1700 m/s, and the wavespeed in the water was 1539 m/s. A dense array of point pressure measurements (1 cm vertical sampling) were acquired for a synthetic vertical aperture starting from 2 m above and going through 2 m below the interface and at three ranges, 78, 485, and 990 m. The results show deep penetration into the sediment at long ranges for frequencies below 5 kHz. The data are examined using a two-fluid parabolic equation model, and the overall environmental acoustics are discussed including the nature of sediment penetration due to evanescent waves, multipath, and interface roughness.
Did paleolithic cave artists choose resonant locations for paintings?
Authors: Lubman David
Published: October 2013

Paleolithic artists were surely impressed by the acoustical properties of their cave dwellings. Since many inhabited caves and grottoes were highly reverberant, it is compelling to speculate that cave acoustics importantly influenced ancient artists. One imaginative suggestion is that Paleolithic artists intentionally chose the most resonant locations for painting. This thesis was proposed by Iegor Reznikoff and Michel Dauvois in the Bulletin de Ia Societe Prehistonque Francaise 85, 238–246 (1988) and was engagingly described by British archaeologist Chris Scarre [“Painting by resonance,” Nature (London) 338, 382 (1989)]. Scarre found this thesis consistent with “the likely importance of music and singing in the rituals of our early ancestors.” The researchers found a greater density of paintings where sonic resonances were judged strongest. However, that finding may be spurious: the same result is predictable from painterly rather than sonic considerations. Painters would likely choose nonporous rather than porous rocks to conserve pigment. Since nonporous rocks are generally highly sound reflective, nonporous rock locations tend to be highly resonant. Since porous rocks tend to absorb sound, porous rock locations are less resonant. Thus the higher density of cave paintings found at resonant locations may not imply the artists’ preference for resonance.
Tehran concert hall—a virtual specimen indicative the conversion processes from architectural design to acoustical remedies.
Authors: Azad H.
Published: October 2013

Over the last year of the B.Sc. period it was investigated by the author to design a 1200-seat concert hall as the thesis project. Now the quoted project has been enriched by the acoustical aspects of the concert hall design regarding the acquired capabilities during the M.Sc. studies. It has become prevalent these days to peruse and analyze the acoustical parameters of enclosures through the different acoustical simulation software such as Odeon 9, CATT-Acoustics 8, EASE 4.1, and Ramsete 2.5, either before or after their construction as an objective assessment but interpreting this data is a real challenge. [H. Azad. “Ali Qapu: Persian Historical Music Room?”. Auditorium Acoustics Conference (2008), Oslo.] The variable height stage canopies taking into account their shape and materials is investigated by calculations with ray-tracing computer model to provide a large part of the audience with early reflections and to create good ensemble conditions for the musicians on stage. The diffusers design as a geometrical shape with design factors of height, surface coverage and orientation in addition to their location will be studied computer aided to improve the acoustics in this virtual concert halls.
Signal processing challenges in the analysis of stereotyped killer whale calls produced in social contexts.
Authors: Bowles Ann E. , Crance Jessica L. , Nash Juliette S.
Published: October 2013

Research on call development and usage by individual killer whales has been limited by signal processing challenges, which include (1) more than one simultaneously operating source, (2) a pulse-burst production mechanism, (3) a complex repertoire (7–17 calls), (4) gradation with nonstereotyped call types, (5) presence of nonstereotyped features encoding state (e.g., arousal), and (6) vocal matching among callers. To date, the most successful signal processing efforts have targeted tonal, modulated “screams.” However, analysis of other component types (bursts, resting call), parsing of calls into components, and reliable classification of calls continue to require the intervention of a human brain. Under controlled laboratory conditions, using data from seven whales at SeaWorld San Diego, these characteristics were analyzed by ear and using traditional statistical methods. The work showed that important insights could be gained by comparing these call characteristics among individuals and social contexts. However, estimates of the sample sizes needed to show significant differences among individuals suggest that automation will be essential. Use of the human ear as a signal processor could also introduce biases. Efficient automation would greatly improve the study of individual call usage. [Work supported by the Hubbs-SeaWorld Society and SeaWorld San Diego.]
Rhythmic analysis of sperm whale broadband acoustic signals.
Authors: Sidorovskaia Natalia , Schexnayder Philip , Ekimov Alexander , Sabatier James , Ioup George E. , Ioup Juliette W.
Published: October 2013

Rhythmic analysis of sperm whale clicks, based on the static Fourier transform, was previously reported in the literature as a tool for extracting the acoustic signature of individual sperm whales [Andre and Kamminga, J. Mar. Biol. Ass. U.K. (2000)]. A new approach, originated from the algorithm developed for human motion analysis [Sabatier and Ekimov, “A Review of Human Signatures in Urban Environments Using Seismic and Acoustic Methods,” Proc. IEEE (2008)], is applied to study time-dependent periodicity in sperm whale signals recorded in the Gulf of Mexico. The new approach for rhythmic analysis of sperm whales clicks, which are used for navigation purposes, shows the presence of stable temporal periodicity that could possibly be attributed to the acoustic portrait of individuals. [Research supported in part by SPAWAR.]
Sei whale localization and vocalization frequency sweep rate estimation during the New Jersey Shallow Water 2006 experiment.
Authors: Newhall Arthur E. , Lin Ying-Tsong , Lynch James F. , Baumgartner Mark F.
Published: October 2013

The Shallow Water 2006 (SW06) experiment was conducted in the Mid-Atlantic continental shelf off the New Jersey coast. A fast-sampling, 48-channel hydrophone array system recorded a number of sei whale (Balaenoptera borealis) vocalizations during this time. This system had 16 hydrophones on the vertical line array (VLA) component covering the water column from 13-m depth to the bottom (80 m) and 32 hydrophones on the 480-m horizontal line array (HLA) component that lay on the bottom. The sei whale receptions on the arrays are low-frequency (less then 100 Hz) downsweep chirps and have the typical acoustic modal arrival pattern seen in shallow-water, low-frequency sound propagation. Due to acoustic modal dispersion, the frequency sweep durations observed from the receptions are found to be longer than the original signal. A horizontal beamforming technique using the HLA component is implemented to determine the bearings to whales. The whale location along the determined bearing and the frequency sweep rate of original whale sound are simultaneously estimated using the VLA component with a multiple-parameter optimization technique. This optimization technique is based on acoustic normal mode theory and is designed to compensate for the effect of acoustic modal dispersion.
Quantifying noise in animal acoustic signals.
Authors: Riede Tobias
Published: October 2013

Variability in noisy calls or call segments is often difficult to quantify but critical in order to understand animal acoustic communication. Such components represent a large proportion of animal acoustic signals and they are behaviorally meaningful. Such elements have also collectively been labeled as nonlinear phenomena. Three approaches are reviewed. The harmonics-to-noise ratio (HNR) can be related to perceptual characteristics of an acoustic signal as well as to the mechanism of its production. The HNR seems to be a useful numeric parameter to quantify vocal variability. Two problems with the procedure will be outlined. The nonlinear measure (NLM) is designed to provide an overall estimate of the strength of nonlinearity in a signal. The rationale is that if signal noise reflects low-dimensional chaos as produced through nonlinear processes, deterministic-nonlinear modeling will produce a relatively small error component. A stochastic linear model, on the other hand, will produce a relatively large error component when applied to such signals. In a third approach, utterances can be segmented in the temporal domain into harmonic and noisy components for separate analyses. Noise-specific procedures are used to analyze nonlinear phenomena. Advantages and disadvantageous of all three procedures are exemplified with sounds from mammals and anurans.
Hidden Markov models for the analysis of animal vocalizations.
Authors: Clemins Patrick J. , Johnson Michael T.
Published: October 2013

Hidden Markov models (HMMs) are commonly used to model sequential data, including time series such as vocalizations. HMMs are uniquely suited to model time series because of their ability to implicitly align samples and to be linked together to create flexible recognition patterns for vocalizations with repeating patterns, such as bird and whale songs. HMMs require less human interaction and manual tuning for successful recognition and are inherently more noise resistant than template-based techniques. Due to their statistical basis, HMM-based recognition systems can incorporate additional statistical models. One such example is a language model, which can be designed to include a priori knowledge about the structure of the vocalizations, such as syllable repetition patterns. HMMs have become the most popular recognition model in speech processing, and this experience has been applied to a variety of vocalization analysis tasks including individual identification, song classification, call type recognition, and physiological state determination. Classification results from complete HMM-based systems for a variety of tasks and species will be discussed.
Improved impulse response for hydrophone measurements in therapeutic ultrasound fields.
Authors: Canney Michael S. , Khokhlova Vera A. , Sapozhnikov Oleg A. , Pishchalnikov Yuri A. , Maxwell Adam D. , Bailey Michael R. , Crum Lawrence A.
Published: October 2013

The accurate measurement of pressure waveforms in high intensity focused ultrasound (HIFU) fields is complicated by the fact that many devices operate at output levels where shock waves can form in the focal region. In tissue ablation applications, the accurate measurement of the shock amplitude is important for predicting tissue heating since the absorption at the shock is proportional to the shock amplitude cubed. To accurately measure shocked pressure waveforms, not only must a hydrophone with a broad bandwidth (>100 MHz) be used, but the frequency response of the hydrophone must be known and used to correct the measured waveform. In this work, shocked pressure waveforms were measured using a fiber optic hydrophone and a frequency response for the hydrophone was determined by comparing measurements with numerical modeling using a KZK-type equation. The impulse response was separately determined by comparing a measured and an idealized shock pulse generated by an electromagnetic lithotripter. The frequency responses determined by the two methods were in good agreement. Calculations of heating using measured HIFU waveforms that had been deconvolved with the determined frequency response agreed well with measurements in tissue phantom. [Work supported by NIH DK43881, NSBRI SMST01601, NIH EB007643, and RFBR.]
Development of a high intensity focused ultrasound (HIFU) hydrophone system.
Authors: Schafer Mark E. , Gessert James
Published: October 2013

The growing use of high intensity focused Ultrasound (HIFU) has driven a need for reliable, reproducible measurements of HIFU acoustic fields. A reflective scatterer approach, incorporating several novel features for improved bandwidth, reliability, and reproducibility has been demonstrated [M. E. Schafer, J. Gessert, and W. Moore, Proc. IEEE Ultrasonics Symposium, 1739–1742 (2005)]. Several design improvements which have increase the signal-to-noise ratio of the system, and potentially reduced the cost of implementation, are now presented. For the scattering element, an artificial sapphire material is used to provide a more uniform radiating surface. The receiver is a segmented, truncated spherical structure with a 10 cm radius, made from 25 micron thick, biaxially stretched PVDF, with a Pt-Au electrode on the front surface. A specialized backing material provides the stiffness required to maintain structural stability, while at the same time providing both electrical shielding and ultrasonic absorption. This new receiver design has improved the noise performance by 8–12 dB; the new scattering sphere has reduced the scattering loss by another 14 dB, producing an effective sensitivity of −298 dB re 1 microVolt/Pa. The design trade-off still involves receiver sensitivity with effective spot size. However, the reduced cost and improved repeatability makes the overall design more robust for routine HIFU system measurements.
Characterization of high-intensity focused ultrasound systems by extrapolating from field measurements conducted in the quasi-linear regime.
Authors: Howard Samuel M.
Published: October 2013

The purpose of this talk will be to discuss a method for characterizing HIFU fields where hydrophone measurements are made by reducing the electrical power input to the system until quasi-linear field propagation conditions are achieved, and until the pressure levels are safe for the hydrophone used. Radiation force measurements of total power are then used to scale up the results. This allows for reliable measurements which do not damage the hydrophone, and avoids complications from involving the many harmonics which can be generated in water which are usually not so strongly present in tissue. Advantages and disadvantages of the method will be discussed.
Ex vivo monitoring of high intensity focused ultrasound treatment.
Authors: Zheng Xinliang , Vaezy Shahram
Published: October 2013

High intensity focused ultrasound (HIFU) is a therapeutic modality that can produce coagulative necrosis of tissues noninvasively. The present study investigated the feasibility of monitoring lesion formation during HIFU treatment using ultrasound backscattered signal, specifically for treatments that do not result in hyperecho in ultrasound images (i.e., thermal lesion with no obvious boiling involved). Ex vivo chicken breast tissue was used and a sequence of two-dimensional radiofrequency (RF) signal was collected for each exposure as a function of time. For each frame in the RF sequence, one-dimensional cross correlation coefficient was calculated between every A-mode line of RF signal and the same line 5 seconds later. A series of experiments was performed with different HIFU treatment durations. The results showed that there was a decrease in the maximum coefficient value in the focal region during HIFU exposure, and a threshold value could be determined to be associated with the time that a thermal lesion was formed. Tissue dissection was performed to confirm the presence of a thermal lesion. This RF analysis technique has shown promise in detecting location and dimension of thermally-induced lesions, as well as being capable of real-time treatment monitoring to guarantee effective and efficient HIFU therapy.
Improved range-discriminating ultrasonic vibrometer.
Authors: Martin James , Gray Michael , Rogers Peter
Published: October 2013

Improvements have been made to a previously reported continuous-wave ultrasonic vibrometer with range discrimination capabilities. In its current realization, the vibrometer incorporates a high-speed 16-bit arbitrary waveform generator as a signal source. The waveform generator shares a sample clock with the data acquisition system so that both the real and apparent spectral purity of each of the tones in a multitone carrier signal is preserved in the measurement. The new configuration provides increased control over the characteristics of the transmitted signal. It permits the exploitation of the full bandwidth capabilities of all of the hardware components of the system by removing a limit to the number of tones in the carrier. It also permits the relative phases of individual tones to be optimized in order to reduce the crest factor of the transmitted signal and better exploit the dynamic range of other system components. A further improvement in dynamic range has been achieved with an analog subtraction of the transmitter-receiver crosstalk, which is normally the largest received-signal component, prior to digitization. The improved vibrometer has been used to image shear wave fields in both tissue phantoms and inert tissue samples. [Work supported by ONR.]
Noise mapping in Portland Oregon: The need to develop an approach to accurately depict the health impacts of short duration automotive racing noises emanating from a variety of motorized race events at the Portland International Racetrack facility.
Authors: van Orden Paul
Published: October 2013

The City of Portland Oregon is in the final phase of completing its efforts to map one quadrant of the city in conjunction with the Greenbusch Group of Seattle Washington. The selection of this section of the city was identified for its notable concentration of noise sources ranging from the normal transportation related road, railway, and international airport sounds, to the unique and constantly varying tonal quality of automobile and motorcycle race groups at an international automotive raceway facility. The challenge to identify an appropriate method or standard to accurately depict the impact of these short term track races and drag races offers a unique opportunity to expand on the value of the noise mapping approach for long term urban planning efforts.
Prediction of noise from Portland International Raceway using environmental noise computer modeling techniques.
Authors: Deuchars Angus M.
Published: October 2013

An environmental noise computer model of Portland International Raceway has been constructed to evaluate noise mitigation at local noise sensitive receptors from making site wide changes to raceway infrastructure. There is little precedent of environmental noise computer modeling of raceways, and this paper presents the methodologies that were used to develop the computer model. Field measurements were made at the racetrack according to European Union Harmonoise methods to determine sound power levels of race vehicles; noise predictions were made using ISO 9613 pt 2 prediction model implemented using SOUNDPLAN software. Results from field measurements and predictions are presented to demonstrate accuracy of the computer model. Additionally, the innovative steps used to construct the computer model using sound plan are also presented.
A revised standard method for ground impedance measurement.
Authors: Attenborough Keith , Taherzadeh Shahram
Published: October 2013

ANSI S1.18 (1998) for ground impedance measurement has been revised to combine the previously existing template method with direct deduction of impedance from data for the complex level difference between vertically separated microphones. It has been found that, despite the improved sensitivity at low frequencies that should be achievable through use of a longer range, the combined approach ceases to give reliable information for horizontal source-receiver separations of more than 1 m. The revised standard contains four worked examples: three grass-covered ground surfaces and a gravel road. The first example has been used to show that although use of consecutive measurements with a single microphone produces data that is usable with the template method, the phase information is not sufficiently robust to allow direct impedance deduction. In the three grassland examples, the widely used single parameter model is found to overpredict the real and imaginary parts of ground impedance at lower frequencies. The fourth example shows that the geometries prescribed in the standard are not appropriate for determining impedance spectra of relatively hard surfaces but are sufficient to establish that the surface is acoustically hard.
Construction noise action area model.
Authors: Magnoni Larry J.
Published: October 2013

A model developed by the Washington State Department of Transportation to be used by biologists to determine the action area for highway construction projects is presented. The model conservatively estimates the distance at which construction noise would match ambient background sound levels. The model propagates the combined maximum sound level (Lmax) for project construction equipment spherically over the soft and hard site characteristics to a point where it would converge with the ambient hourly equivalent (Leq) sound level. The ambient Leq sound level can be input into the model from actual measurements, documented reference sources, or from a table based on population density [Federal Transit Authority (FTA) Noise Assessment Guidance (2006)]. The model then combines the ambient sound level with the traffic sound level based on the national mean emission curves. The input for traffic volume, vehicle type, and speed is propagated cylindrically from the source until the traffic source drops to the ambient level. A macro, that applies a general atmospheric and molecular absorption adjustment to the calculated distances, produces the action area circumference both graphically and numerically as it relates to each of the sound sources. [Work supported by the Federal Highway Administration (FHWA) (1978).]
Leakage identification for a vehicle firewall sound transmission by beamforming technique.
Authors: Ribeiro Yuri , Fonseca William , Samir Gerges
Published: October 2013

The acoustic phased array is a powerful tool for identification and quantification of noise sources on complex systems. Acoustic sources generate waves which propagate through space to the N microphones of the array. The acoustic pressures are sampled and converted to time series data and processed using either time domain or frequency domain delayed and averaged. The output of signal processing is a colored image with relative sound pressure levels distribution. In this paper application of beamforming technique is used to identify leakage areas in a vehicle firewall. Two acoustic rooms are used and the firewall was installed in the apertures inbetween with careful control of noise flank paths. Source room was excited with white noise and a phased array microphone was used to measure the sound field in the anechoic receiving room. The firewall was evaluated both with all holes sealed and with the regular pass-through components installed. Two different acoustic insulators were also tested on the firewall. The results show clearly the areas of leakage with relative noise levels at each test configuration.
Synthesizing the shape of sand grains.
Authors: Barclay David R. , Buckingham Michael J.
Published: October 2013

Digitized outlines of sand grains from a dozen locations have been acquired using an optical microscope. A Fourier decomposition of the outline is calculated providing a spectral description of the grain’s shape. By averaging over several hundred grains, the normalized power spectrum of each sand sample is returned. The desert sands, beach sands, and marine sediments measured all exhibit the same inverse-power-law dependence on the harmonic number, n, varying as n−10/3 for 2⩽n⩽20. This “universal” spectrum provides the basis of a numerical technique for synthesizing the irregular outline of a sand grain: the outline is represented as a random pulse train in which identically shaped microasperities, with normally distributed amplitudes, are randomly superimposed on the perimeter of a circle. Carson’s theorem links the power spectrum of an individual microasperity to that of the real grains and constrains the synthetic outlines to have the identical statistical properties as the outlines of the measured samples. Visually, it is difficult to distinguish between the synthesized and real outlines. This numerical technique for synthesizing irregular outlines of grains has potential for investigating the random-packing of realistically rough particles through computer simulation. [Work supported by the Office of Naval Research.]
Uncertainty estimation of sound attenuation in marine sediments at low frequencies.
Authors: Jiang Yong-Min , Chapman N. Ross
Published: October 2013

Marine sediment attenuation at low frequencies (under 5 kHz) is generally difficult to be directly measured by in situ probes embedded in the sediment, partly due to the very short propagation distances. An alternate experimental technique is to use single bottom bounce signals received by a vertical line array. The frequency dependence of the sediment attenuation is first obtained by comparing the amplitude differences of the sea floor reflection and the sub bottom layer reflection at different frequencies. The absolute attenuation is then obtained by using the previously estimated sound speed and layer thickness. Inherently there is uncertainty introduced in each stage of the attenuation estimation procedure. To evaluate the uncertainty of the attenuation estimate, the standard deviation of the signal fluctuation is mapped to the intermediate result first, and then Bayesian inversion results of the sound speed and the layer thickness are included in the final attenuation estimates. This uncertainty analysis is demonstrated by the estimation of the sediment attenuation from the low frequency chirp data collected in a variable water column environment in the Shallow Water 06 experiment. [Work supported by ONR Ocean Acoustics.]
On the use of acoustic particle motion in geoacoustic inversion.
Authors: Crocker Steven E. , Miller James H. , Hines Paul C. , Osler John C.
Published: October 2013

Geoacoustic inversion estimates sediment properties based on one or more parameters of an observed acoustic field via inverse mathematical methods. The observed acoustic parameter is usually derived from the acoustic pressure measured at one or more locations. Recent advances in acoustic sensor technology have enabled the simultaneous measurement of the acoustic pressure and particle motion in three dimensions. The additional information provided by such acoustic vector sensors can be used to improve the performance of existing and novel geoacoustic inversion techniques. Current research seeks to use the additional information that is provided by the acoustic vector sensor to pose new inverse problems for the estimation of seabed sediment properties. In particular, data collected during the Sediment Acoustics Experiment 2004 (SAX04) included acoustic pressure and particle acceleration from a small number of acoustic vector sensors arranged in a vertical line, spanning the water-sediment interface. A variety of waveforms were transmitted into the seabed to measure the direct, reflected, and transmitted waves. Experiment details, forward modeling, inversion methods, and results are discussed. [Work supported by ONR.]
Intelligibility of foreign-accented speech in noise for younger and older adults.
Authors: Ferguson Sarah Hargus , Keum Kyung Ae , Jongman Allard , Sereno Joan A.
Published: October 2013

In a previous experiment [Ferguson et al., J. Acoust. Soc. Am. 118, 1932 (2005)], young listeners with normal hearing, older adults with essentially normal hearing, and older adults with hearing loss were found to be similarly affected by the presence of a foreign accent on a word identification task in various listening conditions. This result stands in sharp contrast with an extensive literature demonstrating that the negative effect of speech distortions such as noise, reverberation, and time compression is disproportionately great for older versus younger adults. The conclusions of the previous study, however, were tempered by the presence of floor effects in the data for the older adults with hearing loss identifying accented speech in noise (12-talker babble at a signal-to-babble ratio of +3 dB). The present experiment extended the earlier one by presenting the same monosyllabic word materials to new groups of young normal, elderly normal, and elderly hearing-impaired listeners in three new signal-to-babble ratios: +6 dB, +9 dB, and +12 dB. The results will be discussed in terms of their implications for older adults interacting with foreign-accented speakers in healthcare settings. [Project supported by University of Kansas General Research Fund.]
The aquisition of sonorant voicing.
Authors: Sieczkowska Jagoda
Published: October 2013

In the view of Exemplar Theory phonetic variants of words are stored by speakers in their memory as a set of exemplars organized into clusters and grow stronger or decay over time depending on language experience (Bybee, 2002). In Polish and French sonorant consonants in clusters are devoiced word finally, which does not occur in German and English because of phonotactical constraints. A study based on automatic speech alignment and voicing profile extraction from Polish, French, German, and English speech corpuses aims at defining voicing sonorant temporal structure and voicing profile acquisition. The hypothesis is that Polish and French native speakers transfer their L1 exemplars during L2 acquisition of English and German and thus devoice sonorant. After extracting voicing profiles from recordings of native Polish and French speakers speaking English and German as an L2/L3, comparison of voicing profiles of English and German native speakers will be made in order to define the exemplar transfer.
Effectiveness of a perceptually weighted measure on automatic evaluation of a foreign accent in a second language.
Authors: Kato Hiroaki , Nakamura Shizuka , Matsuda Shigeki , Tsuzaki Minoru , Sagisaka Yoshinori
Published: October 2013

An empirical study is carried out to achieve a computer-based methodology for evaluating a speaker’s accent in a second language as an alternative to a native-speaker tutor. Its primary target is the disfluency in the temporal aspects of an English learner’s speech. Conventional approaches commonly use measures based solely on the acoustic features of given speech, such as segmental duration differences between learners and reference native speakers. However, our auditory system, unlike a microphone, is not transparent: it does not send incoming acoustic signals into the brain without any treatment. Therefore, this study uses auditory perceptual characteristics as weighting factors on the conventional measure. These are the loudness of the corresponding speech segment and the magnitude of the jump in loudness between this target segment and each of the two adjacent speech segments. These factors were originally found through general psychoacoustical procedures [H. Kato et al., JASA, 101, 2311–2322 (1997); 104, 540–549 (1998); 111, 387–400 (2002)], so they are applicable to any speech signal despite the difference in language. Experiments show that these weightings dramatically improve evaluation performance. The contribution of psychoacoustics to evaluation methodology of second language speech is also discussed. [Work supported by RISE project, Waseda Univ. and KAKENHI 20300069, JSPS.]
Articulatory analysis of foreign-accented speech using real-time MRI.
Authors: Tepperman Joseph , Bresch Erik , Kim Yoon-Chul , Goldstein Louis , Byrd Dani , Nayak Krishna , Narayanan Shrikanth
Published: October 2013

We present the first study of nonnative English speech using real-time MRI analysis. The purpose of this study is to investigate the articulatory nature of “phonological transfer”—a speaker’s systematic use of sounds from their native language (L1) when they are speaking a foreign language (L2). When a non-native speaker is prompted to produce a phoneme that does not exist in their L1, we hypothesize that their articulation of that phoneme will be colored by that of the “closest” phoneme in their L1’s set, possibly to the point of substitution. With data from three native German speakers and three reference native English speakers, we compare articulation of read phoneme targets well documented as “difficult” for German speakers of English (/w/ and /dh/) with their most common substitutions (/v/ and /d/, respectively). Tracking of vocal tract organs in the MRI images reveals that the acoustic variability in a foreign accent can indeed be ascribed to the subtle articulatory influence of these close substitutions. This suggests that studies in automatic pronunciation evaluation can benefit from the use of articulatory rather than phoneme-level acoustic models. [Work supported by NIH.]
Automatic auditory discrimination of vowels in simultaneous bilingual and monolingual speakers as measured by the mismatch negativity (MMN).
Authors: Molnar Monika , Baum Shari , Polka Linda , Menard Lucie , Steinhauer Karsten
Published: October 2013

MMN responses reflect whether language users have developed long-term memory traces in response to phonemes and whether they are able to perceive small acoustic changes within speech sound categories. Subtle acoustic changes within phonemes are often irrelevant to monolingual perceivers, but can be crucial for bilingual perceivers if the acoustic change differentiates the phonemes of their two languages. In the present study, we investigated whether bilinguals are sensitive to such acoustic changes. We recorded MMN responses from monolingual (English, French) and simultaneous bilingual (English/French) adults using an auditory oddball paradigm in response to four vowels: English [u], French [u], French [y], and an acoustically-distinct (control) [y]. In line with previous findings, monolinguals were more sensitive to the phonemic status of the vowels than to the acoustic properties differentiating the sounds. Bilingual speakers revealed a different pattern; they demonstrated overall slower discrimination responses to all sounds, but showed almost equal sensitivity to phonemic and phonetic/acoustic differences. The results suggest that bilingual speakers exhibit a more flexible but less uniquely-specified perceptual pattern compared to monolingual speakers.
Assessing phonetic talent in second language performance: The relationship between perception abilities in the native and the second language and production.
Authors: Jilka Matthias , Rota Giusy , Reiterer Susanne
Published: October 2013

An extensive project that aims to assess innate phonetic talent is introduced. It investigates language performance of 103 native speakers of German in their native language, in English and (to a limited degree) in Hindi. The project’s original priority is to establish a talent score that serves as the basis for a neurolinguistic study which attempts to find correlates between phonetic talent and brain anatomy/function. For this purpose tasks investigating neurobiological, psychological, and other factors influencing performance are also conducted. Innate talent is to be distinguished from superficial proficiency by limiting or eliminating the influence of interfering factors such as language experience. The test subjects are investigated with respect to different manifestations of their production (e.g., quasi-spontaneous speech, reading), perception (e.g., discrimination, identification), and imitation abilities in both German and English. A first analysis of the interactions between all perception and production scores shows a significant correlation at the level of p < 0.05. The correlation is especially high (p < 0.01) between perception tasks involving the interpretation of intonational features of both English and German and production abilities (again both in German and English), suggesting a certain cross-language predictive power.