bluuebunny/crossref_metadata_2025 huggingface dataset:
Multipitch tracking and speaker separation for two or more speakers.
Authors: Vishnubhotla Srikanth , Espy-Wilson Carol
Published: October 2013

Accurate robust pitch tracking in multispeaker environments is an important issue in monaural speech separation. This research presents a spectrotemporal multipitch algorithm designed to detect voiced unvoiced regions in a mixture of multiple speakers, to identify the number of speakers in voiced regions, and to yield the pitch estimates of voiced speakers. Pitch estimation is based on identifying the minima of the multidimensional (MD) AMDF. The number of voiced speakers is determined by the temporal evolution of the MD AMDF. The algorithm has been developed and tested for the two-speaker case and is being extended to the three-speaker case. Evaluation on a framewise basis for the two-speaker case yielded an insertion error rate of 1.78%, a deletion error rate of 14.37%, and a substitution error rate of 3.26%, comparable to the state of the art. Pitch estimation errors resulted primarily from speaker domination. The extension to the three-speaker case shows promising preliminary results. The proposed algorithm can contribute to speaker separation by helping to identify spectrotemporal regions dominated by one of the speakers from those that have comparable contribution from all speakers. Appropriate strategies can then be used to separate speech in these different scenarios. [Research supported by NSF Grant BCS-0519256].
A survey of respiratory system behavior during pauses in spontaneous speech.
Authors: Slifka Janet
Published: October 2013

This work is part of an ongoing study to characterize respiratory system involvement during the generation of pauses in connected speech. At pauses, the speaker takes a breath or the speaker does not take a breath. Previously reported results for read speech observed that without-breath pauses were generated with a sharp movement toward net inspiratory effort followed by a sharp return toward increased net expiratory effort [J. Slifka, In Dynamics of Speech Production and Perception (IOS, 2006), pp. 45-58]. In the present work, 52 utterances of spontaneous speech from four speakers were analyzed. Behaviors in spontaneous speech include not only the activity similar to that observed for read utterances but also additional types of behaviors. Some without-breath pauses were observed to have regions of oscillation in net effort. Such behavior appears to be more common during longer pauses and during hesitations. Secondly, some without-breath pauses were generated with a reversed pattern—sharp movement toward increased net expiratory effort and back to lesser net expiratory effort. This pattern was much less common and appears to be associated with relatively short pauses in regions of hesitations. These results characterize the respiratory system as an active component in speech production. [Work supported by NIH-NIDCD DC007986]
Measuring glottal area and flow in an excised canine larynx model using stereoscopic particle imaging velocimetry.
Authors: Gutmark Ephraim , Murugappan Shanmugam
Published: October 2013

As opposed to the glottal area waveform, the opening and closing phases of glottal flow are asymmetric. Near the end of the closing phase, flow exiting the glottis has a rapid deceleration not seen during the opening phase; this “skewing” of the flow waveform is important for producing loudness and higher frequency harmonics. The major mechanism for the skewing of the flow waveform, relative to the area waveform, has been attributed to the nonlinear interaction between the vocal tract and the glottal flow. Previous work from this laboratory supports a hypothesis that both area and flow skewing are seen even without the vocal tract due to intraglottal flow separation vortices producing relative negative pressure. To help test this hypothesis in an animal model, a technique is necessary that can reliably measure glottal area and flow. In this work, we will show results for both glottal area and flow during phonation, in excised canine larynges, using stereoscopic particle imaging velocimetry and high speed visualization. The results support the hypothesis that intraglottal flow separation can be one mechanism for producing both flow and glottal skewing. [Work supported by NIDCD 5K08DC005421.]
Unsteady laryngeal airflow simulations: An analysis of the generated intraglottal vortical structures.
Authors: Mihaescu Mihai , Khosla Sid , Gutmark Ephraim
Published: October 2013

Unsteady flow simulations in diffuserlike vocal-fold models proved that laryngeal airflow generates vortical structures in the intraglottal region. However, the features of these vortices and their influence on the vocal-fold motion were not clearly addressed. The present work characterizes the intraglottal vortical structures developed during the closing phase of the phonation cycle in order to analyze their possible influence on the voice quality. It is shown that intraglottal vortices are formed on the divergent slope of the glottis, just downstream of the separation point. The core of the vortical structures is characterized by important negative static pressure values (lower as compared with the surrounding pressure field). These vortices gain strength and increase in size as are convected downstream by the flow. The mechanism for which the intraglottal flow structures are becoming stronger is attributed to the entrained air from the supraglottal region. The instantaneous pressure loads on the divergent glottal slope are not uniform (in both time and space) and dependent on the vortical structures traveling nearby the glottis. The negative static pressures associated with the intraglottal vortical structures suggest that the closing phase during phonation may be accelerated by such vortices.
Articulatory timing in sentence production in young normal speakers and speakers with apraxia of speech: A speed history analysis.
Authors: Taguchi Kana , Hashi Michiko , Odell Katharine
Published: October 2013

Apraxia of speech (AOS) is typically considered as characterized by kinematic timing problems among articulators; however, kinematic descriptions of such phenomenon are scarce. The study describes kinematic timing based on a speed history of the lower lip and the jaw, across speaking rates in young normal speakers and speakers with AOS. Kinematic data were acquired through the x-ray microbeam system, and data of young normal speakers were taken from the x-ray microbeam speech production database. A short sentence was the speech material/target. Acoustic correlates of major speed peaks of the jaw and the lower lip, as well as their temporal relationships with each other, were investigated. Of particular interest was how changes in speech rate affect such temporal relationships in normal speakers and speakers with AOS. The goal of this study was to establish a description method of inter- and intra-articulator timing for use in analyses of point-parametrized articulatory data in sentence-level materials.
Perceptual cues for consonant identification.
Authors: Li Feipeng , Allen Jont
Published: October 2013

This research quantitatively explores the perceptual cues of initial consonants by using psychoacoustical methods. Speech sounds are encoded by across-frequency temporal onsets called events. To determine the time-frequency importance function of the consonant sounds, speech stimuli (16 nonsense CVs from the LDC-2005S22 database) are high-pass or low-pass filtered and time-truncated before being presented to normal hearing listeners. Databases of speech perception under various signal to noise ratio (SNR) conditions are constructed to investigate the effect of noise on speech recognition. A visualization tool that simulates the auditory peripheral processing, called the AI-gram, is used for the analysis of the speech events under various SNR conditions. To verify the nature of the events, a special software has been developed to convert one sound into another, starting from real speech sounds, by removing primary noise robust cues. In pilot studies with a hearing-impaired subject, it is shown that feature boosting improves the robustness of select speech consonants to noise.
The effect of articulatory placement on acoustic characteristics of nasalization.
Authors: Rong Panying , Kuehn David P.
Published: October 2013

Vowels adjacent to nasal consonants (/m/, /n/, /ŋ/ in English) tend to be nasalized. The effect of velopharyngeal opening on vowel nasalization has been studied by Feng et al. (1996) and Pruthi et al. (2007) with a transmission line model of the vocal tract based on MRI data of the area function. In order to find out the effect of articulatory placement on acoustic properties of nasalized vowels, the current study simulated the transfer function of coarticulated vowels in different nasal-vowel utterances. The results revealed different acoustic characteristics of the same vowel in various nasal contexts, which demonstrated the effect of articulatory placement on vowel nasalization and suggested the possibility for oral articulation to compensate for spectral changes caused by failure of velopharyngeal closure among hypernasal patients. The articulatory parameters for compensatory articulation were optimized by minimizing the spectral differences (first four formants) between the compensated nasalized vowel and its oral counterpart. The nasalized vowel was then synthesized with an articulatory synthesizer and evaluated for vowel quality and nasality in the following perceptual study.
The effects of coarticulatory timing and lexical frequency on vowel nasalization in English: An aerodynamic study.
Authors: Bishop Jason B.
Published: October 2013

Recently it has been suggested that a certain degree of variability in coarticulatory vowel nasalization is due to variation in the temporal alignment of nasal and oral gestures for N. In particular, the extent of vowel nasalization in VNC sequences is shown to be inversely related to the duration of the oral gesture for the nasal. Beddor [nasals and nasalization: the relation between segmental and coarticulatory timing, ICPhS (2007) present acoustic data which suggests this to be the case in English; in environments where nasals are shorter, such as VNC[−voice] versus VNC[+voice], anticipatory nasalization is longer. The present study examines vowel nasalization in such environments and attempts to corroborate the aforementioned acoustic findings with aerodynamic data. Additionally, other possible sources of variation in the extent of anticipatory nasalization are explored, namely, the effects of lexical frequency, which has been claimed to correlate negatively rather than positively with coarticulation [carborough (2004).]
Children form tense and lax vowel classes under aerodynamic constraints.
Authors: Messum Piers
Published: October 2013

In English, lax vowels contrast with tense vowels as follows: (1) They are always “checked” by a following consonant, (2) they require only moderate displacement of the tongue from its resting position, and (3) they are “short.” If these characteristics are arbitrary and independent–as currently believed–then it is a remarkable coincidence that each divides the vowel inventory into classes with the same membership. Alternatively, these characteristics emerge together under constraints in child speech that do not appear in the adult model. Aerodynamically, the pressures in child speech are higher, flows are similar, and airways are smaller. Mechanically, the respiratory drive that supports speech is pulsatile rather than smooth. This latter difference is heightened in stress-accent languages, such as English, where a child must reinforce pulses for greater loudness on stressed syllables. The constraints created by these factors require a child to check vowels made with open articulations and lengthen those made with close ones. The former behavior protects the subglottal pressure head. The latter is the indirect result of limiting airflow to avoid unwanted turbulent noise at the point of maximum oral constriction. Increasing laryngeal resistance to do this prolongs the time it takes to dissipate a pulse.
Interarticulator timing in a sentence production in young normal speakers and speakers with apraxia of speech.
Authors: Fujishita Mayuko , Hashi Michiko , Odell Katharine
Published: October 2013

Apraxia of speech (AOS) is typically considered as characterized by kinematic timing problems among articulators; however, kinematic descriptions of such phenomenon are scarce. The goal of this study was to describe articulatory timing between the lips and jaw and the lips and tongue in young normal and AOS speakers across three speaking rates, using x-ray microbeam data. Data of young normal speakers were taken from the x-ray microbeam speech production database. Lip protrusion and tongue elevation timings were derived from the kinematic and acoustic data in the production of “too” in the sentence “The other one is too big” and lip and jaw closing timings for /b/ and /w/ were derived from the kinematic data of the same sentence. The temporal relationships between these articulatory movements in each event were then examined with consideration of the effect of speaking rate. The results are discussed in relation to the effect of speaking rate on interarticulator timing and interspeaker variability in interarticulator timing.
Multitarget tracking using acoustic arrays.
Authors: Costley R. Daniel , Williams Jay E. , Clark Robert C. , Gray Matthew A. , Williams Zachary , Harrington Gary , Frazier William G. , Singleton Jere
Published: October 2013

A tracking system has been developed, which consists of three or more five-microphone arrays. In previous experiments, these arrays have been deployed and used to track broadband, high-speed (subsonic) airborne, acoustic sources. Recent developments to this system have been made so that multiple acoustic sources can be tracked simultaneously. A field test was conducted this past year, which demonstrated this capability by tracking two low-flying learjets. The tracking system consisted of six acoustic arrays spaced approximately 400 m apart. The bearing angles from each array to the acoustic sources were determined from the blind source separator (BSS) algorithm. The BSS utilizes a mathematical propagation model along with an optimization technique, which minimizes the difference between the model and the measured data to yield a bearing to the target. The bearing angles from two or more arrays are transmitted to a master node, which uses this information to estimate the position, speed, and heading of the targets using a tracking algorithm based on the extended Kalman filter algorithm. The field test will be described and the results will be presented and discussed.
A novel approach for the localization of sound sources.
Authors: Rizzo Piervincenzo , Bordoni Giacomo
Published: October 2013

Targeting people or objects by passive acoustic sensors is of relevant interest in several military and civil applications, spanning from surveillance and patrolling systems to teleconferencing and human-robot interaction. To date methods and patents focused solely on the use of beamforming algorithms to compute the time of arrival of sounds detected by using omnidirectional microphones that are sparsely deployed. This paper describes the preliminary results of a novel approach devoted to the localization of ground borne acoustic sources. It is demonstrated that an array made of at least three unidirectional microphones can be exploited to detect the position of the source. Pulse features extracted either in the time domain or in the frequency domain are used to identify the direction of the incoming sound. This information is then fed into a semianalytical algorithm devoted to the identification of the source location. The novelty of the method presented here consists of the use of unidirectional microphones rather than of omnidirectional microphones and of the ability to extract the sound direction by considering features such as the pulse amplitude rather than the pulse arrival time. It is believed that this method may pave the road toward a new generation of reduced size sound detectors and localizers.
Real-time noise source identification using field programmable gate array technology.
Authors: Veggeberg Kurt
Published: October 2013

Acoustic beamforming is usually used as an off-line analysis tool for noise source identification (NSI). The computational requirements of beamforming makes real-time processing difficult to achieve in conventional NSI measurement systems. This limitation prevents conventional NSI measurement systems for applications such as on-line monitoring. This paper presents the theory, design, and development of a real-time NSI measurement system based on reconfigurable input/output field programmable gate array (FPGA) technology. Some techniques used in applying beamforming analysis methods with an FPGA are proposed. Signal processing includes time filtering of input data and spatial beamforming with special treatment to obtain the best result. Pipelining method is used to make full use of the FPGA resources and reach high performance for real-time applications. Testing examples are given to demonstrate the application for NSI.
Directional sources and beamforming.
Authors: Bouchard Christian , Havelock David I. , Bouchard Martin
Published: October 2013

Beamforming is done with an array of sensors to achieve a directional or spatially specific response. It relies on a model of the wave front (source model) arriving at the array to calculate the time delay, or frequency domain phase shift, that must be applied to the signal of each sensor so that they may be summed coherently. Beamforming may be used to improve signal to noise ratio, reduce reverberation, cancel interference, or estimate source location. In this talk the directionality of some real world sources that deviate from an ideal point source is discussed. Performance measures used to evaluate the directivity properties of a beamformer are reviewed. The validity of assuming a point source is examined and challenges for beamforming with nonpoint sources are discussed.
Robust direction-of-arrival estimation by understanding global acoustic scene.
Authors: Mizumachi Mitsunori
Published: October 2013

Direction-of-arrival (DOA) is an important clue in acoustic signal processing. It is, however, difficult to accurately estimate DOAs in the presence of acoustic interference such as background noises and reverberation. The author has proposed a robust DOA finder with an environmental noise model, which describes spectral characteristics of background noises. Knowledge on acoustic interferences helps to make DOA estimation robust, but is difficult to be estimated exactly. The method works well only using the dominant subband components, in which the target signal is distinguished compared with interferences, in some noisy conditions. The noise model is updated time by time based on the DOA estimates to cope with nonstationary noises. Better DOA estimation gives a more accurate noise model and vice versa. In contrast, even a little error starts to cause negative feedback in DOA estimation and finally it goes to the fatal error. This paper proposes to introduce a switching scheme, which evaluates the reliabilities of DOA estimates in a stochastic manner and judge whether the noise model should be updated or not into the previously proposed DOA estimator. It is confirmed that the DOA estimator with the proposed scheme can improve its noise robustness even under nonstationary noises.
Imaging surface roughness with continuous-wave ultrasound reflectometry.
Authors: Mitri Farid G. , Greenleaf James F. , Fatemi Mostafa
Published: October 2013

Measurement of surface roughness irregularities is an important indicator of product quality for many nondestructive testing industries. Many techniques exist; however, because of their qualitative, time-consuming and direct-contact modes, it is of some importance to work out new experimental methods and efficient tools for quantitative estimation of surface roughness. Continuous-wave ultrasound reflectometry (CWUR) is presented here as a novel nondestructive modality for imaging and measuring surface roughness in a noncontact mode. In CWUR, voltage variations due to phase shifts in the reflected ultrasound waves are recorded and processed to form an image of surface roughness. An acrylic test block with surface irregularities ranging from 4.22 to 19.05 μm as measured by a coordinate measuring machine (CMM) is scanned by an ultrasound transducer having a diameter of 45 mm, a focal distance of 70 mm, and a central frequency of 3 MHz. It is shown that CWUR technique gives very good agreement with the results obtained through CMM inasmuch as the maximum average percent error is around 11.5%.
Finite element modeling of acoustic scattering from rough interfaces.
Authors: Yarbrough R. Abe , Isakson Marcia J.
Published: October 2013

The finite element method is applied to the problem of acoustic scattering from rough surfaces. This method has the advantage that the surface can have almost any form whereas analytic approximation methods generally require constrained surface roughness parameters. Scattering from a rough pressure release boundary and from a rough interface between two fluid media is considered in two dimensions. The problem domain is truncated with perfectly matched layers. These greatly decrease the number of degrees of freedom in the finite element problem while minimizing unwanted reflections of outbound energy back into the physical domain. The Helmholtz–Kirchhoff integral is used to extend the solution to points outside the modeled domain. Using finite elements as well as an exact integral equation method, solutions are obtained for multiple realizations of the rough scattering surface. The scattering strengths of energy reflected back into the water and of energy penetrating the sea floor in the case of bottom scattering are calculated over a range of scattering angles. The two exact numerical methods show good agreement. Results are compared with those of analytic approximation methods. [Work sponsored by ONR, Ocean Acoustics.]
An initial look at a continuing shallow water vector sensor acoustic ambient intensity study.
Authors: Deveau David , Lyons Anthony
Published: October 2013

Deployed in June 2008 off the Coast of Andros Island, Bahamas, seven compact Wilcoxon TV-001 vector sensors have been placed in 15 m of water with the goal of characterizing the ambient noise acoustic intensity with respect to environmental conditions. The study's objective is to better understand how intensity fields describe the shallow water noise environment. The system is designed to gather raw ambient acoustic noise pressure and particle acceleration every hour for a period of one year. The environmental database includes weather and wave information for the local area that can be correlated to the acoustic data. This report will present insight into the overall array development and deployment as well as an initial correlation of intensity fields to the recognized environmental stimulus. Results will be from individual sensors and these same sensors combined into vertical and horizontal arrays utilizing classic linear beamform processing. In addition to gathering data on acoustic ambient noise, data from local passing rain storms, marine mammals, fixed sources, and moving platforms will also be collected and analyzed.
Excitation of low-frequency modes of solid cylinders by evanescent and ordinary propagating waves.
Authors: Espana Aubrey , Marston Phillip L.
Published: October 2013

When using sound to detect objects buried beneath the seafloor, often the acoustic source has a large horizontal stand-off distance. In such situations there is evidence that the incident acoustic wave in sand can have a significant evanescent component. In prior work, organ-pipe modes of water-filled shells were excited in a laboratory simulation, the most significant result being the “double spatial decay rate” effect [Marston et al., J. Acoust. Soc. Am. 122, 3034 (2007)]. To further understand the coupling by evanescent waves into low-frequency modes of cylinders, backscattering by small solid aluminum cylinders was studied with ordinary-wave illumination. Free-field experiments identified resonances worthy of investigation in an evanescent wave experiment. Several of the features identified have been reproduced in FEM-COMSOL calculations by Williams of APL-UW. For both ordinary and evanescent waves, strong modes were often excited when the cylinder was highly tilted. A further understanding of the modal features was gained by evaluating the temporal variance of the response. Furthermore, the aluminum cylinders also showed an enhanced spatial decay rate when compared to that of the evanescent soundfield. [Work supported by ONR.]
Higher moment estimation for reverberation simulation.
Authors: LePage Kevin D.
Published: October 2013

In recent work the second, third, and fourth spatial moments for exponentially distributed roughness with either Gaussian or von Karman spatial spectra have been presented by the author. In this talk results for the second, third, and fourth moments of reverberation pressure for perturbation theory scattering from surfaces with von Karman spectra are obtained using numerical quadrature. Results are compared to closed form expressions for the second and fourth moments obtained for Gaussian correlation functions. Methods to extend these results to the class of chi-square distributed roughness heights of arbitrary order, which encompasses both exponential and Gaussian height distributions, are also presented.
Time-domain solutions for Rayleigh and Stoneley waves using the single-scattering parabolic equation method.
Authors: Metzler Adam M. , Siegmann William L. , Collins Michael D. , Zingarelli Robert A. , Chin-Bing Stanley A.
Published: October 2013

The parabolic equation method implemented with the single-scattering correction accurately handles range-dependent environments in elastic layered media. Interfaces between elastic media may be treated efficiently by subdividing into a series of two or more single-scattering problems [Küsel et al., J. Acoust. Soc. Am. 121, 808–813 (2007)]. In addition to environmental waveguide parameters, the procedure uses several computational parameters. The impacts of the number of interfacial scattering problems, an iteration scheme convergence parameter, and the number of iterations for convergence are shown on the accuracy and efficiency of the method. In particular, selection criteria for these parameters are developed. Fourier transforms and syntheses generate time-domain solutions for seismic applications of interest. Examples for model waveguides show features of Rayleigh and Stoneley wave propagation, and comparisons with solutions from other methods are shown. [Work supported by the ONR.]
Three-dimensional spatial coherence measurements: Vertical, longitudinal/transverse horizontal coherence.
Authors: Wan Lin , Zhou Ji-Xun , Rogers Peter , Knobles David
Published: October 2013

In the Shallow Water ’06 experiment, two L-shape arrays were deployed. Two horizontal components of these arrays were laid on the sea bottom. One was exactly perpendicular to the direction of sound propagation. The other was exactly parallel to the direction of sound propagation. This configuration offered an opportunity to directly measure the vertical, longitudinal horizontal, and transverse horizontal coherence. The results of spatial coherence were averaged over different pairs of hydrophones and over a frequency bandwidth of 100 Hz. The vertical coherence showed receiver depth dependence. When the source and the receivers were located below the thermocline, both the vertical and longitudinal horizontal coherence lengths (in units of wavelength) increased with increasing range and frequency. The longitudinal horizontal coherence length was much larger than the vertical coherence length. These results were similar to the predictions by Smith’s model [J. Acoust. Soc. Am. 60, 305–310 (1976)] with a frequency dependent bottom reflection loss [Zhou, Chin. Phys. 1, 494–504 (1981)]. The transverse horizontal coherence length/wavelength decreased with frequency. When the source depth was within the thermocline, the transverse horizontal coherence lengths exhibited weak range dependence in the 100–300 Hz range and was larger than 40 wavelengths. [Work supported by ONR.]
A green pathway to classroom acoustics: A comparison of classroom acoustic standards.
Authors: Bruck Daniel , Kurtz Alexis
Published: October 2013

Towne seems to have started the modern classroom acoustics movement, and Nixon was one of its earliest volunteers. Nixon contributed to an important success of that movement, the development of ANSI standard for classroom acoustics, S12.60-2002. Some of Nixon’s collaborations with Towne are remembered. Nixon’s efforts are also honored by showing important ways in which the standard influences contemporary practice of school building. The U.S. Green Building Council has adopted elements of the ANSI standard in its evolving LEED for Schools. Why not all? The acoustical requirements of LEED for Schools are compared with S12.60-2002. Ongoing challenges of integrating good acoustics into school buildings in the context of sustainable design are discussed. The authors discuss the issues encountered in development of the LEED standard when acoustics vies with competing needs. The importance of involvement by members of the classroom acoustics community to further Nixon’s work is also discussed.
Issues in the sound absorption treatment in classrooms.
Authors: Sutherland Louis , Lubman David
Published: October 2013

Classrooms require adequate sound absorption to establish appropriate environments for listening and learning. Sound absorbing materials improve the classroom learning environment by reducing reverberation time and by reducing background noise levels. ANSI standard S12.60-2002 provides simple methods for determining the amount of sound absorption needed to limit reverberation time. One of those methods is briefly reviewed. Throughout his career, Nixon’s forte was the sound absorption treatment of rooms. He and his colleague, the late Robin (Buzz) Towne, were pioneers—sounding the alarm that, with hardly anyones notice, poor classroom acoustics had become a widespread problem adversely impacting scholastic achievement. The two were also instrumental in motivating the Acoustical Society of America to lead the change toward a solution. The Society is indebted to Nixon both for his pioneering effort to awaken acousticians and others to the need for good classroom acoustics and for his diligent participation in the subsequent effort to develop an ANSI standard for that purpose.
A survey of unoccupied and occupied acoustic conditions in existing modular classrooms.
Authors: Philipp Norman H. , Wang Lily M.
Published: October 2013

Modular classrooms are widely in use across the United States, and an addendum on their acoustic performance is being prepared to supplement the classroom acoustics guidelines given in ANSI S12.60-2002. This paper presents measurements made in a large number of existing modular classroom units in Omaha, NE, both in unoccupied and occupied conditions. In the unoccupied classrooms, the gathered data include (a) reverberation times; (b) background noise levels in heating, cooling, and ventilation modes; and (c) airborne sound attenuation across all four exterior walls. In the occupied classrooms, sound levels were logged in both the room interior and exterior throughout an entire school day. Correlations have been made between the interior and exterior data to determine how often the interior noise levels could be attributed to exterior sources. Summaries of the results will be provided in an effort to benchmark the current conditions of modular classroom constructions.
Comparing classroom acoustics in green and nongreen schools.
Authors: Harght Pamela , Coffeen Robert C.
Published: October 2013

Minimal data are available on existing conditions regarding the acoustical environment in public schools throughout the U.S. Furthermore, rarely is there communication between those involved in the design/build process and the professionals who utilize these spaces on a daily basis. The research presented for this paper was conducted in two parts. First, an electronic survey was issued to teachers of public elementary, middle, and high schools in all 50 states in green and nongreen classrooms. This survey gathered results on questions on the present condition of the teachers’ classrooms in terms of background noise levels, noise from adjacent spaces and the outdoors, speech intelligibility, and the level of importance of acoustics for a teacher with regard to other indoor environmental concerns, i.e., thermal comfort, daylighting, and layout of classroom. The second part of this research involved testing of classrooms, both green and nongreen. The testing complied with ANSI S12.60-2002 for testing procedures for reverberation time, background noise levels, transmission loss, and speech interference level. Finally, the results from both parts were compared for nongreen and green classrooms regarding acoustics and classroom design for potential future applications in the school and architecture communities.
Analysis of commonly witnessed vehicle accident sounds in situ.
Authors: Neale William , Terpstra Toby
Published: October 2013

Research by Harber and Harber demonstrates the inaccuracy human beings have when recounting their experience witnessing a vehicle accident. However while witness statements can be unreliable, what a witness might have seen or heard is nonetheless important and cannot simply be disregarded. A witnesss recollection of audible sounds such as tire screeching, vehicle acceleration, or impacts between objects can profoundly affect how an accident sequence is interpreted by experts in vehicular accident reconstruction who may rely on witness statements when there is a dearth of physical evidence that properly defines the sequence of events. This paper provides a means for assessing the validity of what witnesses hear by recreating, recording, and measuring commonly heard accident sounds in real world conditions. Sounds, such as engine noise, tire screeching, and impacts, are recreated using various vehicles and in various environments. These sounds are recorded, observed, and analyzed to provide an understanding of how these sounds might be experienced by a witness. Among some of the testing variables are vehicle speed, roadway temperature, vehicle type, roadway surface, and background environment. These factors are evaluated in how they affect sound quality and clarity, frequency spectrum, sound level, and the sound's reverberation and directivity.
Statistical learning approach applied to road surface classification.
Authors: Paulo Joel , Bento Coelho José
Published: October 2013

Measures aiming environmental noise abatement usually consider acoustic barriers alongside the road. However, the cost associated with these measures is usually considerably high and its performance in urban areas is reduced. The problem of the visual impact is another issue affecting the communities. Nowadays, road planners have started to consider silent surfaces as an alternative. These types of surfaces are constituted basically by changing the texture and/or porosity of the mixtures. In some conditions, noise level abatements up to 15 dB can be achieved. Therefore, a considerable variety of different surfaces are available. The main goal of this research is to identify and classify different types of road pavements by analyzing the noise profile, using the close proximity method. Feature extraction and selection is one of the first procedures on a classifier algorithm. Moreover, the accuracy of the results is strongly dependent on the right choice of the selected feature vector. Standard classifiers are being tested in order to establish guidelines for future developments of this research. In situations of net road surveillance, searching for inhomogeneities on the surface and the presentation of the results in a geographic map, showing the locations of the surface types and the noise levels, can improve the accuracy of the noise mapping models.
Comparison of several acoustic models with measured data for a pure tone sound source.
Authors: Ikelheimer Bruce , Downing Micah , James Michael
Published: October 2013

A unique data set of outdoor acoustical measurements has been collected in a study area of over 800 sq km. Over 100 pure tone, high-level sources are uniformly distributed within this area. These sources were sounded simultaneously for 4 min on two consecutive days and were recorded by 24 sound level meters also distributed within the study area. In addition to the sound data, surface meteorological data were collected at the sound level meters site along with local area weather service data. The sound level meters collected third-octave band 1-s Leqs, while the weather data collected included wind speed, direction, temperature, humidity, and atmospheric pressure every 5 s. The terrain in the study area is hilly with varying degrees of vegetation and some large bodies of water. This data set provides a unique opportunity to compare various outdoor sound propagation models with measured data for propagation distances from 375 m up to several km. This presentation will compare the measured results with model calculations including simple and complex ray-tracing methods and a parabolic equation method.
Frequency response, acoustic impedance and background noise of microbarometers.
Authors: Ponceau Damien , Alcoverro Benoit , Olivier Serge
Published: October 2013

Frequency response and background noise of microbarometers are derived from their principle of operation. Lumped element models are proposed to describe each part of these absolute infrasound sensors and are used to derive analytical expressions for frequency response, acoustic impedance, and background noise of microbarometers. A set of experimental methods to estimate all elements is discussed in this paper. Some of them need a specific instrumentation developed in our laboratories. Others are very simple as they require only common equipment and can be applied on field to calibrate these sensors. All these methods have been applied to a set of MARTEC MB2005 microbarometers. Results from the theory and measurements are compared in this paper.
Subsurface windscreen for the measurement of outdoor infrasound.
Authors: Shams Qamar A. , Burkett Cecil G. , Comeaux Toby , Zuckerwar Allan J. , Weistroffer George R.
Published: October 2013

A windscreen has been developed that features two advantages favorable for the measurement of outdoor infrasound. First, the subsurface location, with the top of the windscreen flush with the ground surface, minimizes the mean velocity of the impinging wind. Secondly, the windscreen material (closed cell polyurethane foam) has a sufficiently low acoustic impedance (222 times that of air) and wall thickness (0.0127 m) to provide a transmission coefficient of nearly unity over the infrasonic frequency range (0–20 Hz). The windscreen, a tightly sealed box having internal dimensions of 0.3048×0.3048×0.3556 m3, contains the microphone, preamplifier, and a cable feed thru to an external power supply. Provision is made for rain drainage and seismic isolation. A three-element array, configured as an equilateral triangle with 30.48 m spacing and operating continuously in the field, periodically receives highly coherent signals attributed to emissions from atmospheric turbulence. The time delays between infrasonic signals received at the microphones permit determination of the bearing and elevation of the source, which correlate well with locations of pilot reports within a 320 km radius about the array. The test results are interpreted to yield spectral information on infrasonic emissions from clear air turbulence.
High-order parallel discontinuous Galerkin method for time-domain acoustic simulations.
Authors: Lähivaara Timo , Huttunen Tomi , Simonaho Simo-Pekka
Published: October 2013

The modeling of acoustic waves in the time-domain poses a significant challenge in scientific computing. A promising candidate for solving the three dimensional wave equation is the discontinuous Galerkin (DG) method. Advantages of the DG method are the easy parallelization and a special matrix structure which can reduce the overall time and the computer memory needed for solving the problem. In this study, a high order parallel DG method is investigated. The DG solver is implemented using the C++ programming language. Communication between processors of the parallel computer is performed using the message passing interface. In the solver, the polynomial degree of the basis functions is chosen individually for each element of the computation mesh (up to ninth order polynomials can be used). The unbounded problem is truncated using the perfectly matched layers. The method is evaluated with numerical simulations that are performed on a personal computer cluster.
Foreign-accented speech in noise.
Authors: Dalby Jonathan
Published: October 2013

Studies of the perception of foreign-accented English speech have shown that native-speaking listeners are able to adapt quite quickly and effectively to the phonetic and phonological characteristics of non-native speech [Bradlow and Bent, 2003, XVth ICPS Proceedings] and can do so even when the accent is unfamiliar [Clark and Garrett, J. Acoust. Soc. Am. 116 (2004)]. Such studies raise the question of how much cognitive effort underlies this adaptation as well as questions about the robustness of the adaptation under less-than-optimal listening conditions. Rogers et al. (2004) have shown that the intelligibility of connected speech, even that from highly proficient non-native speakers, was degraded more than that of native speech when presented in noise. This study attempts to replicate that noise effect for isolated words. Two Spanish-speaking adults with differing oral English proficiency and two American English speakers recorded 50 CVC words from each of three lists that have been shown to be of equivalent perceptual difficulty (NU lists 1, 2, and 4). Words were presented with no noise and at −5 and −10 dB SNR to native listeners. Early results suggest that word identification scores are similar to those found for sentence comprehension. CDT, Inc. markets a speech intelligibility training software. [Work supported by Purdue Research Foundation.]
Tracking non-native acquisition of the Spanish tap-trill distinction: Cross-modal priming differences between native and non-native Spanish speakers.
Authors: Herd Wendy , Sereno Joan
Published: October 2013

English-speaking learners of Spanish often fail to achieve nativelike pronunciation of the tap-trill distinction in words like caro “expensive” and carro “car.” The trill proves difficult because it is neither a phoneme nor an allophone in English. Although the tap exists as an allophone of /t/ and /d/ in American English, learners of Spanish must learn to process it as a phoneme rather than an allophone. Similarly, English learners of Spanish have difficulty acquiring the spirantization of voiced stops, i.e., /d/ spirantizes in codo “elbow,” which occurs in the same environment as flapping. This study uses a cross-modal priming task to investigate whether L2 Spanish learners are able to process intervocalic tap, trill, /d/, and /t/ in the same way as L1 Spanish speakers. Using a cross-modal priming paradigm, eight English-speaking learners of Spanish were compared to eight native Spanish speakers. Results show that auditorily presented words with intervocalic taps resulted in faster response times for words like cada for learners of Spanish, but the same auditory stimuli resulted in faster response times for identically matching targets like cara for native Spanish speakers. These results suggest that cross-modal priming can be used to track L2 acquisition.
A computer graphic three-dimensional tongue and lip movement synchronized with English fricatives for Japanese learners.
Authors: Isei-Jaakkola Toshiko , Suzuki Shigeki , Morishima Shigeo , Hirose Keikichi
Published: October 2013

Some English fricatives are difficult specifically for Japanese learners of English (JL2) to produce. Simultaneous articulation of the lip and teeth (as in lebiodentals), or the tongue and teeth (as in dentals), or protruding lips (as in postalveolars) do not exist in standard Japanese. To understand partially or completely, invisible articulatory movements are unavoidable for JL2 in order to produce these fricatives properly. Thus, as an aid for the basic pronunciation training, a visualized automatic lip and tongue movement program synchronized with these fricatives was developed, utilizing three-dimensional computer graphic technologies. In this program, not only the lips, teeth, and tongue but also the other necessary speech organs were made to be half transparent. Consequently, it enables the learner to listen to and repeatedly model the target segmental sound while looking at these articulatory movements vividly from all kinds of angles. This system changes the phonetic practice situation from two-dimensional paper-based learning/teaching methods with audio-visual tools into revolutionary user-friendly method in phonetic class and outside the classroom. In addition, this system is applicable to any language sounds in the near future. [Work supported by JSPS and Chubu University Grant (A).]
Production and perception of English vowel categories by native Korean speakers.
Authors: Son Ga Yeon
Published: October 2013

This study deals with the production and perception of a second language (L2) by non-native speakers assessing phonetic convergence in their production and perceptual divergence. Production and perception of nine American English monophthongs and eight Korean monophthongs by the experienced and inexperienced Korean speakers were acoustically analyzed. The experiments consist of the production test, identification tests, and discrimination test. The experienced group showed relatively successful production of English vowels that have dissimilar acoustic properties with Korean vowels, while the inexperienced group showed complete phonetic interlingual identification. However, in perception, both groups successfully scored in the identification and discrimination tests, not showing perceptual identification between English and Korean. It suggests that two language systems mutually influence in one native phonetic space, and linguistic experience with L2 brings the reorganization of the phonological system, adding new phonemic categories for dissimilar L2 phones. The biased results in production and perception tests demonstrate that production of L2 is strongly related to the native phonemic categories, but perception is not. Therefore, the findings suggest that two different phonological systems are incorporated into the same phonetic space and linguistic experience with L2 boosts this process.