bluuebunny/crossref_metadata_2025 huggingface dataset:
Vibro-acoustography beam formation with reconfigurable arrays
Authors: Urban Matthew W. , Kamimura Hermes A. S. , Carneiro Antonio A. O. , Fatemi Mostafa , Alizad Azra
Published: October 2011

Vibro-acoustography (VA) is an imaging modality that measures the acoustic response from stimulation produced by the interaction of two ultrasound beams at different frequencies. In this work, we present a numerical study of the use of reconfigurable arrays (RCA) for VA beam formation. A parametric study of the aperture selection, number of channels, number of elements, focal distance, and steering parameters is presented in order to show the feasibility and evaluate the performance of VA imaging based on RCA. Furthermore, an optimization for beam steering based on the channel assignment is proposed for balancing the contribution of the two waves in the steered focus. The point-spread function is calculated based on angular spectrum methods using the Fresnel approximation for rectangular sources. Simulations considering arrays with 50 × 50 to 200 × 200 elements with the number of channels varying in the range of 32 to 128 are evaluated to identify the best configuration for VA. We concluded that RCA transducers can produce spatial resolution similar to confocal transducers, steering is possible in elevation and azimuthal planes, and effective setting parameters including number of elements, number of channels, maximum steering, and focal distance are suggested for VA clinical imaging.
State space modeling of surface mount silicon microphones
Authors: Onaran A. Guclu , Garcia Caesar T. , Liu Alex , Christensen Matt , Kuntzman Michael , Kirk Karen , Hall Neal A.
Published: October 2011

Most surface mount microelectromechanical system (MEMS) microphone packages are similar in construction, consisting of a printed circuit board with sound inlet, a MEMS die with a through-wafer etch aligned over the sound inlet, and cap which serves to protect the structure and render an enclosed back volume. From a lumped modeling perspective, this system is a network of acoustical and mechanical elements. Network models (i.e., equivalent circuit models) have proven to be the most common modeling technique for simulating important features of these microphones, including frequency response functions and internal noise floors. While these models have many advantages including their ability to be solved efficiently using modern circuit simulation software, they do not lend themselves well to an understanding of system dynamics as a decomposition of the fundamental mechanical modes of the packaged system. We present a state space model for complete MEMS microphone packages and present frequency response simulations as a superposition of the system's eigenmodes. In addition to offering insight into package behavior, we believe these models are better equipped to address advanced features such as feedback altered dynamics using internal actuation capabilities. Simulations are compared with measurements on a surface mount optical MEMS microphone discussed prior.
Investigations of the just-noticeable-difference of clarity index (C80)
Authors: Jasinski Christopher M. , Ahearn Meghan J. , Giacomoni Clothilde B. , Ormsbee Caitlin I. , Schaeffler Matthew J. , Wells Adam P.
Published: October 2011

The clarity index for music (C80) is a valuable room acoustic parameter, as it is an objective measure of how listeners perceive clarity. Knowing the just noticeable difference (JND) of C80 is of great importance to concert hall designers. Limited previous research has been conducted to find the C80 JND, and the major studies have limitations, including a small sample size. The reported JND is approximately 1 dB. An ongoing investigation is being conducted at the University of Hartford to establish the validity of these results by experimentally determining the JND, including an investigation on test method. In the first study (Ahearn etal., 2009), the subject pool was increased to 51 subjects, using test methods from prior research. The next study (Giacomoni etal., 2010) compared two C80 JND test methods, including allowing subjects to switch between signals in real-time. The results revealed the importance of subject training. In the third study (Wells etal., 2010), the switch method was used. The resulting JNDs were 1.6, 3.8, and 4.0 dB for each of the studies, respectively, showing that the C80 JND may be significantly higher than previously thought. The testing methods, results, and comparison to the previous studies will be described.
Source localization in a reverberant environment using first reflections
Authors: Haapaniemi Eric S. , Femminineo Andrew J. , Williamson Laura M. , Dowling David R.
Published: October 2011

Matched field processing (MFP) has been shown to be effective for remote sound source localization when the receiving array clearly records direct-path sound. Unfortunately, in imperfectly characterized confined environments, echoes and reverberation commonly degrade localization performance. However, inclusion of first reflections in the requisite MFP field model should improve localization performance and such inclusion may be tractable as well. This presentation describes an acoustic technology development effort focused on source localization in a simple confined environment using both direct-path and once-reflected sounds. Experiments were conducted in a 1.0-m-deep and 1.07-m-diameter cylindrical water tank having a reverberation time of ~10 ms using a single sound projector and a linear receiving array of 16 hydrophones. Measured localization performance is reported for impulsive (100 μs) and longer duration sounds having frequencies from 20 to 150 kHz. Appropriate cropping of the signal coda and inclusion of a simplified characterization of the reflection properties of the tank's walls allows successful integration of first reflections that improve MFP localization results. The intended application of this research is localizing sub-visual cavitation bubbles and other hydroacoustic sound sources in hydrodynamic test facilities. [Work sponsored by NAVSEA through the Naval Engineering Education Center.]
On the directivity of musical instruments: Effects on room acoustics calculations
Authors: Sorensen Kristina M. , Eyring Nicholas J. , Leishman Timothy W. , Eyring Nathan G. W.
Published: October 2011

Measurements of musical instruments in an anechoic chamber at Brigham Young University are yielding high-resolution directivity data and balloon plots that may be analyzed and visualized as functions of time or frequency. Historically, room acoustics calculations involving directivity of sound sources have relied on well-defined radiation characteristics including principal radiation axes, directivity factors, beamwidths, etc. In recent times, room simulation software packages have incorporated more comprehensive steady state directivity data for loudspeakers that vary as functions of proportional frequency bands. However, in contrast to loudspeakers, many musical instruments have directivity patterns that are not well defined and that may vary more erratically as functions of pitch or other musical characteristics. This presentation explores the measured directivity properties of a few musical instruments and discusses how they might be encapsulated and incorporated into predictive room acoustics calculations.
Nonlinear coupling and tension effects in a real-time physical model of a banjo
Authors: Bader Rolf , Pfeifle Florian
Published: October 2011

A real-time physical model of the Banjo as proposed by Pfeifle and Bader [J. Acoust. Soci. Am. (2009)] is extended with nonlinear effects on the membrane resulting from the added force of the bridge and the thereby arising anisotropic tension distribution on the membrane. It is shown that this effect directly influences the vibrational behavior of the physical model and yields a more realistic sound. In a second step a non-linear excitation mechanism of the string is presented, modeling the interaction between a standard metal banjo fingerpick and the string of the banjo. The pressure and velocity of the fingerpick are used as control parameters for the model. The whole model is implemented on a FPGA and can be played and controlled in real-time.
Evaluation of hearing protection devices with high-amplitude impulse noise
Authors: Buck Karl , DeMezzo Sebastien , Hamery Pascal
Published: October 2011

Presently, the evaluation of HPDs (hearing protection devices) is mainly based on audiometric threshold methods. However, in the military environment soldiers may be exposed to impulse noise levels of 190 dB and higher if large caliber weapons or IEDs (improvised explosive devices) are considered. In this case it seems reasonable to expect that an evaluation done at hearing threshold might not represent the protection which will be encountered at these highest levels. Therefore, it is necessary to evaluate the HPDs with signals close to the effective exposure. ISL has developed procedures using explosive charges creating very high levels of impulse noise (up to 195 dB peak) for the evaluation of HPDs. ISL also has developed an artificial head that can withstand this type of exposure and has sufficient self-insertion loss for the measurement of double hearing protection. The presentation will give a description of the techniques used at the ISL. It will present some results showing the possibilities to determine the performance of different types of HPDs when subjected to extreme impulse noise levels. Moreover, the new version of the ISL artificial head, compatible ANSI-ASA-S12.42-2010, and possible problems when using it will be presented.
Attenuation performance of active noise reduction headsets using American National Standard s12.42 and the proposed U.S. Environmental Protection Agency Hearing Protector Labeling Regulation
Authors: McKinley Richard L. , Gallagher Hillary L. , Theis Melissa A. , Schley Paul C.
Published: October 2011

The attenuation performance and noise reduction rating (NRR) of six commercially available active noise reduction (ANR) headsets was assessed using the proposed environmental protection agency (EPA) regulation. The passive attenuation results were collected using American National Standard Institute (ANSI) S12.6 method for measuring real-ear attenuation at threshold (REAT) of hearing protectors while the active attenuations results were collected using ANSI S12.42 methods for the measurement of insertion loss of hearing protection devices in continuous or impulsive noise using microphone-in-real-ear (MIRE) or acoustic test fixture procedures. ANSI/ASA S12.68 methods of estimating effective A-weighted sound pressure levels when hearing protectors are worn was used to compute noise reduction metrics including the noise reduction statistic A-weighted (NRSA) and the graphical noise reduction statistic (NRSG). The proposed NRR labels for the ANR headsets were computer per the guidance in the draft U.S. EPA regulation. The presentation will include the baseline passive, active, and total attenuation, the NRSA and the Graphical NRSG, and the proposed EPA labels for passive attenuation and total attenuation while in an active mode.
Calibration details for the impulse peak insertion loss measurement
Authors: Murphy William J. , Vernon Julia A.
Published: October 2011

The American National Standard ANSI S12.42-2010 specifies the measurement of hearing protector performance in the presence of impulse noise. A series of calibration impulses are recorded from an acoustic test fixture (ATF) and a field microphone for peak sound pressure levels of 130, 150, and 170 dB. The averaged acoustic transfer function between the ATF and field microphone is calculated as follows: H¯ATF¯−FF¯(f)=〈FFT¯(PATF,i(t))/FFT(PFF,i(t))〉.¯ The transfer function is computed for each of the ranges of impulse levels and is applied to the field microphone measurements to estimate the unoccluded fixture levels of the ATF when hearing protection is being tested. This method allows a comparison between occluded and unoccluded waveforms. The calibration transfer function is affected by the time-alignment of the field impulse peaks, time-windowing of the impulses, and compensation for any dc bias. Time-alignment significantly affected the accuracy of predicting individual calibration levels with H¯ATF¯−FF¯. The prediction error variance was less at 170 dB than at 130 dB impulses. The time-window was varied from 2.5 to 100 ms preceding the peak of the field impulse. [Portions of this work were supported by the U.S. EPA Interagency Agreement DW75921973-01-0.]
Real-ear attenuation of custom-fit earplugs with the communications earplug (CEP)
Authors: Gordon Elmaree , Reeves Efrem R.
Published: October 2011

Hearing loss is one of the most common occupational injuries in the Department of Defense. One reason is limited access to adequate hearing protection during combat operations. Even if traditional pre-formed foam earplugs are used during combat, approximately 15% of the military population remains unprotected because the pre-formed earplugs do not adequately fit extremely small ear canals, extremely large ear canals, or ear canals with sharp turns. Custom-fit earplugs provide a potential solution for this hard-to-fit population. However, producing traditional wax-dipped custom-fit earplugs in a combat environment is not always practical. Fortunately, methodology exists to scan ear impressions to create a digital data set, transmit this data to a remote manufacturer via the Internet, and fabricate a set of custom-fit earplugs. The purpose of this study was to investigate how the attenuation provided by custom-fit earplugs created using the digital scanning technique compares to the attenuation provided by custom-fit earplugs created using the traditional wax-dipped technique when used with the CEP. Results show the digitally scanned custom-fit earplugs provide significantly poorer attenuation than the traditional wax-dipped custom-fit earplugs. Comply Canal Tip foam earplugs were also evaluated and shown to provide significantly greater attenuation than custom-fit earplugs manufactured by either method.
Distributed point source method and its applications in solving acoustic wave scattering problems
Authors: Kundu Tribikram , Kabiri Rahani Ehsan , Hajzargerbashi Talieh
Published: October 2011

A recently developed semi-analytical technique called distributed point source method is used for solving various wave scattering problems. Scattering of focused ultrasonic fields by air bubbles or cavities in fluid and solid media is investigated in this presentation. Results for both single and multiple cavity geometries are presented. It is investigated when two cavities in close proximity can be distinguished and when it is not possible. The interaction effect between two cavities prohibits simple linear superposition of single cavity solutions to obtain the solution for the two cavities placed in close proximity. Therefore, although some analytical and semi-analytical solutions are available for the single cavity in a focused ultrasonic field, those solutions cannot be simply superimposed for solving the two-cavity problem even in a linear elastic material. The comparison between the ultrasonic energies reflected from two small cavities versus a single big cavity is also investigated. Detection and characterization of cavities is important for both materials science and medical applications because air bubbles in molten metal as well as in blood must be first detected and characterized before taking necessary actions.
Effective medium theory applied to bubble scattering well below resonant frequency
Authors: McElhone Dale I. , Smith Robert W. , Culver R. Lee
Published: October 2011

When ensonifying bubble assemblages using acoustic frequencies well below the resonant frequencies of the individual bubbles, resonant scattering theory for individual bubbles would suggest very low backscatter, whereas an “effective medium theory” predicts a result for backscattering that is orders of magnitude higher. For this case, a number of authors have suggested resonant scattering associated with the size of the bubble clusters, and backscatter measurements from bubble clouds have supported this hypothesis. Scattering can also be considered from a much larger bubble structure: a ship wake. At the stern, ship wakes contain a wide range of bubble sizes. Buoyancy quickly drives the largest bubbles to the surface, but small bubbles can persist for large distances. Ensonification at mid frequencies (1kHz–10 kHz) therefore cannot rely on resonant bubble scattering. Using a generic wake model developed from the literature, acoustic backscatter can be modeled at frequencies below those of the resonant bubbles. The approach discretizes the spatial bubble distribution and sums the backscatter coherently, following the work of Commander and Prosperetti [JASA (1989)]. Ongoing work includes a lab-scale comparison of the backscattering approach applied to the ship wake model. Work sponsored by the Office of Naval Research, Code 321US.
Observations of Biot compressional waves propagating in snow
Authors: Albert Donald G. , Decato Stephen N. , Perron Frank E.
Published: October 2011

Many laboratory measurements of compressional waves propagating in a porous material have been conducted, but similar measurements are less frequent outdoors. Here, we present velocity measurements of Biot compressional waves of the first and second kind propagating through a thick natural snow cover. Vertical sledge hammer blows and blank pistol shots were used to generate horizontally propagating pulses that were recorded by acoustic and seismic sensors. For both sources, seismic P1 compressional, shear, and Rayleigh waves are visible on the measured time series, followed by the airborne acoustic arrival. The acoustic arrival generates additional P1 ice frame waves and Biot P2 pore waves that propagate downward from the snow surface at measured velocities of 460 and 95 m/s, respectively. Theoretical calculations for a rigid porous medium show that the Biot P2 wave can account for the distortion and attenuation observed in the atmospheric acoustic wave. Separate small-scale laboratory measurements demonstrate that the P2 wave in snow is dispersed (the velocity increases with increasing frequency) in agreement with the trend predicted by Biot's theory, but with some parameter adjustments needed to obtain the best agreement. [Work funded by U.S. Army basic research program.]
Numerical examination of the impact of random terrain elevation and impedance variations on sound-field coherence
Authors: Wilson D. Ketih , Parakkal Santosh , Vecherin Sergey N. , Ostashev Vladimir E.
Published: October 2011

Like turbulent fluctuations in the atmosphere, variations in terrain elevations (surface roughness) and ground impedance may be regarded as diminishing the coherence of the sound field and introducing uncertainty into predictions. Here we examine, using a numerical approach, the characteristics and relative importance of terrain-elevation, ground-impedance, and turbulent variations on sound-field coherence. The calculations utilize parabolic equation solutions for a refractive atmosphere, with a Beilis–Tappert transform to incorporate terrain elevation variations and a phase-screen method for the turbulence. The relative importance of terrain variability and turbulence is studied for different refractive conditions. In particular, we examine whether surface roughness has a more significant impact on coherence in downward or upward refracting conditions. Interdependencies between the various mechanisms of coherence degradation are also investigated. The numerical calculations are compared to a recently developed theory for waveguide propagation conditions, which predicts existence of an effective spectrum with independent, additive contributions for turbulence and roughness. Limitations of the Markov approximation for modeling the elevation and impedance variations are also examined.
Angular spectrum representations of focused acoustic beams and a comparison with the complex source point description
Authors: Marston Philip L.
Published: October 2011

One approach to describing the wave fields of focused acoustic beams is to use an angular spectrum representation. This has the advantage of facilitating a simple evaluation of the scattering by a sphere placed at the focal point of the beam in terms of the scattering by a superposition of Bessel beam components. This was recently demonstrated for a quasi-Gaussian beam [Marston, J. Acoust. Soc. Am. 129, 1773-1782 (2011)]. This approach also has the advantage of separating the propagating spectral components from the evanescent spectral components [Marston, J. Acoust. Soc. Am. (submitted)]. While, for most practical acoustic sources, the evanescent spectral components may be neglected, the spectral representation is helpful for explaining why in principle evanescent spectral components can be important in the production of extremely tightly focused beams. A tightly focused beam with a Gaussian profile at the focal plane is not as easily describable using the more widely investigated complex source point description of the focused wave field. In addition to a Gaussian beam, the propagating-component angular spectrum representation is easily applied to the lowest radial-order Laguerre-Gauss helicoidal acoustic beam in the usual case where the beam is not tightly focused. [Work supported by ONR.]
A linearized contrast source method for full-wave modeling of nonlinear acoustic wave fields in media with strong and inhomogeneous attenuation
Authors: Demi Libertario , Verweij Martin D. , van Dongen Koen W. A.
Published: October 2011

The iterative nonlinear contrast source (INCS) method is a numerical method originally developed for modeling transient nonlinear acoustic wave fields in homogeneous (i.e., spatially independent) lossless or lossy media. This method is based on the Westervelt equation and assumes that the nonlinear term describes a distributed contrast source in a linear background medium. Next, the equivalent nonlinear integral equation is iteratively solved by means of a Neumann scheme. In medical diagnostic ultrasound, the observed acoustic nonlinearity is weak, resulting in a weak nonlinear contrast source and a fast convergence of the scheme. To model spatially dependent attenuation, the Westervelt equation has recently been extended with an attenuative contrast source. This presentation deals with the situation of strong local attenuation. In that case, the attenuative contrast source is large and the Neumann scheme may fail to converge. By linearizing the nonlinear contrast source, the integral equation becomes linear and can be solved with more robust schemes, e.g., Bi-CGSTAB. Numerical simulations show that this approach yields convergent results for nonlinear acoustic fields in realistic media with strong local attenuation. Moreover, it is shown that in practical situations, the error due to the linearization remains sufficiently small for the first few harmonics.
A full wave method for modeling nonlinear acoustic wave fields in media with inhomogeneous wave speed, attenuation, and parameter of nonlinearity
Authors: van Dongen Koen W. A. , Demi Libertario , Verweij Martin D.
Published: October 2011

The iterative nonlinear contrast source (INCS) method is a numerical method that has originally been developed for modeling transient nonlinear acoustic wave fields in homogeneous (i.e., spatially independent) lossless or lossy media. Starting from the Westervelt equation, the INCS method considers the nonlinear term of the latter as a nonlinear contrast source in an otherwise linear wave equation and iteratively solves the corresponding nonlinear integral equation using a Neumann scheme. By adding an attenuative contrast source, the method has recently been extended to deal with spatially dependent attenuation, and a spatially dependent parameter of nonlinearity has also been introduced. This presentation is about the introduction of an additional contrast source that account for a spatially dependent wave speed. In this case, convergence problems arise with the Neumann scheme. This problem is solved by replacing the Neumann scheme with a conjugate gradient scheme in which the error functional is based on the complete nonlinear integral equation. Results show that this approach yields very accurate results for the nonlinear wave field (tested up to the fifth harmonic) in media with a spatially dependent wave speed, attenuation, and parameter of nonlinearity, as encountered in realistic medical ultrasound applications.
Use of laryngeal high-speed videoendoscopy systems to study voice production mechanisms in human subjects
Authors: Mehta Daryush D. , Zañartu Matías , Quatieri Thomas F. , Deliyski Dimitar D. , Hillman Robert E.
Published: October 2011

Advances in laryngeal high-speed videoendoscopy (HSV) are making it possible to investigate critical relationships between vocal fold physiology and acoustic voice production in human subjects. Our group has developed HSV systems for clinical research with synchronous acquisition of acoustics, electroglottography, neck skin acceleration, and, in the most comprehensive setup, airflow and intraoral pressure. Key results will be presented from examinations of source-filter coupling and studies of the acoustic impact of vocal fold vibratory asymmetry in subjects with and without voice disorders. Findings hold potential clinical significance by revealing acoustic-HSV relationships not observable using standard stroboscopic imaging, as well as contributing to the direct evaluation and eventual improvement of voice production models. The work of T. F. Quatieri was supported by the Department of Defense under Air Force contract FA8721-05-C-0002. The work of other authors was supported by grants from the NIH National Institute on Deafness and Other Communication Disorders (T32 DC00038 and R01 DC007640) and by the Institute of Laryngology and Voice Restoration.
Voice quality as a correlate of stop coda voicing in developing speech
Authors: Hanson Helen M. , Shattuck-Hufnagel Stefanie
Published: October 2011

Previous work reported that two children from the Imbrie Corpus (2005) tend to produce noise at the end of the nuclear vowel in words with [-voice] stop codas (duck, cup), but not in words with [+voice] codas (tub, bug) [JASA 123, 3320]. This noise resembles the preaspiration reported for adult speech in some languages. In this paper the earlier work is extended to the remaining eight children in this corpus of speech from 2- to 3-year old children learning American English. Furthermore, for nine subjects we compare their behavior to that of their primary female caregiver. Many of the children produce unvoiced coda stops that are coded as being strongly preaspirated. Their caregivers also produce vowel-final noise preceding [-voice] stops, but these tend to be coded as being weakly preaspirated. For both child and adult subjects preaspiration is more likely to occur for /k/ than for /p/. A few subjects produce irregular voicing rather than noise before [-voice] coda stops. More detailed acoustic measures such as durations and relative intensities will be presented. In addition, the acoustics are used to infer the physiology behind these voice quality differences between children and adults. [Work support from NIH R01DC00075, R01HD057606.]
Simulation of the coupling between vocal-fold vibration and time-varying vocal tract
Authors: Tanabe Yosuke , Mokhtari Parham , Takemoto Hironori , Kitamura Tatsuya
Published: October 2011

Acoustic coupling between the voiced sound source and the time-varying acoustic load during phonation was simulated by combining the vocal-fold model of [S. Adachi etal., J. Acoust. Soc. Am 117(5) (2005)] with the vocal-tract model of [P. Mokhtari etal., Speech Commun. 50, 179–190 (2008)]. The combined simulation model enables to analyze the dynamic behavior of the vocal folds due to the time-varying shape of the vocal tract. An example of vocal-fold behavior is shown for sustained phonation of the Japanese vowel sequence “aiueo.” [This research was partly supported by Kakenhi (Grant Nos. 21500184, 21300071).]
Speaker adaptation in infancy: The role of lexical knowledge
Authors: van Heugten Marieke , Johnson Elizabeth K.
Published: October 2011

The acoustic realization of words varies greatly between speakers. While adults easily adapt to speaker idiosyncrasies, infants do not possess mature signal-to-word mapping abilities. As a result, the variability in the speech signal has been claimed to impede their word recognition. This study examines whether exposure to a speaker may allow infants to better accommodate that speaker's accent. Using the Headturn Preference Procedure, 15-month-olds were presented with lists containing either familiar (e.g., ball) or unfamiliar words (e.g., bog). In experiment 1, these words were produced in infants' own accent (Canadian English); in experiment 2, they were produced in a foreign accent (Australian English). Comparable to previous work (Best etal., 2009), only infants presented with their own accent preferred to listen to familiar over unfamiliar words. Thus, without access to speaker characteristics, word recognition is limited to familiar accents. In experiment 3, the same Australian-accented stimuli were preceded by exposure to the Australian speaker. Speaker adaptation tended to correlate with the infants' vocabulary size, with greater vocabularies being indicative to more robust adaptation. We are currently testing whether vocabulary size is a mediating factor caused by general processing abilities or whether speaker adaptation is lexically driven.
Social accountability influences phonetic alignment
Authors: Mitterer Holger
Published: October 2011

Speakers tend to take over the articulatory habits of their interlocutors [e.g., Pardo, JASA (2006)]. This phonetic alignment could be the consequence of either a social mechanism or a direct and automatic link between speech perception and production. The latter assumes that social variables should have little influence on phonetic alignment. To test that participants were engaged in a “cloze task” (i.e., Stimulus: “In fantasy movies, silver bullets are used to kill ...” Response: “werewolves”) with either one or four interlocutors. Given findings with the Asch-conformity paradigm in social psychology, multiple consistent speakers should exert a stronger force on the participant to align. To control the speech style of the interlocutors, their questions and answers were pre-recorded in either a formal or a casual speech style. The stimuli's speech style was then manipulated between participants and was consistent throughout the experiment for a given participant. Surprisingly, participants aligned less with the speech style if there were multiple interlocutors. This may reflect a “diffusion of responsibility:” Participants may find it more important to align when they interact with only one person than with a larger group.
The influence of socioindexical expectations on speech perception in noise
Authors: McGowan Kevin B.
Published: October 2011

Previous research has shown that listener percepts can be altered by the manipulation of listener expectations of speaker identity [e.g., Niedzielski (1999), Hay etal. (2006)]. This study examines the extent to which socioindexical information can be useful to listeners. English speakers with little or no Chinese experience and Heritage Mandarin English speakers transcribed Chinese-accented speech in noise. In a between-subjects design, listener expectations about speaker identity were manipulated by presenting an Asian face, a socioindexically uninformative silhouette, or a Caucasian face as the purported speaker of high and low predictability sentences. Consistent with previous findings, listeners presented with an Asian face were significantly more accurate transcribers than those presented with a Caucasian face. Intriguingly, the silhouette condition patterned with the Asian face for experienced listeners but with the Caucasian face for inexperienced listeners. Unexpectedly, inexperienced listeners, while overall less accurate than experienced, Heritage listeners, saw the same magnitude improvement with the Asian face. Furthermore, transcription errors are inconsistent with suggestions (e.g., Staum Casasanto, 2009; Johnson, 2006) that listeners will alter base activations of prelexical or lexical forms to accommodate an expected accent. Implications for theories of speech perception and word recognition will be explored.
Reading between the turns: Social perceptions of turn-taking cues in conversation
Authors: Tice Marisa , Henetz Tania
Published: October 2011

The perception of turn-taking cues provides an alluring new avenue for speech perception research since it requires complex cognitive and social processes that speakers must perform online during conversation. This study investigates whether inter-speaker gap duration, overlap, and backchanneling play a role in the social perception of speech during conversation. In a matched-guise task, participants heard clips of spontaneous, dyadic conversation. Clips were phonetically manipulated to control for the three turn-taking cues of interest by stretching and shrinking gap durations and replacing the naturally occurring backchanneling with silence. Participants then rated the speakers and conversations on a range of social measures, including dominance, formality, and similarity. Initial results indicate that listeners' social judgments are sensitive to these cues. For example, the removal of backchanneling in a male–male conversation resulted in ratings of higher speaker interest and lower speaker similarity, running contrary to some previous descriptions of backchannels as supportive signals (Yngve, 1970). This effect may interact with gender, such that backchanneling is interpreted differently in female-female or mixed-gender dyads, supporting Pearson etal.'s (2008) finding that turn-timing effects interact with (mis)matches in speaker race. It is clear these judgments follow from a complex interaction of subtle acoustic cues, social information, and context. [This work was supported by a National Science Foundation Graduate Research Fellowship to M. Tice.]
The interrelation between perceived and produced vowel spaces in Brazilian and European Portuguese
Authors: Chladkova Katerina , Escudero Paola
Published: October 2011

Previous research has shown that the interrelation between the perception and production of vowels is not straightforward. For instance, it has been shown that the vowel spaces of Spanish and Czech may have different structures in production than in perception (Boersma and Chladkova, 2011). This study compares vowel perception and production of 40 speakers in Brazilian- (BP) and European Portuguese (EP). Productions were elicited in a sentence-reading task and were reported in Escudero etal. (2009). Perception was tested by means of a phoneme identification task with tokens sampled from the whole vowel space. Perceived and produced vowel spaces were compared through the location of vowel categories (the vowels' F1 and F2 averaged across speakers/listeners) and their boundaries with neighboring categories. The results show that the perceived vowel space of BP listeners differs from that of EP listeners. Specifically, low-mid vowels have a higher F1 in BP than in EP, a finding that is in line with BP and EP vowel production differences. However, BP low-mid vowels are less peripheral than EP vowels along the F2 dimension only in perception. Other dialectal differences as well as the degree of correspondence between the production and perception of Portuguese vowels are discussed.
Hearing vocalizer race from voiced laughter sounds
Authors: Stenson Anais F. , Martin Noel B. , Amoss R. Toby , Owren Michael J.
Published: October 2011

Previous experiments testing whether an individual's race can be identified from speech sounds have produced inconsistent results. Current work examined this question using spontaneous laughter sounds to help isolate voice quality from dialect effects. In Experiment 1, six laughs from each of six black and six white vocalizers were edited to create 72 bouts of three to four contiguous bursts. These bouts were then further edited to create 72 single bursts. Fourteen undergraduate participants heard both sets of laughs in separate blocks. Sounds were presented twice on each trial and categorized as “Black” or “White.” Mean percentage-correct was 61.1 (SD = 10.1) with multi-burst versions, and 55.8 (SD = 12.7) with single bursts. Both outcomes were statistically higher than chance performance in t-test comparisons. Experiment 2 tested 12 new listeners in a balanced, same-different design with paired, bout-length stimuli from same- or different-race vocalizers. There were 120 stimuli in all, combining 5 laughs from each of 5 laughers of each race. Mean d' was 0.92 (SD = 0.44), which was statistically different from chance. Taken together, these experiments suggest modest, but reliable sensitivity to vocalizer race from voice-quality alone.
Fundamental frequency and perceptual adaptation to gender and emotion
Authors: Hubbard Daniel J. , Assmann Peter F.
Published: October 2011

Previous work has documented perceptual adaptation to nonlinguistic properties in speech using voice gender and emotion categories. Exposure to voice gender and expressive speech adaptors produced a response shift away from the adaptation category. This project extends those findings by examining the contribution of fundamental frequency (F0) to perceptual speech aftereffects following adaptation to VCV syllables. Voice recordings from six speakers were processed using the STRAIGHT vocoder and an auditory morphing technique to synthesize gender (experiment 1) and expressive (experiment 2) speech sound continua ranging from one category endpoint to the other (female to male; angry to happy). Continuum endpoints served as adaptors for F0 present and F0 removed conditions. F0 removed stimuli were created by replacing the periodic excitation source with broadband noise. Aftereffects were found only in the F0 present condition, resulting in a decreased likelihood to identify test stimuli as part of the adaptation category. Aftereffects did not appear when F0 was removed. The findings highlight the important role F0 plays in perceptual adaptation to gender and expressive properties of speech, and further identifies a common acoustic basis for speech adaptation among several different classes of stimuli.
Automated rhythmic discrimination of dysarthria types
Authors: Utianski Rene L. , Berisha Visar , Liss Julie M. , Lansford Kaitlin
Published: October 2011

This study examines whether features of rhythm, extracted via an automated program, can match the success of hand-extracted metrics in discriminating among control and different types of dysarthric speech. Previously, Liss etal. [J. Speech, Language, and Hearing Res., 52(5), 1334–1352 (2009)] demonstrated that rhythm metrics could successfully separate individuals with perceptually distinct rhythm patterns, with the majority of classification functions more than 80% successful in classifying dysarthria type. However, the hand extracted rhythm measurements are labor intensive and require expertise in acoustic analysis to achieve valid and reliable measures. In this study, digitized speech was automatically segmented into vocalic and voiceless intervals using an autocorrelation-based algorithm, and, from these intervals, rhythm metrics were computed. Discriminant function analyses were performed and revealed the majority of functions were more than 90% successful in classification, matching, or exceeding the success of the hand- extracted measurements. Removing subjectivity, automating this procedure, and verifying its comparability with traditional methods improve the viability of rhythm metrics as a clinical tool in speech therapy intervention. Additionally, the ease of manipulation of the automated program will allow for development of metrics that capture severity and individual speaker differences that will inform models of speech perception.
Movements of the tongue, lips, and jaw for selected vocalic nuclei in speakers with dysarthria
Authors: Weismer Gary
Published: October 2011

Speech movement data in persons with dysarthria are relatively rare, and much of this sparse data is restricted to the lips and jaw. The purpose of this presentation is to provide a survey of speech movements for selected vocalic nuclei, in speakers with dysarthria due to Parkinson's disease and Amyotrophic Lateral Sclerosis, and to compare these movements to those of healthy controls. The speech materials used in this analysis are those collected over a ten-year period at the x-ray microbeam facility at the University of Wisconsin-Madison. The selected vocalic nuclei are extracted from connected speech samples, including both read sentences and an extended reading passage; multiple repetitions of sentence data are available to estimate the variability of the selected movements. Both qualitative and quantitative analyses of these speech movements, with emphasis on lingual motions, will be employed to generate hypotheses concerning the speech movement deficit in the dysarthrias. This inductive analysis approach will pay special attention to across-speaker variability, among both healthy controls and speakers with dysarthria. A working hypothesis is that the lingual motions for speakers with dysarthria are no more variable across repetitions than those in healthy controls, but show deficits in scale and form.
A continuing study of the temporal structure of the speech of a person with dementia
Authors: Carozza Linda , Quinn Margaret , Nack Julia , Bell-Berti Fredericka
Published: October 2011

The question of motor speech degeneration in the course of dementing illness is a relatively unexplored area; rather, extensive research has focused on cognitive and language processes in dementia. The potential for early dissociation of motor functions of language at the level of speech production has not been explored; an interaction between motor speech and language production and perception changes should inform our understanding of the deterioration in dementia. We have previously reported results of a preliminary study of the temporal structure in the speech of persons with dementia. In a continuing longitudinal study of one of our subjects, we found that after 6 months, utterance-level timing patterns were more disrupted than segment- and syllable-level patterns in his speech. This report explores changes over time in the temporal structure of speech of that participant, who was recorded 12 months after the second recording session. The results of this third recording session, in which we will examine utterance-level timing patterns (phrase-final lengthening and compensatory shortening) and segment- and syllable-level timing patterns (VOT and the effects of final consonant voicing on vowel duration) will be presented.
A principal components analysis of tongue motion patterns in normal and glossectomy speakers
Authors: Langguth Julie , Woo Jonghye , Stone Maureen , Chen Hegang , Prince Jerry L.
Published: October 2011

This study determines whether a small excision of the tongue, due to removal of a lateral T1 tumor, will cause a change in tongue motion patterns during speech. Principal components analysis (PCA) compared the motion patterns of ten control subjects to those of three glossectomy patients. The patients were 7–39 months post-glossectomy removal of T1N0M0 tumors and had primary closures. Cine- and tagged- magnetic resonance imaging (MRI) data included three sagittal slices, which were used to compare the midsagittal and bi-lateral motions of the tongue during speech. The data were internal tissue point motions extracted from the tagged MRI images using harmonic phase MRI (HARP-MRI) (NessAiver and Prince, 2003). The motion patterns of the tongue (velocity field) were observed during the motion from “ee” to “s” which may be difficult for glossectomy patients as it requires subtle changes in the tongue tip. The results demonstrated normal variability across the control subjects. The three patients retained some components of normal tongue motion; however, there were also notable differences from the controls. The glossectomy patients plotted at the extremes of the PC loadings, and the details revealed how they differed from the controls, from each other, and used individual compensatory strategies. [Work supported by NIH-R01CA133015.]
Acoustic characteristics of diphthongs in Parkinson's disease
Authors: Tjaden Kris
Published: October 2011

The second formant frequency (F2) is sensitive to dysarthria. For example, vowel F2 slope has been shown to correlate with intelligibility, such that persons who are less intelligible or who have more severe speech impairment also tend to have shallower F2 slopes (Weismer etal., 1992). This result is consistent with studies suggesting the perceptual importance of transitions (Hillenbrand and Nearey, 1999), although the extent to which F2 slope explains reduced intelligibility in dysarthria is unclear. F2 slope also may serve as an acoustic metric of effort (Moon and Lindblom, 1994; Wouters and Macon, 2002). Therapeutic techniques for dysarthria, including Clear and Loud speech, are thought to be beneficial because they involve a scaling up of effort. Clear speech emphasizes hyperarticulation or increased articulatory effort, while Loud speech focuses on increasing respiratory–phonatory effort, although it has been suggested that increased loudness involves a system-wide scaling up of effort (Fox etal., 2006). Empirical evidence that Clear or Loud speech is associated with increased articulatory effort in dysarthria is limited. The current study sought to further evaluate this suggestion by investigating F2 slope characteristics for vowels produced in Habitual, Clear and Loud conditions by individuals with dysarthria secondary to Parkinson's disease.
Music masking speech in cochlear implant simulations
Authors: Hossain Shaikat , Assmann Peter F.
Published: October 2011

While the question of how speech is segregated among competing talkers has received quite a lot of attention in the cochlear implant (CI) literature, an equally important question has been largely neglected: To what extent does music mask speech? Recent studies have shown that bimodal stimulation, using a hearing aid to boost low frequency acoustic information in the ear with residual hearing, can markedly improve CI users' ability to separate multiple talkers, recognize speech in noisy conditions and discriminate between melodies. In this study, normal hearing listeners completed a speech recognition task in musical maskers under two different processing conditions: bimodal and normal CI simulations. Results showed that listeners performed significantly better under the bimodal simulation in all three SNR conditions (0, -3, -6 dB). It is thought that the low frequency acoustic information provided listeners with a release from masking as compared to the normal CI condition, particularly at low SNRs. The amount of masking release varied as a function of instrument type and SNR. The findings of this study may be especially relevant for the perceptual separation of instrumentation and lyrics in popular music which remains to be a highly challenging task for CI users.