bluuebunny/crossref_metadata_2025 huggingface dataset:
Study of the effects of different endpin materials on cello sound characteristics
Authors: Fleming Clinton R. , Stypowany Cassey R. , Celmer Robert D. , Vigeant Michelle C.
Published: October 2011

The purpose of this study was to investigate the effects of changing a cello's endpin material and boundary conditions on the sound and vibration characteristics of a cello. It was hypothesized that an endpin made of a denser material than stainless steel, which is traditionally used, would improve the tone quality of the cello. In terms of endpin boundary conditions, it was hypothesized that using a shorter endpin with fixed end conditions might also improve the vibration characteristics and sound radiation efficiency of the cello. Objective and subjective tests were conducted to examine the effects of the different endpin materials. Sound power level output and vibration measurements of a cellist playing on different endpins were obtained following ISO 3741. In general, sound power levels and measured vibrations were consistent for all endpins for all notes tested. For the subjective study, volunteer cellists played selected excerpts with the different endpins, not knowing which endpin they were using. This testing showed that the tungsten endpin gives the cello a tone quality with similar warmth to the stock endpin but makes the cello less responsive. The results of both the objective and subjective tests for all endpin materials will be presented.
Using acoustics to study the effectiveness of fish habitat restoration: A comparison of methods
Authors: Bollinger Michael , Miksis-Olds Jennifer L. , Madden Laura E.
Published: October 2011

Many Pennsylvania reservoirs are created by clearing valleys prior to damming streams. These impoundments are typically devoid of appropriate benthic habitat for fish. The Pennsylvania Fish and Boat Commission has been placing artificial habitat structures in reservoirs across Pennsylvania for 30 years in an effort to increase game-fish production. However, the effectiveness of these efforts is unknown. Active acoustic technology is being considered as an alternative to conventional survey methods for assessing the effectiveness of artificial structure in F.J. Sayers Reservoir in Howard, PA. An autonomous echosounder [(ASL Environmental Sciences, 460 kHz Acoustic Water Column Profiler (AWCP)] was deployed horizontally near artificial rock structures to monitor associated fish activity. The AWCP recorded acoustic backscatter at 1-s intervals continuously for two days. Data was analyzed by (1) manually counting fish tracks on an echogram for each hour and range and (2) calculating integrated volume backscatter values per hour for each range. Both analysis methods showed similar trends in fish activity during the study period. Results agreed with expected behavioral patterns for the species in this body of water; fish activity increased during the night and decreased during the day, with the exception of a mid-day increase on one day.
Extracting heterogeneous structured features from music performance
Authors: Ren Gang , Headlam Dave , Roessner Stephen , Lundberg Justin , Bocko Mark F.
Published: October 2011

In music performance, the musician adds artistic expressive elements beyond the information contained in conventional western music scores. As micro-fluctuations these expressive dimensions convey emotions and essential interpretative information and can be measured and compared quantitatively, over large and small scales, and evaluated for their effect on aspects of the performance. We present a heterogeneous expressive music feature description that include both inter-note features that extend over musical phrases composed of several notes, and intra-note features that represent the internal variations within each musical note. The intra-note features include pitch and pitch deviation, dynamic level, timber, articulation and vibrato. The inter-note features include timing and dynamics, as well as timber, pitch deviation, articulation, and vibrato extending across multiple notes and musical phrases. A complete multi-dimensional feature description for every note is unnecessary because there is a hierarchy of note functions in a musical phrase and certain features may be more or less important depending upon their function in the phrase. Thus, we introduce a heterogeneous representation that affords the flexibility to reflect the hierarchical significance of each note or phase segment. Specific visualization and animation schemes of the proposed representation, to facilitate user interaction with the music, are also presented.
Playing a virtual drum from a real one
Authors: Puckette Miller
Published: October 2011

An electronic musical percussion instrument is described which can be played by tapping or otherwise playing on a small physical object, whose vibrations are picked up and used to activate the virtual one. To do this, a ceramic tile or other small rigid object is attached to a contact microphone. Any resonances of the physical system are filtered out using linear predictive analysis, so that the resulting residual signal approximates the excitation the player introduces into the system. The resulting audio signal could then be forward filtered to recover the tile's own sound, but instead it is fed into a nonlinear reverberator to simulate a variety of real or fanciful percussion instruments. The result is a highly expressive and playable electronic percussion instrument, that is, both easy and inexpensive to build. The effects of various design parameters of the nonlinear reverberator on the resulting sound are discussed.
Music from “an evil subterranean beast.”
Authors: Chafe Chris
Published: October 2011

Two recent compositions have capitalized on the great expressive range of a custom-designed music synthesis algorithm. The “Animal” plays a part in Tomato Quintet (a music installation) and Phasor (a contrabass and computer piece). The algorithm is played with live controls, respectively, updates from CO2 sensors and signals from a sensor bow. Nuanced deflections of parameter values elicit characteristic and sometimes quirky behavior. The bounded set of sonic behaviors goes into making up an identifiable personality whose moods or temperaments can be traversed in the music which exploits it. “The computerized sounds were spacey and sometimes menacing, sounding at times like Chafe was trying to tame an evil subterranean beast.” (H. Ying, Global Times, 2011). Animal's algorithm is comprised of the logistic map with two parallel waveguides in its feedback path. Animal can be categorized as a “meta-physical” model or modeling abstraction as seen in earlier integrations of families of physical models (P. Cook, ICMC, 1992) and recombinations of physical model components in physically impossible ways (C. Burns, Composition, 2003). Abstraction opens the door to inclusion of mathematical “parts” from other domains. In Animal's case, the logistic map has been borrowed from population biology.
An active learning-based interface for synthesizer programming
Authors: Cartwright Mark B. , Pardo Bryan
Published: October 2011

Commercial software synthesizer programming interfaces are usually complex and tedious. They discourage novice users from exploring timbers outside the confines of “factory presets,” and they take expert users out of their flow state. We present a new interface for programming synthesizers that enables both novices and experts to quickly find their target sound in the large, generative timber spaces of synthesizers. The interface does not utilize knobs and sliders that directly control synthesis parameters. It instead utilizes a query-by-example based approach combined with relevance feedback and active learning to create an interactive, personalized search that allows for exploration. We analyze the effectiveness of this new interface through a user study with four populations of varying types of experience in which we compare this approach to a traditional synthesizer interface. [Work supported by the National Science Foundation Graduate Research Fellowship.]
Hearing protector labeling—Yes or no?
Authors: Feith Ken
Published: October 2011

The federal government has been accused of requiring needless product labeling in order to protect citizens from themselves. While there is probably some truth in these accusations, the key point of labeling is lost in the noise of criticism—at least in the case of hearing protector devices. If we were to consider the one word that most influences our life actions we would find it to be the word “decisions.” A moment's thought would reveal that from the minute we wake in the morning until we achieve a deep state of sleep, we are confronted with the need to make decisions. Every move, every action, is determined by either conscious or subconscious decision making. To that end, we need to consider those elements that influence our decision making process—understandable and accurate information. Thus, the role of federal labeling of select products is to provide an understandable and accurate means for making decisions that may have an impact on our ability to function effectively in our work, maintain our good health, avoid personal injury, and the list goes on. This presentation reveals the secrets behind regulatory labeling.
1.Comparison of the HPDLab and REATMaster software/hardware systems for ANSI S12.6 testing
Authors: Byrne David C. , Perry Caryn C. , Murphy William J.
Published: October 2011

The American National Standard Methods for Measuring the Real-Ear Attenuation of Hearing Protectors (ANSI S12.6-2008) requires a Békésy procedure for testing occluded and unoccluded thresholds. Since 2002, the National Institute for Occupational Safety and Health (NIOSH) has used the custom-designed HPDLab software operating Tucker-Davis Technologies System 3 hardware. ViAcoustics, Nelson Acoustics, NASA, and NIOSH researchers recently developed REATMaster which runs on National Instruments hardware in the LabVIEW environment. Ten subjects were trained by the experimenter on how to fit a passive earmuff and were qualified according to the requirements of ANSI S12.6-2008. The laboratory was configured such that diffuse sound field thresholds were tested with either the HPDLab or REATmaster hardware by flipping a toggle switch. The earmuff was not touched or re-positioned between test trials with the two different hardware/software systems. The test sequence for the order of open and occluded measurements was counterbalanced across occluded conditions and hardware system. Results from this testing were used to validate the REATMaster system for its ability to produce accurate threshold data. Preliminary results indicate no significant differences between the two systems.
Measuring, rating, and comparing the real ear attenuation at threshold of four earplugs
Authors: Murphy William J. , Stephenson Mark R. , Byrne David C.
Published: October 2011

The effect of training instruction, whether presented as the manufacturer's printed instructions, a short video training session, specific to the product, or as a one-on-one training session, was evaluated using four hearing protection devices with eight groups of subjects. The Howard Leight Fusion and Airsoft premolded earplugs and the Moldex PuraFit and EAR Classic foam earplugs were tested. Naïve subjects were recruited and tested using three different forms of training: written, video, and individual training. The differences between group averages for A-weighted attenuation were not statistically significant when compared between the video or the written instruction conditions, regardless of presentation order. The experimenter- trained A-weighted attenuations were significantly greater than the written and video instruction for most of the protectors and groups. For each earplug, the noise reduction statistic for A-weighting (NRSA) and the associated confidence intervals were calculated for the 90th and 10th percentiles of protection. Across subject groups for each protector, the differences between NRSA ratings were found to be not statistically significant. Several comparisons evaluating the order of testing, the type of testing, and statistical tests of the performance across the groups are presented. [Portions of this work were supported by the U.S. EPA Interagency Agreement DW75921973-01-0.]
Noise surveys and hearing protection at the Royal Canadian Mint plant
Authors: Yan (Jimmy) Jianping
Published: October 2011

Royal Canadian Mint is a Crown Corporation to produce circulation and non-circulation coins for Canada and other countries. Noise measurements have been conducted regularly for production environment and before any new equipment operation in Royal Canadian Mint, and to protect our employee from hearing loss is one of our health and safety issues applied in whole plant. The formula used in the noise measurement is introduced in this paper, and the noise calculations are also discussed for the continuous noise sources, intermittent noise sources, and the combinations of individual noise sources. Some noise survey results are listed and analyzed, and the related hearing protective methods in our plant are also described.
Analysis and extension of scattering from rigid infinite wedge
Authors: Bhatta Ambika , Pflanz Timothy , Thompson Charles , Chandra Kavitha
Published: October 2011

A new representation of the scattered pressure of a point source due to infinite rigid wedge is derived by simplifying the Bessel functions in the modal solution and by implementing the image based representation of the source. The solution comprises of six impulses contributing to the reflected and transmitted pressure and an integral for the diffracted pressure. The inverse Fourier transform of the solution yields the time domain pressure and the asymptotic solution for higher frequency is evaluated and compared with solution obtained by steepest descent method. The analytical approach based on wedge diffraction is extended to evaluate the scattered pressure from finite rigid polygon with three vertices and edges. The computational results for the scattered pressure will be explained in detail.
Acoustic scattering strength of turbulence generated waves in a shallow water flow
Authors: Horoshenkov V. Kirill , Nichols Andrew , Tait J. Simon , Maximov A. German
Published: October 2011

This work examines an airborne method of ultrasonic characterization of the dynamically rough water surface in an shallow water flow. A new experimental setup has been developed and tested in a 12 m long, 450 mm wide laboratory flume. This equipment includes a source of a 45 kHz airborne wave, microphone array, and an array of wave probes. It has been shown experimentally and theoretically that the directivity of the ultrasonic wave scattered by the dynamically rough water surface can be related directly to the roughness correlation function which can be measured non-acoustically. It is shown that the form of the roughness correlation function is determined unambiguously by the scale and intensity of the turbulence processes in the flow. The proposed acoustic method is accurate, remote, and rapid. It can be used to relate the spectral and statistical characteristics of the rough flow surface to the scale and intensity of the turbulent flow structures which cause the water surface to appear rough.
Application of the Beilis and Tappert parabolic equation to long-range sound propagation over irregular terrain
Authors: Parakkal Santosh , Gilbert Kenneth E. , Di Xiao
Published: October 2011

The Beilis and Tappert parabolic equation method is studied in the applications of sound propagation over irregular terrain. The exact ground impedance condition for porous ground is derived and applied to propagation over hills with slopes from 5 ° to 22 °. It is found that for slopes less than approximately 20 °; the flat-ground impedance condition is sufficiently accurate. For slopes greater than about 20 °, the limiting factor on numerical accuracy becomes the narrow-angle approximation used in the Beilis and Tappert method. The generalized polar coordinate parabolic equation method (Polar-PE method) was developed partly to provide a very solid numerical benchmark to the Beilis and Tappert PE method.
Hemilarynx experiments: Statistical analysis of vibrational parameters over changing glottal conditions
Authors: Doellinger M. , Berry D. A.
Published: October 2011

In vitro (humans, animals) or in vivo (animals) hemilarynx experiments are still the only way to visualize and analyze 3-D vibrations of the entire medial surface of the vocal fold. However, to date, only a few studies have been performed. For three different larynges during sustained phonation, vibrational output was statistically analyzed as a function of glottal airflow (800 to 1800 ml/s), adductory forces (10, 50, 100 g), and two different anterior–posterior pre-stress conditions (10, 50 g). Using a homogeneous grid of 30 microsutures sewn along the medial surface of the vocal fold, global parameters (empirical eigenfunctions/EEF, f0, subglottal pressure, and sound intensity level) were computed. Similarly, local parameters (displacements, velocities, and accelerations) were analyzed at the 30 different suture locations. The recordings were obtained using a digital high-speed camera (2000 and 4000 fps). Increased airflow resulted in significant statistical changes in all parameter values except the empirical eigenfunctions. For increased adduction and a lower pre-stress (10 g), the local parameters increased more than for the higher pre-stress condition (50 g). For the two pre-stress conditions and an increased adductional force, most global parameters exhibited significant changes (f0, sound intensity, EEF2), while other global parameters (subglottal pressure) did not.
Electroglottographic and acoustic measures of phonation across languages
Authors: Keating Patricia , Kuang Jian-Jing
Published: October 2011

Many less-commonly studied languages use phonation contrastively: vowels, and therefore words, can differ in phonation type. Most such languages contrast two or three phonation categories. In other languages, a lexical tone (contrastive pitch) may typically have a redundant non-modal phonation. We have collected electroglottographic and audio recordings from speakers of six languages with one or both of these uses of phonation (Bo, Gujarati, Hani, White Hmong, Mandarin, Yi) and analyzed them with EGGWORKS. EGGWORKS is a free program which automates several EGG measures and works together with VoiceSauce, a free program which automates many acoustic voice quality measures. Comparisons of the EGG and acoustic measures within and across languages, and their correlations, will be presented. We have also applied functional data analysis to EGG pulse shapes and will describe the relations of the factors underlying the shapes to the EGG and acoustic measures. [Work supported by NSF.]
The two-source problem: Producing voiced obstruents
Authors: Shadle Christine H.
Published: October 2011

The relationship between the physiological parameters and the acoustic output of the voice source is complex, and even more so when voiced obstruents are considered. Standard models assume that the subglottal pressure is divided across the glottis and the supraglottal constriction, which explains both the weaker noise source of voiced compared to voiceless fricatives and the tendency of voiced stops and fricatives to devoice. However, such models do not take into account small adjustments in vocal fold posture that may, for example, decrease the phonation threshold pressure, or in the constriction that may act to optimize noise production. Larger adjustments are made for voiced fricatives, such as lowering the larynx and advancing the tongue root, that may also enhance a speaker's ability to produce both voice and noise sources at a given subglottal pressure. In this talk results from a wide range of studies will be considered to arrive at a more nuanced understanding of the interaction of two source mechanisms in voiced obstruents.
Lexical-neighborhood and competing-task effects on word recognition and word recall by native and non-native listeners
Authors: Rogers Catherine L. , Frenton April M. , Hoefer Heather A.
Published: October 2011

In spoken-word recognition, high-frequency words with few and less frequently occurring minimal-pair neighbors (lexically easy words) are recognized more accurately than low-frequency words with many and more frequently occurring neighbors (lexically hard words). This easy-hard word effect has been found to be larger for non-native listeners with a relatively late age of immersion in an English-speaking environment. Previous research found no effect of a competing digit-recall task on spoken-word recognition and no effect of listener group or listening environment on digit-recall accuracy. The present study compares word recognition by native English-speaking listeners and non-native listeners with either earlier (age 10 or earlier) or later (age 14 or later) ages of immersion in an English-speaking environment. Spoken word lists composed of equal numbers of lexically easy and hard target words were presented in an open-set word-identification task. Spoken words were presented in quiet and in moderate background noise and preceded by a list of zero, three, or six visually presented words, which listeners were asked to recall following the spoken word-recognition task (competing task). The size of the easy-hard word effect, spoken-word recognition accuracy competing-task word-recall accuracy, and word-entry response time will be compared across listener groups and listening conditions.
Formant frequencies, vowel identity, and the perceived relative tallness of synthetic speakers
Authors: Barreda Santiago , Nearey Terrance M.
Published: October 2011

Listeners can make consistent judgments regarding the tallness of speakers [Rendall etal., J. Exp. Psych: Human Percep. Perform. 33, 1208 (2007)]. These judgments are informed by the f0 and formant frequencies (FFs) of a speaker's voice. However, FFs are also cues to vowel identity, such that a small speaker producing an /u/ might have lower average FFs than a larger speaker producing an /æ/. Do listeners use absolute FFs to judge speaker tallness, or do they “correct” for phonetic identity and consider the FFs of a vowel relative to those expected for that category? To test this, a series of synthetic vowels (/i æ u/) with different FF scalings and different numbers of formants (2–5) were created. These scalings were intended to replicate speakers of different vocal tract lengths (i.e., sizes). Participants were presented with pairs of vowels and asked to indicate which vowel sounded like it had been produced by a taller speaker. Results indicate that listeners consider both absolute and phonetically “corrected” FF information and that formants higher than F3 greatly reduce listener's reliance on absolute F1 and F2 information in making speaker size judgments.
Adapting to foreign-accented speech: The role of delay in testing
Authors: Witteman Marijt J. , Bardhan Neil P. , Weber Andrea , McQueen James M.
Published: October 2011

Understanding speech usually seems easy, but it can become noticeably harder when the speaker has a foreign accent. This is because foreign accents add considerable variation to speech. Research on foreign-accented speech shows that participants are able to adapt quickly to this type of variation. Less is known, however, about longer-term maintenance of adaptation. The current study focused on long-term adaptation by exposing native listeners to foreign-accented speech on Day 1, and testing them on comprehension of the accent one day later. Comprehension was thus not tested immediately, but only after a 24 h period. On Day 1, native Dutch listeners listened to the speech of a Hebrew learner of Dutch while performing a phoneme monitoring task that did not depend on the talker's accent. In particular, shortening of the long vowel /i/ into /I/ (e.g., lief [li:f], 'sweet', pronounced as [lIf]) was examined. These mispronunciations did not create lexical ambiguities in Dutch. On Day 2, listeners participated in a cross-modal priming task to test their comprehension of the accent. The results will be contrasted with results from an experiment without delayed testing and related to accounts of how listeners maintain adaptation to foreign-accented speech.
Phonetic imitation by school-age children
Authors: Nielsen Kuniko
Published: October 2011

Phonological representations such as phoneme and feature are often assumed in prevailing linguistic theories. However, little is known about how these representations are formed during the course of language acquisition. The current study aims to investigate developmental changes of phonological representations by examining how children imitate fine phonetic details of recently heard speech. Previous studies have shown that adult speakers implicitly imitate the phonetic properties of recently heard speech (e.g., Goldinger, 1998; Pardo, 2009). Recently, Nielsen (2011) showed that phonetic imitation was generalized at phoneme and feature levels, providing support for sub-lexical representations. The current study extends these findings and examines whether school-age children manifest similar patterns of phonetic imitation. The experiment employed the imitation paradigm in which participants' speech is compared before and after they are exposed to model speech with extended VOT on the target phoneme /p/. A preliminary analysis of data revealed that participants produced longer VOTs after being exposed to model speech with extended VOTs. The change was generalized to new instances of the target phoneme /p/ and the new phoneme /k/, indicating that sub-lexical units were also involved in phonetic imitation by children.
Recognition memory for speech of varying intelligibility
Authors: Van Engen Kristin J. , Chandrasekaran Bharat , Ayres Lauren , Czimskey Natalie , Smiljanic Rajka
Published: October 2011

Previous research has demonstrated that the internal representation of spoken words includes both a phonetic description and information about the source characteristics of specific talkers. Spoken utterances also vary greatly in intelligibility. However, relatively little is known about how such variability affects the encoding of speech signals in memory. In the current study, we hypothesize that increased intelligibility leads to better recognition memory for speech. We test this hypothesis by comparing recognition memory for clear and conversational speech produced by native and non-native speakers of English. In experiment 1, listeners were exposed to semantically anomalous sentences produced by a native speaker of English in clear and conversational speech styles. In experiment 2, listeners heard sentences in both styles produced by a Croatian-accented speaker of English. In each experiment, after initial exposure, recognition memory was tested with an old-new identification task. Preliminary results suggest that native-accented speech and clear speech may facilitate recognition memory compared to foreign-accented and conversational speech. The neural mechanisms involved in processing speech of varying intelligibility are currently being investigated using fMRI. The results of this project will expand our understanding of how different sources of variability in the speech signal affect speech processing and memory representations.
Non-symmetric time resolution for spectral feature trajectories
Authors: Zahorian Stephen A , Wu Jiang , Karnjanadecha Montri
Published: October 2011

In a study presented at the fall 2010 meeting of the Acoustical Society of America (Zahorian etal., “Time/frequency resolution of acoustic features for automatic speech recognition”), we demonstrated that spectral/temporal evolution features which emphasize temporal aspects of acoustic features, with relatively low spectral resolution, are effective for phonetic recognition in continuous speech. These features are computed using discrete cosine transform coefficients for spectral information from 8 ms frames and discrete cosine series coefficients (DCSCs) for their temporal evolution, over overlapping intervals (blocks) longer than 200 ms. These features are presented as an alternative to mel-frequency cepstral coefficients, and their delta terms, for automatic speech recognition (ASR). In the present work, it is shown that these features are even more effective for ASR, using a non-symmetric time window which is tilted toward the beginning of each block when computing DCSCs. This non-symmetry can be implemented by combining two Gaussian windows with different standard deviations. This work also supports the hypothesis that the left context is somewhat more informative to phonetic identity than is the right context. Experimental results for automatic phone recognition are given for various conditions using the TIMIT and NTIMIT databases.
Compensation for altered feedback is sensitive to speaking style
Authors: Katseff Shira , Houde John F.
Published: October 2011

Many speakers oppose alterations to auditory feedback, using a higher pitch when they hear their pitch lowered or a lower first formant when they hear that formant raised. But there is substantial variation in compensation within and across individuals. This case study asks whether a speaker's individual vowel space influences compensation for altered auditory feedback. A generalized linear regression model was used to investigate the influence of vowel density on trial-to-trial formant movement. First, two density maps, showing “hot spot” regions where the subject produced many vowels and “cold spot” regions where the subject produced few vowels, were constructed for (1) casual speech (from a 30-min mock interview) and (2) citation form speech (from 360 HVD words). Maps were produced for one non-phonetician speaker of California English. Afterward, the speaker participated in Five altered feedback experiments on Five separate days. The analysis showed that the subject's between-trial formant changes were not correlated with spontaneous speech vowel density but were positively correlated with citation form vowel density. This pair of results suggests that speech motor decisions are influenced by speech style, accessing citation form vowels more readily when producing citation form speech.
Temporal and spectral characteristics of fricatives in dysarthria
Authors: Kim Heejin , Hasegawa-Johnson Mark , Perlman Adrienne
Published: October 2011

Imprecise production of fricatives has been noted in various types of dysarthria, but their acoustic properties have not been fully investigated. Using durational measures and spectral moment analysis, this study examines the acoustic characteristics of alveolar and post-alveolar voiceless fricatives produced by speakers with cerebral palsy (CP) in three different intelligibility levels. The following questions are addressed: (1) Are fricatives longer in CP-associated dysarthric speech, compared to controls? (2) What is the evidence of reduced acoustic distinctions between alveolar vs. post-alveolar fricatives? and (3) Are the intelligibility levels associated with acoustic measures of fricatives? Duration and the first three spectral moments were obtained from word initial fricatives produced by 18 American English native speakers (9 speakers diagnosed with spastic CP and 9 controls). Results showed that speakers with CP exhibited significantly longer duration of fricatives and a reduced distinction between alveolar versus post-alveolar fricatives compared to control speakers. A reduced place distinction in dysarthric speech was mostly due to lower first moments and higher third moments compared to normal speech. The group difference was greater for alveolar fricatives than for post-alveolar fricatives. Furthermore, as the intelligibility level decreased, durational increase and the degree of place overlap were consistently greater.
The effect of listener age on the recognition of dysarthric speech
Authors: McAuliffe Megan J. , Gibson Elizabeth M. R. , Kerr Sarah E. , Anderson Tim
Published: October 2011

A growing number of studies have attempted to address the cognitive–linguistic source of intelligibility deficits in dysarthria. These have focused predominantly on the ability of young adults with normal hearing to decipher dysarthric speech. However, dysarthria is commonly associated with aging; older listeners often form primary communication partners. It is also known that older adults exhibit difficulty understanding speech that has been temporally or spectrally degraded. It follows that the spectral and temporal degradations present in dysarthric speech may pose a greater perceptual challenge for older, as opposed to younger, listeners. Twenty younger listeners (mean age = 20 yr) and 15 older listeners with good hearing for their age (mean age = 65 yr) transcribed the speech of individuals with moderate hypokinetic dysarthria. Percent words correct (intelligibility) was calculated and underlying error patterns at the suprasegmental and segmental levels of processing were examined. While the younger and older listener groups achieved similar intelligibility scores, the younger group showed greater reliance on syllabic strength cues to inform word boundary decisions. Similar levels of attention to segmental cues were observed across both groups. These results suggest that the recognition of dysarthric speech may be comparable across younger and older listeners.
Reliability and validity of a simple user interface (EarPrint™) for the personalization of mobile audio
Authors: Ramani Meena , Michael Nick , Pavlovic Chaslav
Published: October 2011

People hear differently and have different aesthetic criteria. Due to a number of these reasons, individuals may prefer different processing parameters for both speech and music. For example, some individuals may prefer more low frequency emphasis and longer reverberation time while listening to music. Alternatively, certain individuals with hearing loss may prefer higher compression ratios at high frequencies. To perform this “personalization,” a number of psychometric procedures can be used. They vary in terms of accuracy, complexity, testing time, etc. For consumer devices, such as headsets, personal amplifiers, personal music players, etc., we have developed a new methodology, called EarPrint™, which is fast, intuitive, and requires minimum instructions. The procedure is based on reducing a multidimensional parametric space into a 2-D surface representation where the user perceives a consistent quality increment as he/she moves his/her finger along a straight line. In this paper, we evaluate how valid and reliable the EarPrint™ results are when compared to more rigorous psychometric procedures (e.g., category scaling or magnitude estimates.) Fifteen subjects were tested and the results were analyzed in terms of validity, dependence on listening material and reliability.
Geoacoustic inversion with sequential Bayesian filtering and multipath arrivals
Authors: Michalopoulou Zoi-Heleni
Published: October 2011

When sound propagates in the ocean, multipath arrival receptions at spatially separated hydrophones provide a wealth of information on geometry parameters, including source location and water column depth, and sound speed in medium layers. Additionally, attenuation can be extracted using arrival amplitudes. We extend a method previously developed for arrival time estimation in space using particle filters, which now employs backward filters that smooth arrival time and amplitude estimates. The new approach is applied to synthetic and real data for source localization and geoacoustic inversion. Results are compared with estimates obtained using traditional arrival time estimation methods. The comparison reveals that the new technique provides more accurate results with reduced uncertainty in comparison to other approaches; uncertainty is expressed via posterior probability density functions. It is shown that the smoother plays a significant role in the quality of the time estimates and, subsequently, in geoacoustic inversion. [Work supported by ONR.]
Geoacoustic inversion from ambient noise data using a trans-dimensional Bayesian approach
Authors: Quijano Jorge E. , Dosso Stan E. , Dettmer Jan , Siderius Martin , Zurk Lisa M.
Published: October 2011

Geoacoustic inversion of seabed parameters from ambient noise in shallow water is a promising technique, with potential advantages over active survey methods such as low environmental impact, easier deployment procedures, and less restrictive hardware requirements. Bayesian inversion provides a framework to estimate the posterior probability density (PPD) of geoacoustic parameters. Parameter and uncertainty estimates can be obtained from PPD moments, such as the maximum a posteriori model, means, correlations, and marginal distributions. A fundamental step in the inversion is the selection of a model parametrization (i.e., number of seabed layers) consistent with the data information content. Recent developments in Bayesian inversion of seismic and active-source acoustic data have considered a trans-dimensional approach to model selection, where the number of model parameters is treated as unknown. Different models are sampled according to their support by the data, accounting for parametrization uncertainty in the geoacoustic parameter uncertainty estimates. This work applies a trans-dimensional reversible-jump Markov chain Monte Carlo algorithm to ambient noise reflection coefficient data. The approach is demonstrated with data collected on the Malta Plateau using a vertical line array. [Work supported by ONR.]
Sequential geoacoustic inversion for mobile and compact source–receiver configuration
Authors: Carriére Olivier , Hermand Jean-Pierre
Published: October 2011

The development of light instrumentation for geoacoustic characterization requires higher-frequency transmissions and inversion methods suitable for mobile configurations that efficiently combine any a priori knowledge about the environment or the time-varying source–receiver geometry and the acoustic transmission measurements. Sequential filtering methods provide a framework to achieve these objectives. In this paper, repeated short-range acoustic transmissions (between 750 and 1500 m) acquired on a drifting 4-hydrophone array that spans a small part of the water column, are sequentially inverted in nonlinear Kalman filters. The sequential filtering approach is demonstrated on actual data from the MREA/BP07 sea trial, with a space-coherent processing of multitone signals and a phase-coherent processing of linearly frequency modulated signals, in low (250–800 Hz) and medium (800–1600 Hz) frequency bands. The sequential inversion of repeated acoustic transmissions shows a good agreement with hydrographic and geophysical data with more stable estimates than conventional metaheuristic inversion results and a reduced computational cost. The effect of filter covariance tuning is also examined and monitored with statistical tests. For large propagation modeling uncertainty, extended Kalman filter and ensemble Kalman filter are of comparable accuracy in parameter tracking, but the ensemble method should be preferred to get reliable associated uncertainty estimates.
Bottom reflections from rough topography in the long-range ocean acoustic propagation experiment (LOAPEX)
Authors: Udovydchenkov Ilya A. , Stephen Ralph A. , Duda Timothy F. , Worcester Peter F. , Dzieciuch Matthew A. , Mercer James A. , Andrew Rex K. , Howe Bruce M.
Published: October 2011

Data collected during the 2004 long-range ocean acoustic propagation experiment form a unique set of measurements containing information about absolute intensities and absolute travel times of acoustic pulses at long ranges. This work primarily focuses on sound reflected from the seafloor at the shortest transmission range of the experiment, approximately 50 km. At this range, bottom-reflected energy interferes with arrivals refracted by the sound channel. Stable bottom-reflected arrivals are seen from broadband transmissions from acoustic source at 800 m depth with 75 Hz center frequency, and from a source at 350 m depth with 68.2 Hz center frequency. Standard acoustic propagation modeling tools such as parabolic equation solvers can be successfully used to predict acoustic travel times and intensities of the observed bottom-reflected arrivals. Inclusion of range-dependent bathymetry is necessary to get an acceptable model-data fit. Although there is no direct ground truth for actual sub-bottom properties in the region, a good fit can be obtained with credible sub-bottom properties. The potential for using data of this type for geoacoustic property inversion in the deep ocean will be discussed. [Work supported by ONR.]
Finding baiji and freshwater finless porpoises in the Yangtze River, China
Authors: Akamatsu Tomonari , Dong Lijun , Wang Ding , Wang Kexiong , Li Songhai , Dong Shouyue , Zhao Xiujiang , Kimura Satoko
Published: October 2011

A stereo passive acoustic event recorder (A-tag) has been applied for range-wide monitoring of baiji and finless porpoises in China. As the pilot study, two research vessels were operated in 1700 km historic habitat of both species from Yichang to Shanghai in 2006. There was no detection of baiji, but 204 and 199 porpoises were counted acoustically by two vessels, respectively. In order to investigate the population trends of cetaceans, periodical survey is necessary. We installed A-tag on the cargo ship, which was operated 1100 km in the river once every month. An average of 6059 clicks and 95 porpoises were acoustically detected in each survey. Detected group sizes of the animals in 120-s time window were not significantly different among the surveys, but the distribution pattern suggested seasonal migration. The animals were detected in most of the survey range except two gap sections with 40 and 60 km lengths, down from Wuhan and Nanjing cities, respectively, where no animals were detected in the first three surveys. Fragmentation of population by anthropological factors was concerned. The cargo ship based passive acoustic survey was effective to monitor the distribution and population trend over time.
Acoustic monitoring of beluga whales (Delphinapterus leucas) in Cook Inlet, Alaska
Authors: Castellote Manuel , Small Robert J. , Atkinson Shannon , Lammers Marc O. , jenniges Justin , Rosinski Anne , Garner Chris , Moore Sue , Au Whitlow W. L.
Published: October 2011

Cook Inlet belugas (CIB) form a small geographically and genetically isolated population, endangered under the U.S. Endangered Species Act. They summer in the northern end of the inlet, but their wintertime distribution is essentially unknown. Factors impeding the recovery of this population over the last decade are unknown, yet could include anthropogenic activities that impact their acoustic ecology, including coastal development, oil and gas exploration, and shipping and military activities. Beginning in 2008, a cooperative research project has acquired new information on background noise levels and seasonal presence of CIB using passive acoustic monitoring. Mooring packages containing ecological acoustic recorders and echolocation loggers (C-PODs) have been deployed at 10 sites for continuous monitoring. It is a challenging environment for acoustic monitoring because of extreme tides and currents, sediment dynamics, debris from rivers, and seasonal ice. Noise from both natural and anthropogenic sources often make beluga call detection challenging. However, the effort to date has met with success and is providing valuable insights into beluga movement patterns and the acoustic environment they face. This methodology also allows monitoring other odontocetes such as killer whales (Orcinus orca), detected mostly in the lower inlet, and harbor porpoise (Phocoena phocoena) detected throughout the inlet.
Digital beam-steering of parametric array sound beams using piezoelectric micromachined ultrasonic transducers
Authors: Je Yub , Lee Haksue , Moon Wonkyu
Published: October 2011

In a previous study [H. Lee etal., J. Acoust. Soc. Am. 125, 1879–1893 (2009)], a parametric array (PA) transducer in air using micromachined ultrasonic transducers (MUT) was examined as an ultrasonic ranging sensor. It was shown that the piezoelectric micromachined ultrasonic transducer (pMUT) may be effective for PA ultrasonic ranging sensors. Since the pMUT are usually used in their array, PA source using pMUT can steer its directional sound beam by manipulating phases of radiator in the array. The beam steering capability of the PA source can provide very useful function for parametric loudspeakers as well as directional ultrasonic range sensors. In this work, 16-channel pMUT array for digital beam-steering of parametric array was designed and fabricated. Each channel of the pMUT array consisted of 9 PZT unimorph elements. The two resonance frequency-type pMUT units are designed for generating two primary beams. Beam-steering of each primary beam was controlled separately by applying complex weighting to each channel. An acoustic experiment verified the PA beam-steering of the pMUT. The result shows that the proposed pMUT is suitable for digital beam-steering of parametric array and that it is also useful for the parametric loudspeaker and the PA ultrasonic range sensor.
Protecting beyond the bone-conduction limit
Authors: Dietz Anthony J. , Audette William E. , Wilbur Jed C. , Passow Christian H.
Published: October 2011

Workers operating in extreme noise may need levels of hearing protection that are beyond that possible with double hearing protection comprising earplugs and circumaural hearing protectors. In such noise fields, sound conducted along bone-conduction transmission paths that bypass the ear canal can be sufficient to cause hearing damage. To provide sufficient protection beyond this bone-conduction limit, hearing protectors must attenuate sound that is transmitted to the cochlea via bypass mechanisms. The design of a passive hearing protection helmet that provides protection beyond the bone-conduction limit is described here. The helmet was developed for Navy aircraft carrier deck crews who are exposed to extreme sound levels of 150 dB during aircraft launch operations. The helmet design was based on extensive measurements with human subjects and with a human head simulator built to measure bone-conducted sound. These measurements demonstrated that a helmet shell fitted with an edge seal that created an acoustic seal between the shell and the head was effective in attenuating bone-conducted sound. Measured attenuation data are presented in addition to fit and performance data from tests in the laboratory and the field. Lessons learned from this development effort are also discussed. [Work sponsored by the U.S. Navy].
Ambiguity in phoneme recognition when using a bone conduction microphone
Authors: Buck Karl , Zimpfer Vronique
Published: October 2011

Bone conduction microphones are often promoted as good means for recoding voice signals in very noisy and/or harsh environments. However, it is reported in the literature that the speech intelligibility when using contact microphones is, in all noise conditions, worse than when using a boom microphone. Therefore, long term spectra of speech signals simultaneously recorded with bone and air microphones have been analyzed. This analysis shows, for all measured subjects, that the module of the transfer function between the signal recorded at the air-microphone and, because of this, at the contact microphone increases toward higher frequencies. The results also show large interindividual differences. In order to get more perception related information, a corpus of French vowels has been recorded simultaneously with air and contact microphone. This corpus has been used for a listening test aiming to show confusion between different vowels. The main confusions are for vowels having the same frequency of the first formant (e.g., [i][y][u]) which then are perceived as the central vowel. These confusions could explain the systematically lower intelligibility reported for speech when recorded with contact microphones. Planned experiments using plosive and fricative consonants should give more information about transfer mechanisms of bone conducted voice.