bluuebunny/crossref_metadata_2025 huggingface dataset:
A mechanism of tissue emulsification by high intensity focused ultrasound
Authors: Simon Julianna C. , Sapozhnikov Oleg A. , Khokhlova Vera A. , Wang Yak-Nam , Khokhlova Tatiana D. , Crum Lawrence A. , Bailey Michael R.
Published: October 2011

High intensity focused ultrasound (HIFU) has been shown to emulsify tissue in histotripsy created by cavitation clouds and millisecond boiling produced by shock wave heating; however, the mechanism by which millimeter-sized boiling bubbles or cavitation bubble clouds emulsify tissue into submicron-sized pieces is not well understood. Here, we experimentally test the hypothesis that acoustic atomization occurs and a miniature acoustic fountain forms (due to radiation force) at the edge of the millimeter-sized boiling bubble. Through high-speed photography, we observed the violent removal of bovine/porcine liver fragments upon focusing a 2-MHz transducer of 70 MPa peak positive and 15 MPa peak negative pressures at the tissue–air interface. Velocity of the fountain projectiles ranged from 5 to 15 m/s. When the focal amplitudes were reduced to 15 MPa peak positive and 7 MPa peak negative pressures, atomization occurred intermittently. The fountain projectiles were collected and examined histologically using H&E and NADH--diaphorase stains. Results show fragments ranging from tens of microns down to submicron in size. The larger fragments contain distinct tissue aggregates with intact cells and the small particles are cell fragments with few to no intact nuclei. [Work supported by NIH DK43881, DK070618, EB007643, DK007742, and NSBRI through NASA NCC 9--58.]
Capacitive micromachined ultrasonic transducers: A platform technology
Authors: Khuri-Yakub Butrus (Pierre) T.
Published: October 2011

Capacitive micromachined ultrasonic transducers (CMUTs) have been under development as alternative transducers for generating and detecting ultrasound. The advent of silicon micromachining enables the realization of the full potential of these transducers and provides performance that makes CMUTs competitive and superior to piezoelectric transducers. In immersion applications, CMUTs are possible with fractional bandwidth of over 100%, an electromechanical coupling coefficient close to unity; are made in the form of a single element or 1-D or 2-D arrays of tens of thousand of elements, as well as annular arrays. They have been operated in the frequency range of 100 kHz to 50 MHz, and included in systems with a dynamic range of the order of 150 dB/V/Hz. Custom electronics have been developed and integrated with arrays of transducers to form compact imaging catheters. In airborne applications, CMUTs have been used as mass sensors with atto-gram resolution, and in pulse echo systems operation at high temperature (600^C). This presentation will first review the operation of CMUTs, the technology used to make them, and some airborne and imaging applications.
Nickel on glass acoustic microsystems
Authors: White Robert D. , Zhao Zhengxin , Shin Minchul , Krause Joshua S. , Liu Shuangqin
Published: October 2011

An electroplated nickel-on-glass surface micromachining process has been developed and applied to fabricate condenser MEMS microphone arrays and ultrasound arrays. The major advantage of these arrays over polysilicon surface micromachined sensors is the significant reduction in stray capacitance that can be achieved by using a glass substrate. For 600 μm diameter elements with a 1.5 to 2μm sense gap, the stray capacitance per element was reduced from approximately 140–1 pF when compared to similar polysilicon devices fabricated using the PolyMUMPS process. It is important to reduce stray capacitance, since it is the driving factor for two dominant noise sources: preamplifier voltage noise and bias feedthrough noise. In addition, stray capacitance can place a limit on system bandwidth for high frequency applications such as ultrasound. In this paper, we report on cMUT ultrasound arrays for in-air Doppler applications, as well as microphone array-on-a-chip devices for aeroacoustic measurements. Characterization of system performance and comparison to previously reported polysilicon devices [Krause etal. and Shin etal., ASA Spring Meeting, 2011] is ongoing. Other properties such as material damping, structure residual stress, temperature sensitivity, and substrate charging are also under investigation. [Work supported by Spirit Aerosystems and Draper Labs.]
Piezoelectric microspeakers built on various diaphragms
Authors: Kim Eun S.
Published: October 2011

This paper describes and compares four different types of diaphragm-based piezoelectric microspeakers built on (1) a compressively stressed silicon nitride diaphragm, (2) a parylene diaphragm, (3) a dome-shaped silicon nitride diaphragm, and (4) a PZT bimorph diaphragm. The main innovation in the first device is the usage of a wrinkled diaphragm that supports a flat diaphragm, where piezoelectric actuation happens, and allows a large bending displacement. The second device is to exploit the very low elastic modulus of parylene (a polymer material), and is built on a 1.5 μm thick parylene diaphragm with electrodes and piezoelectric ZnO film. Also described in this paper is an acoustic transducer built on a 1.5 μm thick dome-shaped silicon nitride diaphragm (2 mm in radius, with a circular clamped boundary on a silicon substrate) with electrodes and piezoelectric ZnO film. The dome diaphragm is shown to effectively release residual stress through volumetric change of its shape. Finally, this paper describes a microspeaker (composed of 8 mm square PZT bimorph and bulk-micromachined silicon) that shows flat diaphragm displacement from DC to 8 kHz. A bimorph diaphragm is formed by gluing two 127 μm thick PZT sheets and attaching them to a micromachined silicon substrate.
Piezoelectric thin films for a high frequency ultrasound transducer with integrated electronics
Authors: Griggio Flavio , Kim Hyunsoo , Kim Insoo , Jackson Thomas N. , Choi Kyusun , Tutwiler Richard L. , Trolier-McKinstry Susan
Published: October 2011

Lead zirconate titanate (PZT) piezoelectric films were integrated into prototype one-dimensional array transducers. Linear arrays of diaphragm transducers were prepared using PZT films of 0.5–1.7 μm in thickness and surface micromachining techniques. For this purpose, the PZT and remaining films in the stack were patterned using ion-beam or reactive ion etching and partially released from the underlying silicon substrate by XeF2 etching. The PZT films were prepared by chemical solution deposition, and have e31,f piezoelectric coefficients of −6 to −12 C/m2, depending on the crystallographic orientation. Impedance measurements on the fabricated structures showed resonance frequencies between 3 and 70 MHz for fully and partially released structures depending on the transducer dimensions and vibration modes. In-water transmit and receive functionalities have been demonstrated. A bandwidth on receive of 66% has been determined. Because of the small thickness of the piezoelectric element, the elements can be driven at less than 5 V. This enables the ultrasound system to be CMOS compatible, and hence massive miniaturization. A custom designed CMOS chip which enables beamforming on transmit, transducer excitation, amplification, digitization, and data storage was designed and fabricated.
Minimizing noise in micromachined piezoelectric microphones
Authors: Littrell Robert , Grosh Karl
Published: October 2011

Piezoelectric microelectromechanical systems (MEMS) microphones have been researched for over 30 yr because they are relatively easy to build, output a signal without any biasing circuitry, and are relatively linear. The primary impediment to mass utilization of piezoelectric MEMS microphones has been the noise levels of these devices, which have been unacceptably high. The input referred noise of most piezoelectric MEMS microphones is greater than or equal to 55 dB(A) while commercial capacitive MEMS microphones typically have noise floors between 32 and 38 dB(A), roughly ten times lower. In order to achieve competitive noise levels in a piezoelectric MEMS microphone, a systematic approach to mechanical and electrical optimizations must be used. A key microphone metric for this optimization is acoustically generated electrical energy, as opposed to sensitivity. This optimization can be used to determine the minimum achievable noise floor for piezoelectric sensors with relatively few assumptions. Further, this optimization allows for the determination of materials, electrode patterns, layer thicknesses, and mechanical structures that minimize the noise. These models have been used to design and build piezoelectric MEMS microphones with a noise floor of 34 dB(A) and indicate that further improvements can be achieved.
Electroacoustic parameter extraction of a piezoelectric microelectromechanical systems microphone
Authors: Williams Matthew D. , Griffin Benjamin A. , Reagan Tiffany N. , Sheplak Mark
Published: October 2011

A novel suite of parameter extraction experiments were used to assess the accuracy of individual elements of a lumped element model for a microelectromechanical systems based piezoelectric microphone. The MEMS microphone was developed via model-based design utilizing the lumped element model for use in aeroacoustic applications. Laser vibrometer scans of the microphone diaphragm while subjected to electrical or pressure excitation provided experimental predictions for the effective electroacoustic piezoelectric coupling coefficient, diaphragm compliance, and mass. The experimental results were compared with analytical predictions from a piezocomposite diaphragm model for these individual lumped elements. Associated lumped element model predictions were also compared with the results of device characterization experiments. Similar trends in theory and experiments were observed, though comparative error in element values was attributed to uncertainty in model-inputs, most notably thin-film residual stresses in the microphone diaphragm. [This work was sponsored by Boeing Corporation.]
Ultrasonic analysis of breast tissue for pathology classification
Authors: Sorensen Kristina M. , Doyle Timothy E.
Published: October 2011

The effectiveness of breast conservation surgery (BCS) or lumpectomy relies heavily upon pathology to assure negative or cancer free margins. In a study to develop an intraoperative pathology method, surgical specimens from 17 breast cancer patients were tested with high-frequency (HF) ultrasound (20–80 MHz) to search for pathology sensitive features for the detection of cancer in margins during BCS. Pulse-echo and pitch-catch waveforms were obtained using two single-element 50-MHz transducers. Analysis of time-domain waveforms yielded ultrasonic attenuation and sound speed, whereas fast Fourier transforms of the waveforms produced ultrasonic spectra and cepstra for the evaluation of spectral peak density and cepstrum slope. Spectral peak density indicated significantly higher values for carcinomas and precancerous pathologies than for normal tissue. Cepstrum slope exhibited a substantial distinction between benign and adipose tissues when compared with normal and malignant pathologies. The attenuation coefficients were sensitive to fat necrosis, fibroadenoma, and invasive lobular carcinoma. A multivariate analysis of these parameters was used to further distinguish pathologic classification. Evaluation of ultrasonic attenuation, spectra, and cepstra permits differentiation between normal, adipose, benign, and malignant breast pathologies. These results indicate that HF ultrasound may assist in eliminating invasive re-excision for lumpectomy patients. [Work supported by NIH R21CA131798.]
Noise reduction in deployable infrasound arrays through unconventional clover hose layouts
Authors: Clay John R. , McKenna Mihan H.
Published: October 2011

Infrasound is acoustic energy that lies below the human hearing threshold. This energy is produced from numerous naturally occurring and man-made sources with the ability to propagate hundreds to thousands of miles, depending on the source strength. Infrasound sensors are commonly deployed in networked, multi-sensor arrays for signal localization and characterization. Deployable infrasound sensors, such as Chaparral and IML rely on the use of soaker hoses for noise reduction and long-wavelength spatial averaging. Traditionally, sets of 25–50 ft hoses attach to the ports on each side of the infrasound sensor and are terminated with a cap at the end of the hose. In August of 2010, hose filter geometry testing was conducted using IML ST infrasound sensors designed to minimize the filter footprint by exploring alternatives to the conventional straight hose layouts while maximizing signal-to-noise in desired pass-bands. The optimal arrangement, “Clover Hose Layout,” removed the terminator at the end of the hose and looped it to connect to an adjacent port with substantial environmental noise reduction in the 0.5–50 Hz range. Results are presented comparing a suite of hose arrangements including conventional straight hose layouts under calibrated source testing.
Dynamic behavior of microwave-cured guitar wood
Authors: Lozupone David-M. , Yasay Aldean J. , Dick David A. , Beaudry Allen R. , Sorenson Andrew T. , Vera Stalin L. , Celmer Robert D.
Published: October 2011

An ongoing investigation has been conducted to measure and analyze vibration properties of guitar-body wood samples with different wood finish curing methods for Taylor Guitars. The current method uses ultraviolet radiation to cure a lacquer finish. Virginia Tech's (VT) Material Science & Engineering Department has proposed an alternative method using microwave radiation to cure the finishes, now in its third stage of development. Microwaves (MW) have longer wavelengths than ultraviolet (UV) radiation, allowing greater penetration depth and increased diffusion between wood and lacquer molecules. To investigate the effect of the finish on dynamic behavior, modal analyses have been conducted at the University of Hartford Acoustics Engineering Laboratory using unfinished, UV-cured, and MW-cured guitar wood samples (spruce, maple, ash, and rosewood). Samples were characterized at bending and torsional modes up to 3.2 kHz. Overall results to date demonstrate that both UV-cured and MW-cured samples have higher damping than unfinished samples. Further, the damping provided by the MW method appears to correlate with the number and thickness of filler, finish and top coats. The current results suggest it is possible to produce similar dynamic behavior to UV curing using the MW technique. [Work done in collaboration with Virginia Tech and Taylor Guitars.]
Application of time reversal to sources in non-ideal environments
Authors: Harker Blaine , Anderson Brian
Published: October 2011

Time reversal (TR) is a technique of localizing acoustical sources using a time reversal mirror (TRM) and is especially useful in reverberant environments. TR is commonly used to find acoustically small sources and in many cases pulsed waveforms are used. Here TR is applied to distributed sources using long duration waveforms. This is done using a straightforward, theoretical, point source propagation computer modeling and data from experimental measurements of jet noise. The quality of the TR focusing versus using different numbers of microphones to constitute the TRM is determined. These results are compared to other imaging techniques and theories of sound radiation from jets.
Where is the expressivity in singing?
Authors: Sundberg Johan
Published: October 2011

When computers convert music scores to sounding music, performances void of musical expression typically emerge, mostly perceived as musical disasters. This is a striking illustration of the importance of the contributions of musicians. Excellent tools for exploring these contributions are synthesized performances and/or processing of real performances. In the 1970s, the KTH music group started to formulate rules for music performance, now integrated into the KTH Director Musices performance grammar. Some principles underlying its performance rules can be identified: (1) marking of phrase structure, (2) increasing contrasts along acoustic parameters, and (3) emphasizing important and/or less expected notes. These principles can be observed also in phonetics and speech prosody. Also the acoustic codes used for these principles in music performance are similar to those used in speech. Just as pitch is raised in agitated speech, singers have recently been found to increase contrast along the pitch dimension by sharpening the peak note in agitated phrases. In a recent experiment, audio processing software was used to quasi-correct the intonation of the peak tone in a handful of examples sung by a professional baritone singer. A listening test indicated that the sharpening added to the expressivity of the performance.
Synthesizing melodic expression
Authors: Raphael Christopher S.
Published: October 2011

A method for expressive melody synthesis is presented seeking to capture the prosodic (stress and directional) element of musical interpretation. An expressive performance is represented as a note-level annotation, classifying each note according to a small alphabet of symbols describing the role of the note within a larger context. An audio performance of the melody is represented in terms of two time-varying functions describing the evolving frequency and intensity. A method is presented that transforms the expressive annotation into the frequency and intensity functions, thus giving the audio performance. The problem of expressive rendering is then cast as estimation of the most likely sequence of hidden variables corresponding to the prosodic annotation. Examples are presented on a data set of around 50 folk-like melodies, realized both from hand-marked and estimated annotations.
Virtual instrument player using probabilistic gesture grammar
Authors: Lindemann Eric
Published: October 2011

Expressive instruments such as violin require subtle control gestures: variations in bow velocity and pressure over the course of a note, changes in vibrato depth and speed, portamento gestures, etc. Music synthesizers that attempt to emulate these instruments are often driven from keyboard controllers or directly from score editor or sequencer software. These synthetic control sources do not generally provide the level of detailed control required by expressive instruments. Even if a synthesizer offers a rich set of realistic continuous control inputs, the effort and skill required by the user to manage these controls, e.g., by drawing expression and vibrato controller curves in a MIDI sequencer, is considerable. Often the user would prefer to treat the synthesizer as a combination instrument and virtual player. The virtual player receives limited score-like control input and infers the more detailed gestural control. This presentation describes a virtual player mechanism that employs a probabilistic note-gesture grammar to aid with control inference. The virtual player uses the grammar to select various gestural patterns that are idiomatically appropriate given the limited control input. The virtual player is used to drive a concatenative music synthesizer employing a rich database of recorded note gestures.
Comparison of three acoustics test fixtures for impulse peak insertion loss
Authors: Murphy William J. , Flamme Gregory A. , Meinke Deanna K. , Finan Donald S. , Lankford James , Khan Amir , Sondergaard Jacob , Stewart Michael
Published: October 2011

Acoustic test fixtures (ATF) for testing the impulse peak insertion loss (IPIL) of a hearing protector are described by American National Standard ANSI S12.42-2010. The self-insertion loss, ear simulator design (canals, microphone, and temperature), hardness of the area surrounding the pinna, and the anthropometric shape of the head has been specified in the standard. The IPILs of four protector conditions were evaluated with three ATFs during an outdoor field study using firearm noise. The Etymotic Research ER20 musicians' earplug and electronic (EB1 earplugs), the Peltor Tactical Pro earmuffs, and a combination of the TacticalPro and ER20 protectors were tested at 130, 150, and 170 dB peak sound pressure level with the Institute de Saint Louis heated and unheated fixture and the GRAS 45CB heated ATF. IPILs exhibited good agreement across all three fixtures for earplugs. Significant differences were observed between the fixtures for the earmuff-only condition. These differences were more evident for the double-protection condition. [Portions of this work were supported by the U.S. EPA Interagency Agreement DW75921973-01-0.]
Insertion-loss and transfer-function performance of two new acoustical test fixtures complying with ANSI S12.42-2010, relative to performance of prior test fixtures and to real-ear data
Authors: Berger Elliott H. , Kieper Ronald W. , Stergar Michael E.
Published: October 2011

The most recent American National Standards Institute (ANSI) standard for the measurement of the insertion loss of hearing protection devices (HPDs), ANSI/American Standards Association (ASA) S12.42-2010, specifies a new-concept acoustical test fixture (ATF). It is similar to some existing ATFs but differs in terms of the required earcanal length, inclusion of a simulated flesh lining the earcanal, and a heater to bring the test fixture to approximate body temperature. These features were deemed necessary to develop a device that provides insertion loss data with reasonable correspondence to performance on human heads, as the ATF is the preferred method in the standard for tests on certain electronic earplugs and for all impulse testing. Within a year of the issuance of the standard, at least two ATFs [one produced by G.R.A.S. Sound and Vibration and the other produced by the Institute of St. Louis] became available. The studies reported herein will provide an initial evaluation of these two heads compared to prior art, based on ATF insertion-loss measurements for a sample of passive earplugs and earmuffs versus real-ear attenuation at threshold per ANSI S3.19-1974. Additionally, the ATFs’ transfer function of the open ear in a diffuse field will also be reported.
ANSI S12.42-2010 measurements of impulse peak insertion loss for passive hearing protectors
Authors: Michael Kevin , Schmitt Jeff G.
Published: October 2011

Until recently, a standardized measurement procedure to determine peak insertion loss for hearing protectors was not available. This has led to confusion and uncertainty for hearing protector users who commonly use the devices in impulse noise, such as gunfire. Released in 2010, ANSI S12.42-2010 defines a test method and analysis procedure for measuring hearing protector impulse peak insertion loss. The required test fixture has recently become commercially available and laboratories are gaining experience making these measurements. Impulse peak insertion loss data will be presented for a variety of hearing protector types along with a description of the measurement procedure.
Formulation and applications of an integral-equation approach for solving scattering problems involving an object consisting of a set of piecewise homogeneous material regions
Authors: Bleszynski Elizabeth , Bleszynski Marek , Jaroszewicz Thomas
Published: October 2011

The paper presents the formulation and selected applications of the surface integral equation approach for finding pressure, displacement, and traction fields in a complex object consisting of a set of piece-wise homogeneous regions characterized by different Lame' material parameters. Representative applications of the approach are presented, which involve finding pressure field distribution inside human inner ear treated as an inclusion embedded in the surrounding inhomogeneous material. (the embedding region is treated with volumetric integral equations with suitable coupling to the inclusion). The method uses a set of coupled elastodynamics integral equations for two unknown fields: displacement and traction at each interface, which allow us to find displacement and traction field distribution inside a general complex object composed of sub-regions with different material parameters. The matrix elements of strongly singular kernels appearing in the integral equations were converted, through suitable integration by parts, to equivalent forms involving weakly singular integrands having at most 1/r singularity. Applications of the method are presented in simulation of acoustic and elastic fields in a human head and in the cochlea region. [This work is supported by AFOSR.]
The affect of epilarynx tube dimensions on glottal airflow
Authors: Titze Ingo R.
Published: October 2011

Since the early 1980s, it has been known that an inertive supraglottal acoustic load skews the peak of the glottal flow pulse. It also lowers the phonation threshold pressure. The skewing in turn affects the vocal intensity by increasing the rate of flow declination. High frequencies in the source spectrum are emphasized. More recent is the discovery that the airway between the vocal folds and the aryepiglottic folds, often called the epilarynx tube, provides much of the inertance in a vowel production. This presentation shows how epilarynx tube geometry skews the glottal flow pulse and thereby may affect voice quality.
An ex vivo perfused human larynx for studies of human phonation
Authors: Berke Gerald S. , Howard Scott , Mendelsohn Abie
Published: October 2011

For many years, non-human animal models and excised human larynges have been used to study the physiology and biomechanics of human laryngeal function. While these studies have provided a wealth of information, experimental data concerning actual human laryngeal function have been limited by the inability to parametrically study the complete set of variables that controls vocal fold function in a living human. This presentation describes the development of a perfused ex vivo human larynx. In this model, a human larynx is harvested at the time of organ donation and perfused with the donor's blood to maintain viability of its nerves and muscles, which can then be stimulated as in an in vivo preparation. This model allows parametric, experimental control of the larynx (including stimulation of the thyroarytenoid muscle) while it is still virtually living. Applicability of this new laryngeal model to studies of physiology, biomechanics, and to treatment of human disease will be discussed.
Restraining mechanisms in regulating glottal closure during phonation
Authors: Zhang Zhaoyan
Published: October 2011

Recent experimental studies showed that isotropic vocal fold models were often blown wide apart and thus not able to maintain adduction, resulting in voice production with noticeable breathy quality. This study showed that the capability of the vocal fold to resist deformation against airflow and maintain position can be improved by stiffening the anterior-posterior tension in either the body-layer or the cover-layer of the vocal folds, which presumably can be achieved through the contraction of the thyroarytenoid and cricothyroid muscles, respectively. Experiments in both physical models and excised larynges showed that, when these restraining mechanisms were activated, the vocal folds were better able to maintain effective adduction, resulting in voice production with much clearer quality and reduced breathiness. [Work supported by NIH.]
The effects of age on speech production in a non-pathological system: A case study of 48 years
Authors: Hunter Eric J. , Brown Wesley R. , Pead Patrick , Engar Megan
Published: October 2011

The current study examines 36 recordings (mean: 30 min) of a male speaker, spanning the years 1958 to 2007. These recordings provide a rare opportunity to track a single individual's age-related voice and speech mechanism changes (48–98 y/o). Aging effects became noticeable between the ages of 68–74, indicating a fundamental change in the body's maintenance of the speech mechanism, implying the likely onset of other aging symptoms such as swallowing problems. Voice sF0, which usually begins to drop during puberty, continued to decrease for this individual until 70 y/o; from 70 to 98 y/o, average vocal F0 then increased from 140 to 160 Hz. Rate of speech (syll/min) began to decrease precipitately at 78 y/o (175 to 165 per/min.), while length of speech breathing reduced from 70 y/o (35 to 25 phonemes per breath group). The results of this case study can be used as a baseline for future studies. The aging of the voice and speech mechanism affects breathing, swallowing, and communication. The world's 50+ population is the fastest growing segment, affecting society by its sheer number as well as by historically high life spans. Health-care practitioners must understand and accommodate the needs of this population.
Attention modulates the time-course of talker-specificity effects in lexical retrieval
Authors: Theodore Rachel M. , Blumstein Sheila E.
Published: October 2011

Listeners retain in memory phonetic characteristics associated with individual talkers. It has been proposed that talker-specificity effects, while robust, emerge late in processing, reflecting increased time to activate lower-frequency instantiations for individual talkers. Here we test the hypothesis that attention to talker characteristics modulates the time-course of talker-specificity effects. Two groups of listeners participated in encoding and recognition phases. During encoding, all listeners heard words produced by two talkers, but only half of them were required to identify talker gender. During recognition, all listeners heard a series of words and indicated whether each word had been presented during encoding. Critically, some of the words were previously presented in the same voice and some in a different voice. Results showed no difference in mean hit rate or reaction time between the two groups, indicating that the encoding manipulation did not globally affect recognition memory or response latency. However, the latency data showed an interaction such that a talker-specificity effect emerged only for listeners who identified talker during encoding. These results indicate that the talker-specificity effect reflects an attention shift, not increased processing time, consistent with a dynamic perceptual system where attention to specific aspects of the signal can influence representational frequency.
Use of waveform mixing to synthesize a continuum of vowel nasality in natural speech
Authors: Styler William F. , Scarborough Rebecca , Zellou Georgia
Published: October 2011

Studies of the perception of vowel nasality often use synthesized stimuli to produce controlled gradience in nasality. To investigate the perception of nasality in natural speech, a method was developed wherein vowels differing naturally in nasality (e.g., from CVC and NVN words) are mixed to yield tokens with various degrees of nasality. First, monosyllables (e.g., CVC, NVC, CVN, NVN) matched for vowel quality and consonant place of articulation were recorded. The vowels from two tokens were excised, matched for amplitude, duration and pitch contour, and then overlaid sample-by-sample according to a specified ratio. The resulting vowel was spliced back into the desired consonantal context. Iterating this process over a series of ratios produced natural-sounding tokens along a continuum of vowel nasality. Acoustic measurements of the nasality of output tokens [using A1-P0 (Chen, 1997)] confirmed a relation between the ratio used and the nasality of the output. Stimuli created in this manner were used in a perception experiment (Scarborough etal., 2011) where degree of nasality affected perception without any slowdown in response time for nasality-altered vs. unaltered tokens. This demonstrates the quality of the nasal-altered stimuli and suggests the potential usefulness of this process for other speech perception studies.
Classification of foreign accents
Authors: Atagi Eriko , Bent Tessa
Published: October 2011

Regional dialects and foreign accents account for a significant source of between-talker variability. The auditory free classification task—in which listeners freely group talkers based on audio samples—has been a useful tool for examining listeners' cognitive representations of regional dialects (e.g., Clopper and Pisoni, 2007) and was employed here to examine perceptual representation of foreign accents. This task assesses listeners' perception of variation without imposing category structure. In the present study, native listeners completed four free classification tasks in which they grouped 24 talkers from six native language backgrounds by perceived native language. The four tasks varied in overall sentence intelligibility and whether four native talkers were also included, to explore how accent level and presence of native language exemplars influence classification. Categorization performance was less accurate with lower intelligibility sentences, even when sentence content was known, but more accurate when native talkers were included. Thus, listeners attend to phonological features that characterize differences across foreign accents, and are more attentive to these features when accented and native speech can be directly compared. Moreover, perception of lower intelligibility sentences may impose a higher cognitive load, thus restricting the cognitive resources available for classification. [Work supported by NIDCD R21DC010027.]
Does second language phonological acquisition affect native language word recognition?
Authors: Woodley Melinda D. , Tice Marisa , Sumner Meghan
Published: October 2011

Research with monolingual adults has shown that words whose initial stop consonants have canonical VOTs show larger semantic priming effects than words beginning with phonetically non-canonical stops; the prime [k h ](long-lag) facilitates recognition of a semantically related target (queen) more than [k] (short-lag), though both facilitate target recognition (Andruski etal., 1994; van Alphen and McQueen, 2006). The present study asks how the acquisition of a second (L2) phonological system affects word recognition in the native language (L1). A longitudinal semantic priming study ([k]... queen) is being administered to American students in Paris at the beginning, middle, and end of their 4–6 weeks of immersion class enrollment. Two potential patterns may emerge: (1) exposure to instances of French short-lag /k/ may lead to increased facilitation of short-lag [k] primes, indicating an early established link between shared L1 and L2 phonological categories or (2) since stops in the short-lag VOT range are categorized differently in French versus English ([k] = /k/ vs. /g/), increased ambiguity introduced by the L2 system may incur a greater cost for non-canonical variants, causing [k] to become a less effective prime. Both findings have implications for theories of speech perception and representation.
Perceptual accommodation to rate, talker, and context in Mandarin
Authors: Sereno Joan A. , Lee Hyunjung , Jongman Allard
Published: October 2011

The present research examines how perceptual constancy is achieved by exploring three fundamental sources of acoustic variability: changes in the rate at which a sentence is spoken, differences due to which speaker produced the sentence, and variations resulting from the sentence context. These sources of variability are investigated by examining the processing of Mandarin tone (specifically Tone 2 and Tone 3). Acoustic analyses (fundamental frequency and duration) of tone productions of 12 native speakers were conducted. These data were then used to construct stimuli to evaluate perceptual adjustments to speaking rate, talker, and context. 14 target stimuli were created varying in temporal, spectral, and both temporal and spectral characteristics. These target stimuli were preceded by six different precursor sentences representing different speaking rates, speakers, and contexts. Participants were asked for their tone judgments. Analyses show interactions between precursor type and target manipulation. Specifically, normalization primarily occurs when precursor and target vary along the same acoustic dimension. However, variation in coarticulatory context violated this pattern. These experiments will be discussed in terms of the use of temporal and spectral cues and its interaction with listener normalization of speaking rate and talker. [Research supported by NSF.]
Perceptual shifting versus perceptual learning: How listeners accommodate accented speech
Authors: King Ed , Sumner Meghan
Published: October 2011

Variation is ubiquitous in speech. Differences among individuals ensure that no two speakers pronounce the same sound the same way. Listeners nonetheless understand variable pronunciations as instances of the same category, adapting to familiar variation (e.g., accents they have heard before) or learning novel variants (e.g., unfamiliar accents). Perceptual learning, in which listeners hear features ambiguous between two phonemes and subsequently shift their perceived phoneme boundary, is claimed as a mechanism for accent learning (Eisner and McQueen, 2005). Work examining adaptation to familiar variation has found that listener expectations about speaker dialect influence speech perception (Hay etal., 2010). This study investigates adaptation to a familiar accent, British English, by testing American listeners' identification of “er” on an [e]-[3] continuum before and after hearing a series of British-accented words without the phoneme /3/. The perceptual learning paradigm predicts that training should shift the 50% “er”-identification crossover toward [3], with listeners accepting more [3]-like sounds as “er” after hearing British English. This approach is novel because none of the training words contain [3], suggesting that, unlike traditional perceptual learning, adaptation to a phoneme in a familiar accent does not require immediate exposure to that phoneme.
Listening to a novel foreign accent, with long lasting effects
Authors: Bardhan Neil P. , Weber Andrea
Published: October 2011

In conversation, listeners frequently encounter speakers with foreign accents. Previous research on foreign-accented speech has primarily examined the short-term effects of exposure and the relative ease that listeners have with adapting to an accent. The present study examines the stability of this adaptation, with seven full days between testing sessions. On both days, subjects performed a cross-modal priming task in which they heard several minutes of an unfamiliar accent of their native language: a form of Hebrew-accented Dutch in which long /i:/ was shortened to /I/. During this task on Day 1, recognition of accented forms was not facilitated, compared to that of canonical forms. A week later, when tested on new words, facilitatory priming occurred, comparable to that seen for canonically produced items tested in both sessions. These results suggest that accented forms can be learned from brief exposure and the stable effects of this can be seen a week later.
Investigating the roles of vocal tract size and phoneme content in context effects
Authors: Vitela A. Davi , Lotto Andrew J.
Published: October 2011

In this series of experiments, we examined the role of specific vocal tract and phoneme contextual information on vowel categorization. An articulatory synthesizer was used to create vowels produced by specific vocal tract lengths (VT) of different sizes [Story, J. Acoust. Soc. Am. 117, 3231-3254 (2005)]. Contexts of repeated vowels (/a/ or /i/) produced by long and short VTs preceded a target series that changed perceptually from /bit/-/bet/. If listeners are extracting information about the VT itself, similar context effects should be elicited by either vowels produced by the same VT. If listeners are comparing the ambiguous target to the placement of a known vowel, one might predict similar results for the /aa/ condition or the /ii/. However, a complete cross-over interaction was obtained such that the effect of the context could not be predicted by the VT size or the vowel identity, but instead on the acoustic characteristics of F1. Follow-up experiments used contexts consisting of cross-cutting vowels from both VTs or two different vowels from the same VT. The results suggest that the order of the appended stimuli matters, demonstrating that listeners are not averaging information equally across the entire context. [Work supported by NIH-NIDCD.]
Contributions of static and dynamic spectral cues to vowel identification by cochlear implant users
Authors: Donaldson Gail S. , Rogers Catherine L. , Cardenas Emily S. , Russell Benjamin A. , Hanna Nada H.
Published: October 2011

Relatively little is known about cochlear implant (CI) users' ability to make use of static versus dynamic spectral cues in vowel perception tasks. The present study measured vowel identification in CI users and young normal hearing (YNH) listeners using naturally produced /dVd/ stimuli (deed, did, Dade, dead, dad, dud, and Dodd). Vowel identifcation was tested for (1) the unmodified syllables, (2) syllables modified to retain only 60 or 80 ms of the vowel center (center-only conditions), and (3) syllables modified to retain only 30 or 40 ms of the initial and final vowel transitions, with vowel duration neutralized (edges-only conditions). YNH listeners achieved high levels of performance for the unmodified stimuli (avg. 99.8%) and for the center-only stimuli (90.8%); their performance dropped to more moderate levels (68.1%) for the edges-only stimuli. CI users demonstrated moderate performance for the unmodified stimuli (avg. 72.0%) but demonstrated substantially poorer performance for both the center-only (41.1%) and edges-only stimuli (27.8%). Findings suggest that CI users (1) have difficulty identifying vowels in syllables when one or more cues are absent and (2) rely more strongly on quasi-static cues from the vowel center as compared to dynamic cues at the syllable edges.
Perceptual deficits in Thai sensorineural hearing loss patients
Authors: Tantibundhit C. , Onsuwan C. , Rueangwit N. , Saimai N. , Saimai T. , Thatphithakkul N. , Kosawat K. , Thatphithakkul S. , Chootrakool P.
Published: October 2011

This study explored differences in CVV perception in two groups of Thai listeners: with normal hearing and with sensorineural hearing loss (with/without hearing aids). All participants chose one response in each of 210 Thai stimulus rhyming pairs, e.g., /taa/-/naa/. The rhyming monosyllabic words share an /aa/ vowel and mid tone, but differ in their initial phonemes (symmetrically distributed across 21 phonemes). While all stimuli for the normal hearing group were embedded in 4 signal-to-noise ratio levels, clean stimuli were presented to the patients. Comparisons of confusion patterns and perceptual distance were made. In both groups, /r/ is the most confusable phoneme, while /w/ is among the least. Perceptual representations of initial phonemes show five individual clusters: glide, glottal constriction, nasality, aspirated obstruent, and a combination of liquid and unaspirated obstruent. Patients' perceptual difficulty could be attributed to the nasality grouping, which is normally well separated, shifting closer to the glottal constrictions and aspirated obstruents. Hearing aids seem to improve perception of all phonemes by 10%, with /kh/ and /h/ showing the highest improvement rate, and /d/ the lowest. The instruments are beneficial in moving the nasality cluster further away from the nearby groupings.
Talker and gender effects in induced, simulated, and perceived stress in speech
Authors: Harnsberger James , Betzen Christian , Perry Kristen , Hollien Harry
Published: October 2011

Prior modeling on the speech correlates of psychological stress have typically reported averaged data in which the problem of quantifying degree of stress was not adequately addressed. Harnsberger, Kahan, and Hollien (2006) reported a speech database in which stress was quantified both with (two) physiological measures and (two) self-report scales. In this study, these materials were assessed in two stress perception experiments, involving ratings and classification, and the subsequent analysis was organized by gender and individual talker. Several observations were made. First, both induced and simulated stress samples were perceptibly different from baseline, unstressed samples. Second, up to one third of talkers in a given gender group differed from the overall trend, showing instead a propensity to be rated as less stressed during high degrees of physiological arousal (reflected also in self-report scales); essentially, sounding calmer under stress than the baseline condition. Third, female talkers showed the expected positive correlation between the degree of physiological shift under stress versus the shift in self-reported stress ratings while, in male subjects, higher degrees of physiological arousal were underreported in self-report (a “tough guy” response). Finally, acoustic modeling efforts were conducted by gender and stress response type, for induced, simulated, and perceived stress.
Broadband model-based tracking
Authors: Sullivan Edmund
Published: October 2011

Joint range and bearing estimation using a Kalman-based recursive processor has already been achieved for the narrow band case. The signal is modeled as having a circular wavefront curvature, and the range is estimated as the radius of this curvature. The bearing is also jointly estimated along with the range. Since there is a direct estimation of the range, a maneuver of the towed array receiver is not necessary. To extend this procedure to the broadband case, a preprocessed measurement is computed using a sequence of contiguous Fourier transforms. The phases of the associated frequency components are then aligned, providing one effective frequency. This allows an effective time delay to be computed for each Fourier transform, thus allowing a sequential estimation of the bearing and range to be carried out. A further advantage of this approach is that, since the array is in motion, the Doppler of the effective frequency can be included in the signal model, thereby allowing the additional bearing information in the Doppler to be exploited. This extra bearing information significantly decreases the variances on the estimated parameters, providing a synthetic aperture effect. This permits the use of a relatively short array aperture.
Exploiting multiple sensor modalities for classification of underwater munitions: A Dempster–Shafer theoretic approach
Authors: Wickramarathne Thanuka L. , Premaratne Kamal , Negahdaripour Shahriar , Brisson Lisa N. , Beaujean Pierre P.
Published: October 2011

Detection and classification of underwater UXOs (UneXploded Ordnance) is a task that is receiving considerable attention from the DoD and related agencies that are involved in management of military munitions. Achieving a reasonable accuracy in detection and classification of these objects is extremely difficult mainly due to the harsh cluttered underwater environment. It is now an accepted fact that no single sensing technology can be both accurate and cost-effective. Low visibility in underwater exposes a significant limitation to optical cameras which are usually better suited for identifying the physical shape of objects. While acoustic imaging is a good alternative, the characteristics of imaging and physical system artifacts make the object recognition task non-trivial. Multi-sensory fusion provides an avenue to exploit the strengths of individual sensors and modalities while mitigating their weaknesses. We address the problem of fusion of multiple sensory information for the task of underwater UXO detection via the use of a Dempster–Shafer (DS) theoretic evidence fusion scheme. The DS theoretic foundation of our fusion algorithm also enables it to incorporate domain expert opinions and knowledge from databases to assist in the detection and classification tasks. We illustrate this method via the use of real data obtained at a test site located in the Florida Atlantic University premises.