bluuebunny/crossref_metadata_2025 huggingface dataset:
Miniature kilohertz-frequency ultrasound implant for brain drug delivery.
Authors: Foo John C. , Pannullo Susan C. , Olbricht William L. , Shuler Michael L. , Lewis George K.
Published: July 2011

Glioblastoma multiforme is a highly aggressive brain malignancy currently treated by surgical resection, radiotherapy, and local drug delivery. However, due to heterogeneities in neural tissue and tumor cell migration into surrounding healthy tissue, treatment strategies remain ineffective and patient prognosis poor. We have shown that low intensity focused and nonfocused ultrasound may safely enhance the penetration and distribution of chemotherapeutics delivered locally via convection-enhanced infusions in rodent brain parenchyma in vivo. In scaling the ultrasound-assisted therapy from rodent to large animal and increasing its clinical translatability, we designed a miniature kilohertz (100 and 600 kHz) ultrasound implant that can be placed within the brain tumor resection cavity during patient recovery to assist chemotherapy. The miniature ultrasound system has been tested in a diffusion-based human-sized agarose brain phantom tumor model under various acoustic intensities and pulse sequences. Results indicate that ultrasound increases the volume of tracer distribution by a factor of 2–3 compared with controls. We present the effects of acoustic intensity and pulse interval from the implant on distribution of tracers in vitro and ex vivo in the phantom model and intact equine brain, respectively. The miniature ultrasound implant provides significant drug distribution gains that may improve patient outcome.
Piezoelectric poly-g-benzyl-L-glutamate polymethyl methacrylate composite disks fabricated using molds.
Authors: Hwang Yonghwan , Je Yub , West James E. , Yu Seungju M. , Moon Wonkyu
Published: July 2011

Poly-g-benzyl-L-glutamety Polymethyl methacrylate (PBLG-PMMA) composite polymer, a new piezoelectric transduction material, was invented by Yu et al. In this study a method is developed to fabricate a disk using detachable molding. Unlike film-type piezoelectric polymers such as the polyvinylidenefluoride (PVDF), this new material can be used to form a body of various shapes and sizes such as conventional piezoelectric ceramics. The piezoelectric characteristics mainly come from the state of alignment of polygamma-benzyl alpha, L-glutamate molecules (PBLG), which have large electric dipole moment. The PBLG can be melted with methyl methacrylate by heating, and mixture with applying electric field can be polymerized as a piezoelectric PBLG-PMMA composite in the mold. The fabricated specimens have the dimensions of 20 mm in diameter and 3 mm in height. The piezoelectric coefficient (d33) depends on the concentration of PBLG. Measured values were 0.45 and 1.9 pC/N with concentrations of 10 and 30 wt %, respectively. Even though these values are lower than PVDF, PBLG-PMMA could be applied as a transduction material in various shapes of acoustic transducers because the results suggest that increasing the intensity of electric field and the PBLG concentration may improve the piezoelectricity. Moreover, it is also a great potential to consider new matrix materials instead of PMMA. [Work supported by Grant No. ADD-UD080004DD.]
Capacitive micromachined ultrasound Doppler velocity sensor.
Authors: Shin Minchul , Krause Joshua S. , DeBitetto Paul A. , White Robert D.
Published: July 2011

An acoustic Doppler velocity measurement system using microelectromechanical systems (MEMS) capacitive micromachined ultrasound transducer (cMUT) array technology is introduced. This low power acoustic velocity measurement system operates in both transmit and receive modes. The device consists of 64 0.6 mm diameter polysilicon and gold diaphragms, and operates at approximately 180 kHz with a quality factor of 30. Computational predictions suggest that in transmit mode the system will deliver a sound pressure level (SPL) of approximately 66 dB SPL at 1 m from the source with an 11 deg, −3 dB beamwidth when driven with a 12 V excitation. In receive mode, the predicted sensitivity is 2.2 mV/Pa at the preamplifier output. Based on experimentally determined electronic noise densities of approximately 4×10−16 V2/Hz, resolution may be as good as 0 dB SPL in a 5 Hz band. This suggests that with good reflection, a range of approximately 8 m (4 m out and 4 m back) is achievable. Velocity resolution is expected to be on the order of 1 cm/s with a 5 m/s maximum measurable velocity. At this point, all predictions are computational. Fabrication will be complete in the next month and experimental characterization of system performance will be presented at the meeting.
A homemade Edison tinfoil phonograph.
Authors: McNeese Andrew R. , Lenhart Richard D. , Sagers Jason D. , Wilson Preston S.
Published: July 2011

In 1877 Thomas Edison invented the phonograph, a device capable of recording and reproducing sound. The original design used the sound induced vibrations of a stylus to etch a time-locked copy of the acoustic wave onto a rotating, foil-covered cylinder. Playback was made possible by reading the etched signal with a second stylus attached to a transmitting diaphragm. The transparency of Edison’s original design makes the phonograph a useful tool to demonstrate and discuss the concepts of acoustic waves and sound-structure interaction. A short history of the invention is given, a homemade version of Edison’s original phonograph is presented, and the essence of the sound-structure interaction is explained.
Perception of high-frequency energy in singing and speech.
Authors: Monson Brian B. , Lotto Andrew J. , Story Brad H.
Published: July 2011

Traditionally, acoustical energy above 5000 Hz in voice and speech has received little attention from the scientific community. Some recent studies have explored the role of this high-frequency energy (HFE) in qualitative percepts of speech and music. To determine what other perceptual information is extractable from this frequency range, listeners were presented with tokens containing only HFE from samples of singing and speech (i.e., singing and speech high-pass filtered at approximately 5.7 kHz). Results showed that listeners could successfully discriminate gender and mode of production (speech versus singing) given only acoustical energy above 5 kHz. Additionally, listeners successfully identified the song being sung and the words being spoken. The implication is that HFE potentially plays a more significant role in perception of singing and speech than that suggested by the current understanding. [Work supported by the NIH-NIDCD.]
Harmonic-temporal model with multiple function for sound synthesis.
Authors: Nakashika Toru , Takiguchi Tetsuya , Ariki Yasuo
Published: July 2011

The most popular method for synthesizing musical timbre sound is additive synthesis, where the timbre sound is obtained by adding multiple harmonic partials with different time envelopes. Additive synthesis has many advantages such as high flexibility to control each partial individually and easy way to reconstruct musical tones from the spectrum. However, it is difficult to control the harmonic partials as a whole because time envelopes of each harmonic are independent of one another. Furthermore, the model has many parameters which are difficult to control. For example, many instruments require dozens of harmonics, where each envelope consists of about 1000 time points, and the parameters are hard to handle. In this paper, additive synthesis is extended to easily control a whole sound with only a few parameters. In this approach, all the time envelopes of each harmonic are parametrically modeled by a 2-D function, called multiple function. Two types of multiple functions are introduced with the update rules of their parameters based on an EM algorithm. The harmonic-temporal shape of an instrument sound can be directly controlled by changing the parameters of the proposed function slightly.
Active control of centrifugal fan noise: Modeling design guidelines.
Authors: Esplin J. James , Boyle John K. , Sommerfeldt Scott D. , Gee Kent L.
Published: July 2011

Information technology (IT) noise is very prevalent in today’s society. Active noise control (ANC) has shown promise in minimizing the effect of fan-induced IT noise on users. Much of the previous research has concentrated on axial cooling fans, such as those found in desktops and servers. This approach was based on the concept of minimizing radiated acoustic power in a model of the fan radiation, and using those results to determine appropriate nearfield locations for the error sensor(s). This paper describes modifications to this previous method to develop a modeling approach to implement active noise control with a centrifugal blower, such as those found in fan trays and laptop computers. This model has been used to predict tonal noise inside and outside the duct, as well as how to best develop an ANC system for such an idealized setup. Differences between the axial fan model and the centrifugal blower model are discussed, as well as some limiting assumptions for each model.
Bayesian parameter estimation for porous material characterization.
Authors: Fackler Cameron J. , Xiang Ning
Published: July 2011

This work proposes a method to estimate the material parameters of rigid-frame porous materials through measurement of the acoustic impedance of such materials. Porous materials are characterized by their physical parameters: porosity, tortuosity, and flow resistivity. For many materials, direct measurement of these parameters requires time-consuming or highly sensitive procedures. Based on some existing models for the characteristic impedance of a porous material in terms of the physical material parameters, Bayesian parameter estimation is used to estimate the physical parameters of a material from a measurement of its complex acoustic impedance. In addition to estimation of the values of the physical parameters, Bayesian analysis generates information on the uncertainties and interdependence of the parameters. The results obtained are compared to published data for several rigid-frame porous materials.
Acoustic parameters of sonoluminescence in alcohol aqueous solutions.
Authors: Chen Weizhong , Cui Weicheng , Qi Suibao
Published: July 2011

The single bubble sonoluminescence (SBSL) has been investigated in alcohol aqueous solutions with different concentrations. The amplitude and frequency of the driving ultrasound for SBSL change with the alcohol concentration. That is, the best driving amplitude and frequency decrease as the concentration increases. A drop of alcohol can weaken, even quench the SBSL in pure water if we drive the bubble under the best condition of the pure water. On the other hand, if the driving ultrasound is optimized for the SBSL in alcohol aqueous solution, the decrease of the alcohol concentration will weaken the SBSL. In other words, the SBSL in pure water driven by the best ultrasound of aclohol aqueous solution will brighten as the alcohol is dissolved in. The parametric region for SBSL in pure water and alcohol aqueous solution is experimentally measured. Their difference can be explained in the theoretical framework of Rayleigh bubble dynamics.
Relative loudness of speech feedback during speech production.
Authors: Barac-Cikoja Dragana , Karch Stephanie , Kokx Melissa , Lancaster Lucas
Published: July 2011

Relative loudness of speech feedback during speech production was investigated using an adaptive, two-track, two interval forced choice procedure. The participants indicated which of the two signals, the speech feedback (listening while speaking interval) or its audio recording (listening only interval), sounded louder. Based on the subject’s choice, the gain [indeabel sound pressure level (SPL) on the recording was raised or lowered on the subsequent trial. Relative loudness of the speech feedback was defined as the difference in the sound pressure level of each signal (feedback and replay), when the two were experienced as equally loud. The subject’s speech was recorded with a microphone placed above his/her right ear, and both speech signals were presented via insert earphones. In the canal speech recordings were collected for later analysis. Fifteen normally hearing adults were tested. At the point of subjective equality, the SPL of the recorded and the live signals were equal for some of the subjects, while for others, the SPL of the live speech exceeded the SPL of the recorded speech by as much as 4 dB. The subjects’ preferred speaking level and their acoustic reflex thresholds did not account for the observed individual differences. [Work supported by RERC-HE to Gallaudet University.]
Perceptual mechanism for spectral-ripple discrimination in cochlear implant listeners: Behavioral and model data.
Authors: Won Jong Ho , Jones Gary L. , Drennan Ward R. , Jameyson Elyse M. , Rubinstein Jay T.
Published: July 2011

The present study investigated the perceptual mechanism for spectral-ripple discrimination in cochlear implant listeners. The main goal of this study was to determine whether cochlear implant listeners use a local intensity cue or global spectral shape for spectral-ripple discrimination. The effect of channel interaction on spectral-ripple discrimination was also evaluated. Results showed that it is unlikely that cochlear implant listeners depend on a local intensity cue for robust spectral-ripple discrimination. A phenomenological model of spectral-ripple discrimination, as an ideal observer, showed that a perceptual mechanism based on discrimination of a single intensity difference cannot account for performance of cochlear implant listeners. Instead, there was a significant dependence of thresholds on spectral modulation depth and channel interaction. The evidence supports the hypothesis that spectral-ripple discrimination involves integrating information from multiple channels. [Work supported by NIH Grant Nos. R01-DC007525, P30-DC04661, F31-DC009755, F32-DC011431, and T32-DC000033 and an educational fellowship from Advanced Bionics Corporation.]
Cochlear measures of selective auditory attention.
Authors: Walsh Kyle P. , Pasanen Edward G. , McFadden Dennis
Published: July 2011

The possibility that selective auditory attention can affect the responses of the cochlea via the medial olivocochlear (MOC) pathway was investigated in human listeners using a nonlinear version of the stimulus-frequency otoacoustic emission (SFOAE), called the nSFOAE [Walsh et al. (2010)]. During nSFOAE recording, listeners attended to one of two competing speech streams, each composed of seven randomly-selected spoken digits that were interleaved with the nSFOAE stimuli. The talker was female in one ear and male in the other (randomized across trials). The task of the listener was to match a subset of the digits spoken by the female talker to one of two choices presented visually on a computer screen. The nSFOAE stimulus was a tone (4.0 kHz, 200 ms) presented simultaneously to the two ears, either in quiet (tone-alone) or in noise (tone-plus-noise). When nSFOAEs were measured during periods of selective attention, the tone-plus-noise responses exhibited larger maximum magnitudes, and larger changes from the tone-alone magnitudes, compared to the responses measured during a no-attention condition, in which the competing speech streams were presented, but with no digit-matching required. The differences in magnitudes across conditions were as much as 4–5 dB. [Work supported by the NIDCD.]
Automated detection of alarm sounds.
Authors: Lutfi Robert A. , Heo Inseok
Published: July 2011

Alarm sounds are an important class of sounds with distinctive, easily recognized features that serve as a call to action. These properties make the automated detection of alarm sounds a model problem for computational auditory scene analysis [Ellis, D. (2001). CRAC Workshop, Aalborg, DK, pp. 59–62]. We compare two approaches to alarm detection, one based on a change in overall sound level (rms) and the other a change in the power-normalized autocorrelation function (PNA). ROC curves in each case were obtained for different exemplars of four classes of alarm sounds (bells/chimes, buzzers/beepers, horns/whistles, and sirens) embedded in four noise backgrounds (cafeteria, park, traffic, and music). Detection was based on 1 s time-segments drawn at random from longer recordings of these sounds. The recordings were equated in average power. Both detection algorithms produced asymmetric ROCs with similar hit rates of 80% or more for false-alarm rates above 25%. PNA gave significantly higher hit rates than rms at false alarms rates below 25%, but did not perform as well for sirens or in the background of music. The results are considered in terms of the role periodicity may play in human detection of natural targets in natural noise backgrounds.
Developing acoustic measures to evaluate the emergence of phonological contrast.
Authors: Beckman Mary E. , Edwards Jan , Munson Benjamin
Published: July 2011

Acoustic analysis has been used in studying phonological development since at least the 1960s when Preston and colleagues measured voice onset time (VOT) in English- and Arabic-learning toddlers. Studies applying measures such as VOT, for stop voicing, and spectral peak frequency, for fricative place, show that the emergence of robust phonemic contrast is more gradual than it seems when phonetic transcription is used alone. For example, Japanese-speaking children’s short-lag productions of target /d/ labeled as “[t]” might show somewhat shorter VOT than their productions of target /t/. Studies that supplement acoustic measures with continuous labeling tasks, such as a visual analog scale with “d” and “t” as target endpoints, support the idea that not only is this developmental phenomenon of “covert contrast” ubiquitous, the associated subphonemic variation can be perceived by adult listeners given the appropriate task. Moreover, listeners with different kinds of experience with young children have different sensitivities to this acoustic variation. This finding suggests that appropriate acoustic measures can be developed to provide psychometrically valid measures over the full course of development from initial stages of producing undifferentiated or inappropriate cues to producing the adult community pattern for the contrast. [Work supported by NIDCD grant 02932 and NSF grants BSC-0729306, BSC-0729140, and BCS-0729277.]
Tracking in ocean acoustics for geoacoustic inversion.
Authors: Michalopoulou Zoi-Heleni
Published: July 2011

Tracking acoustic features, namely, multipath arrivals and modal frequencies arriving at a specific time, is performed with the goal of improving geoacoustic inversion in ocean environments. Sequential Bayesian filtering results obtained with particle filters are compared to conventional maximum likelihood estimates in terms of the accuracy of the environmental and geometry parameter estimates that they produce. Previous sequential tracking methods are improved to provide higher resolution in estimation. Probability density functions are calculated for uncertainty characterization of both multipath arrival times or modal frequencies (depending on the circumstances of the measurements) and environmental parameter estimates. [Work supported by ONR.]
Sequential trans-dimensional Monte Carlo methods for range-dependent geoacoustic inversion.
Authors: Dettmer Jan , Dosso Stan E. , Holland Charles W.
Published: July 2011

This paper develops a sequential trans-dimensional Monte Carlo algorithm for geoacoustic inversion in a strongly range-dependent environment. The particle-filter algorithm applies advanced Markov chain Monte Carlo methods in combination with importance sampling techniques to carry out geoacoustic inversions for consecutive data sets acquired along a track. Environmental changes along the track, such as changes to the number of sediment layers, are accounted for with trans-dimensional partition modeling which intrinsically determines the amount of structure supported by the data information content. Abrupt environmental change between consecutive data sets of high information content (peaked likelihood) are addressed by bridging distributions implemented using annealed importance sampling. Such distributions provide an efficient method to locate high-likelihood regions for new data which are distant and/or disjoint from previous high-likelihood regions. The algorithm is applied to reflection-coefficient data along a track, as can be collected using towed arrays. A simulated environment varies rapidly along the track, with changes in the number of layers, layer thicknesses, and geoacoustic parameters within layers. In addition, the seabed contains a geologic fault where all layers are offset abruptly and an erosional channel. Abrupt changes in noise level between consecutive data sets are also considered. [Work supported by ONR.]
Bayesian channel estimation for K-distributed fading models.
Authors: Laferriere Alison B. , Preisig James C.
Published: July 2011

Current underwater acoustic channel estimation techniques generally apply linear MMSE estimation. This approach is optimal in a mean square error sense under the assumption that the impulse response fluctuations are well characterized by Gaussian statistics, leading to a Rayleigh distributed envelope. However, field data suggest that the envelope statistics of the channel are better modeled by the K-distribution. MAP and MMSE estimators for a scalar channel tap have been derived assuming K-distributed fading models. The MAP estimator for a vector of K-distributed channel taps has also been derived. The implementation of these estimators on simulated data demonstrates an improvement in performance over linear MMSE estimation. [Work supported by the ONR Grant No. N00014-05-10085 and NSF Grant No. OCE-0519903.]
A multiscale sparse processing approach to tracking the delay-Doppler spread in shallow water acoustic communications.
Authors: Sen Gupta Ananya , Preisig James
Published: July 2011

Sparse sensing approaches have been applied to track the delay-Doppler spread in shallow water acoustic communications with varying degrees of success. Tracking the complex-valued delay-Doppler spread function coefficients is challenging due to the ill-conditioned, nonanalytic, and time-varying nature of the channel estimation problem. Since the coefficients typically follow a sparse distribution, fast adaptive least squared approaches fall short of the performance goals. We recently proposed an adaptive sparseoptimization technique, which efficiently tracks the time-varying coefficients by imposing a smooth nonconvex cost function over the L1 norm of the sparse coefficients and the L2 norm of the estimation error. However, a key limitation of this approach is the assumption that the sparsity level of the distribution stays fixed and is uniform across the different Doppler frequencies. Due to uncertainty principles governing time and frequency resolution of nonstationary processes, using uniform averaging window length across the entire Doppler spread also leads to nonuniform resolution of the estimates across the different frequency ranges. We present a multiscale algorithm where we employ variable averaging window length as well as variable sparsity factor to track the delay-Doppler spread across different ranges of frequencies over experimental field data.
Striation-based beamforming using the waveguide invariant for passive sonar.
Authors: Rouseff Daniel , Zurk Lisa M.
Published: July 2011

Consider a horizontal array measuring the coherent field produced by a distant acoustic source. If the resulting acoustic intensity were mapped as a function of position along the array and frequency, the image would exhibit striations, bands of high intensity. It is well known that the slope of the observed striations can be related to bearing of the source and the ratio between the so-called waveguide invariant and the range to the source. The canonical value of the waveguide invariant in shallow water is 1, but deviations by 30% or more are not uncommon when there is a sharp thermocline. By beamforming the coherent field conventionally at a single frequency, one can determine the bearing of the source and hence the invariant-to-range ratio. In the present work, striation-based beamforming is developed. Rather than pick a single frequency and then beamform, each point on the array is first evaluated at a slightly different frequency with the frequency offset determined by the slope of the observed intensity striations. It is shown that the striation-based beamformer can be used to produce an estimate for the waveguide invariant that is independent of the source range. Simulation results are presented. [Work supported by the ONR.]
Soundscape of band room: Listening criteria.
Authors: Tsaih Lucky
Published: July 2011

Three band rehearsal videos have been analyzed to investigate the specific issues that conductors address or work on during a band rehearsal. These specific issues were also identified by the conductors and music students as the listening criteria during the rehearsal through interviews with conductors and questionnaires administered to music students. The analysis of video recordings of three band rehearsals shows that the conductor addressed to issues related to intonation (17%), rhythm (27%), dynamics (18%), articulation (19%), and other musical issues such as style, phrasing, tone quality, breath release time, and/or nonmusical issues such as student’s discipline (21%) of the time. Questionnaires given to student musicians and faculty instructors yielded similar results with 73%–94% of students and 50%–83% of instructors naming these as primary issues dealt with in college and high school music rehearsals. This raises questions that these issues should form the basis for acoustical design criteria for this building type.
Plumbing specifications for low noise, where are the codes and standards?
Authors: Brooks Bennett M.
Published: July 2011

Effective materials and installation techniques for low noise plumbing designs have been known for many years. However, with the advent of value engineering (VE), many of the traditional materials and installation methods have disappeared from all but a few projects. For example, what happened to the cast iron and copper that we all know and love? Also, separate, well insulated plumbing chases are now a rare luxury. These factors increase the challenges to develop acceptable plumbing designs from a noise perspective. Further, what do the applicable codes and standards say about plumbing noise specifications? The implications that these aspects of the design process have on plumbing noise performance are discussed.
Optimizing transient beamforming with FOCUS ultrasound simulation software.
Authors: McGough Robert J. , Kruse Dustin E.
Published: July 2011

The speed and accuracy of FOCUS relative to other programs provide a significant advantage for the evaluation of transient beamforming algorithms. In the farfield of individual array elements, effective transmit focusing for transient beams is achieved when the time delays are determined from the distance between the center of the element and the desired focal point. However, when focusing in the paraxial region near the phased array, the farfield approach for calculating time delays is suboptimal. To address this problem, an optimal beamforming algorithm is derived for transient nearfield simulations in FOCUS. This beamforming algorithm utilizes an approach that rapidly converges to the optimum value by computing the transient pressure at three or four equally spaced time points for each element. The beamforming algorithm sequentially evaluates the optimal time delay for each array element, achieving an additional reduction in the computational effort by applying the computed offset as the initial offset for the next element. After quickly determining the location of the pressure peak in each transmitted waveform, the algorithm selects the time delays that achieve the maximum value of the peak pressure at the intended focus. [Work supported in part by NIH Grant No. EB012079].
Acoustic and articulatory characteristics of clinically resistant /r/.
Authors: Boyce Suzanne E. , Combs Sandra , Rivera-Campos Ahmed
Published: July 2011

Acoustically, there is strong evidence to suggest that achievement of an appropriately low third formant for American English /r/ depends on successful simultaneous production of two vocal tract constrictions by the tongueone constriction by tongue tip or dorsum at the palate and one constriction by tongue root in the pharynx [Tiede et al. (2004); McGowan et al. (2003); Boyce (2008)]. The timing and extent of movement for these constrictions also appear to differ according to pre- versus postvocalic word position. In a study of older children with persistent difficulty producing /r/, clinic notes at their initial clinic visit indicate that two-thirds were missing a pharyngeal constriction during /r/ for at least one of these prosodic conditions. Clinic notes also indicate that improvement in /r/ production was accompanied by achievement of simultaneous palatal and pharyngeal constrictions. In this paper, we report an analysis of the correlation between these subjective clinical observations and more objective ultrasound imaging and acoustic data collected simultaneously on the same subjects during therapy for /r/ production.
Constraints on the learning of foreign accents.
Authors: Trude Alison M. , Brown-Schmidt Sarah
Published: July 2011

Understanding foreign-accented speech presents challenges for listeners. Three experiments tested learning of a foreign-accented vowel shift and its application during on-line speech processing. Native Quebec French and English speeches were compared using a visual world eye-tracking task. The French talker pronounced /i/ as /ɪ/ before all consonants except voiced fricatives. On critical trials, participants viewed pictures of an unaccented /i/ word (e.g., bees) and accented /i/ word (e.g., beet, pronounced [bt] by the French talker), then heard one talker say the target word. On French-talker trials, listeners should rule out the competitor because it does not share a vowel with the target, reducing competition. However, fixation measures showed comparable competition on “bees” trials and increased competition for the French talker on “beet” trials, indicating difficulty in applying knowledge of the accent. Performance on bees trials improved in Exp2-3, where stimulus variability was reduced by presenting only critical words with /t/ codas. On beet trials, performance improved most on Exp3, in which the unaccented counterpart to the accented word (e.g., bit, for beet) never appeared. The results suggest reducing linguistic variability is helpful in learning the accent, but lexical competition must also be reduced for successful processing of accented words.
Underlying sources of individual differences: Cognitive abilities in second language phonological development.
Authors: Park Hanyong , Darcy Isabelle , Yang Chung-Lin
Published: July 2011

Research on individual differences has identified factors constraining L2 acquisition in terms of a global performance; yet little progress has been made in identifying specific predictors of phonological acquisition. To explore such potential predictors in L2-learners, working memory, selective attention, processing speed, vocabulary size, and naming speed in L1 and/or L2 were assessed. These scores were then compared to individual L2-phonological acquisition index scores, which were obtained by combining the scores from three tasks: an ABX task (testing acquisition of English phonetic categories), a sequence repetition task (testing acquisition of English stress), and a speeded lexical decision task (testing the encoding of English syllable structure). The results were based on data from 20 Korean learners of English with varying lengths of US-residence (LOR) and 10 English native speakers as controls. In all the tasks, the native speakers performed better than the longer-LOR learners, who, in turn, did better than the shorter-LOR learners. There were large individual differences within each group. Individual working memory measures were significantly correlated with the individual L2-phonological acquisition index scores. Further, specific task scores correlated independently with specific cognitive measures, suggesting that a balanced mix of cognitive abilities can be the key to better phonological acquisition. [NIH-NIDCD Training Grant No. T32-DC00012.]
Intra- versus inter-talker effects in multi-talker perceptual training.
Authors: Wright Richard
Published: July 2011

Learners of non-native speech sounds benefit from intensive multi-talker training [e.g., Logan et al. (1991), Yamada (1993), and Bradlow et al. (1999)]. This training is thought to produce highly generalized perceptual learning and result in long-term modifications of the perceptual system. However, it is unclear whether the multi-talker benefit is due to inter-talker variation, to intra-talker variation, or to some combination thereof. In this experiment intra-talker (style) variation was fully crossed with inter-talker variation. 48 subjects were trained in one of four set stimuli [(1) single-talker single-style, (2) single-talker dual-style, (3) dual-talker single-style, and (4) dual-talker, dual-style]. Stimuli were nonsense words composed of either all native sounds or containing non-native speech sounds (⌊y⌋ or ⌊x⌋) presented with randomly paired nonsense shapes (fribbles). Subjects were probed at each quartile of the training to monitor learning of fribble names (correct versus incorrect fribble-name pairing). Subjects were then tested in a generalization task with new talkers. Results of this experiment indicate an important role for intra-talker variation in the ability of learners to generalize to novel exemplars of trained words. This result holds for both types of stimuli: nonsense native and nonsense non-native, indicating that multitalker training benefits extend to native word learning.
The influence of amplitude modulation frequency on perceived roughness of vowels.
Authors: Shrivastav Rahul , Eddins David A.
Published: July 2011

The goal of the current experiment is to determine how perceived roughness of vowels is influenced by the frequency of inherent amplitude modulation. A set of 10 vowel stimuli was synthesized using a Klatt synthesizer and were modeled after ten talkers selected from the Satloff/Heman-Ackah disordered voice database. These vowels were selected using stratified sampling procedures to represent a wide range of roughness. The vowel f0 and the first three formants were determined and used to generate a synthetic copy of these vowels. The aspiration noise level was adjusted to subjectively match the overall quality of the talkers. To each of the ten prototypical vowels, sinusoidal amplitude modulation was superimposed with a modulation depth of −6 dB and modulation frequencies ranging from 10 to 70 Hz. Listeners judged roughness in a matching task where the modulation depth of a comparison sound was adjusted to match the perceived roughness of each of the 70 speech stimuli (10 talkers × 7 modulation frequencies). The dependence of perceived roughness on amplitude modulation frequency will be compared to the perceived roughness of sinusoidal carriers as reported by Fastl and Zwicker (Springer, New York, 1997).
Improving classification of underwater objects by optimal signal design.
Authors: Hamschin Brandon , Loughlin Patrick
Published: July 2011

Detection, classification, and localization of underwater objects is a primary function of active sonar systems. Detection involves making a decision on whether or not an object of interest is present. Once a positive detection has been made, further information may be needed to classify the object as one among a set of possible objects of interest. Previous efforts have been directed at designing transmit sonar waveforms to maximize detection performance. In this work, we extend the optimal sonar design approach to enhance classification after detection. In particular, we present an optimal signal design approach that is aimed at maximizing the probability of correctly classifying the true target from among a set of assumed candidates. The approach is evaluated theoretically and via simulations, by which it is shown that waveform design can yield improvements in classification performance. [Work supported by ONR.]
A method of image fusion based on bidimensional empirical mode decomposition.
Authors: Jian Wan , Chunhui Zhao , Longtao Ren
Published: July 2011

Based on the principle of bidimensional empirical mode decomposition (BEMD), a new method of image fusion is presented. The processing procedure can be divided into three steps: (1) Every image to be fused is decomposed to several intrinsic mode functions (IFMs) and a residual component by BEMD; (2) according to the characteristic of BEMD, the IFMs in different frequency bandwidths are fused separately into new components, using linear weighted rule, whose coefficients can be required by the calculation of the mean-square-error or the information-entropy of the IFMs; (3) these new components are fused finally, obtaining the fused image. The simulation results show that the image fusion based on BEMD using the character of frequency has more advantages than that using the local energy of the pixel, and is also superior to the wavelet image fusion on pixel level.
Multi-carrier synthetic aperture communication in shallow water.
Authors: Kang Taehyuk , Song Hee Chun , Hodgkiss William
Published: July 2011

Orthogonal frequency division multiplexing (OFDM) communications in the presence of motion is investigated using data collected from the Kauai Acomms MURI 2008 (KAM08) experiment, conducted off the western side of Kauai, HI, in June-July 2008. The experiment involved a vertical array moored in 106-m-deep shallow water and a source towed at a speed of 3 kn at ranges between 600 m and 6 km. In order to attain reliable communications with only a single receive element, a synthetic aperture approach is applied. After combining multiple transmissions, an error-free reception is achieved with a low-density parity check code, confirming the feasibility of coherent synthetic aperture communications using OFDM.
Computation of bit error rate for underwater communication with and without low-salinity water in the western sea of Jeju.
Authors: Bok Tae-Hoon , Kim Juho , Paeng Dong-Guk , Lee Chong Hyun , Bae Jinho , Pang Ig-Chan , Kim Seongil
Published: July 2011

Salinity is one of variables in the empirical formula of the sound speed (SS) of seawater. Its effects on the SS are generally neglected because the average salinity is 34 psu and varies within a few psu seasonally and spatially in the ocean. Recently, low-salinity water around 25 psu flows into the western sea of Jeju Island in Korea due to the Yangtze River flood in China during summer. In this paper, it was analyzed how the low-salinity water affected the SS profile (SSP) and the underwater communication. The SSP was calculated with and without the low-salinity layer, and the communication channel was estimated by the simulated acoustic eigen-rays. The bit error rate (BER) was computed using binary phase-shift keying modulation, and the effects of the low-salinity water on the BER were investigated. The SSP was changed to a positive slope by the low-salinity layer at the sub-surface up to 20 m of depth, forming the acoustic wave guide which mostly resulted in the decrease of the BER. Consequently, this paper suggests that it is important to consider the low-salinity water near Jeju for the underwater acoustic modeling and communication. [Work supported by UD100014DD.]
Anti-crosstalk technology based on the adaptive estimator of instantaneous frequency.
Authors: Jin Fu , Guolong Liang , Guangpu Zhang , Yan Wang
Published: July 2011

The bandwidth of underwater acoustic channel is limited. Anti-crosstalk technology is used in many acoustic fields to increase the utilization ratio of bandwidth. Based on the principle of adaptive estimator of instantaneous frequency, a new anti-crosstalk method is presented, which uses the matched filter to resist the noise and estimate the time-delay and uses the mean square error of instantaneous frequency as detection statistics to pick out the crosstalk interference from the same or nearby acoustic channel. The simulation results show that this method is not only useful in the detection of signals in low SNR but also is effective in the suppression of channel crosstalk, and the method can adapt to Doppler mismatch very well.
Case study: Patient admissions/waiting area speech privacy.
Authors: Tocci Gregory C. , Foulkes John T.
Published: July 2011

Patients forming a queue at a transaction counter are being overheard by persons next in line in the queue. The compromised speech privacy, besides being a potential violation of HIPAA and HITECH regulations, may lead to patients not fully disclosing conditions they may have, fearing they might be overheard by others in the queue. This might led to important details of medical conditions not being addressed during subsequent medical treatment. This case history summarizes the applicable regulations, design objectives for speech privacy, and the methods for improving speech privacy including cordoning the queue away from the transaction desk, sound absorption, barriers, and the use of electronic sound masking. The presentation describes an existing medical reception area. Speech privacy improvements are currently being installed. Privacy indices were computed using EASE. This paper will discuss the benefits of the speech privacy improvements studied, particularly the use of electronic sound masking which was both required to achieve the design objective and the technique that provided the greatest improvement in speech privacy.
Equine vocalization: A comparison of Arabian and Morgan horse whinnies.
Authors: Browning David G. , Herstein Peter D.
Published: July 2011

Horses tend to be more vocal when in a barn in order to compensate for reduced visibility, restricted mobility, and close access by intruders. This can express itself in such ways as a vocal greeting to the arrival of a feed cart. Under these conditions it has established that the spectral content of whinnies can vary between specific behavioral situations, although we are still a long way from determining what this implies. Here we compare whinnies for two different horse breeds (at two different locations), Arabians (Michigan State University) and Morgans (University of Connecticut) under similar situations. Arabians appear to have a wider range of frequency variation at times even approaching the rapid sweep of their fellow artiodactyls, the tapirs. On the other hand, Morgans can exhibit a stronger tremolo, which many associate with whinnies. In general there does appear to be a significant difference between the two horse breeds in their vocal responses.
Some reflections on chorusing.
Authors: Stocker Michael S.
Published: July 2011

Sound and song production in many animals is sexually dimorphic inasmuch as the males of most species either exclusively produce sounds, or produce the more complex sounds of the species. As a consequence, chorusing is commonly framed under the rubric of competitive strategies of breeding fitness advertisement for individual animals while ambiguating the individual sources of sound from predators within the sound of the group. Chorusing can be either synchronous (common in many stridulating insects) or asynchronous (common in anurans and fish). Other strategies may also be present in chorusing, including collective annunciation of the group fitness and physical extents, and identification of external threats to the chorusing “acoustic community.” This paper explores some other aspects of “chorusing” that includes other group-acoustic behaviors such as fish schooling and synchronous bird flocking behaviors.
Nanoparticle-mediated ultrasound-guided photoacoustics in imaging and therapy of cancer.
Authors: Emelianov Stanislav
Published: July 2011

A morphological, functional, and molecular imaging technique capable of visualizing biochemical, pharmacological, and other processes in vivo and repetitively during various stages of tumor progression and cancer treatment is desired for many biomedical, preclinical, and clinical applications. Over the past several years, we have been developing an integrated imaging approach—ultrasound-guided photoacoustic imaging. Ultrasound is used to visualize anatomical structures and photoacoustics is used to provide functional information about tissue using spectral differences between absorption of oxygenated and deoxygenated hemoglobin and cellular activity using molecularly targeted plasmonic nanosensors. In this paper, the ability of ultrasound and nanoparticle-mediated photoacoustic imaging to simultaneously obtain the anatomical and molecular map of tumor in vivo will be demonstrated. Furthermore, the role of ultrasound and photoacoustic imaging in therapy of cancer will be discussed. Finally, the development and future directions in ultrasound-guided photoacoustics and contrast agents will be discussed.
Photoacoustic imaging of ultrasound-induced displacements in ocular tissues.
Authors: Silverman Ronald , Urs Raksha , Lloyd Harriet , Ketterling Jeffrey , Kong Fanting , Chen Y.C.
Published: July 2011

Acoustic radiation force produces compression that can be used for assessment of tissue stiffness, generally by detection of displacements in the phase-resolved pulse/echo ultrasound waveform. In this report, we describe use of photoacoustics for tracking displacements produced by acoustic radiation force in the iris and retina. The probe consisted of a 20 MHz transducer with a central aperture through which 532 nm laser pulses were introduced. Laser pulses were emitted simultaneously with excitation of the transducer by a 20 MHz monocycle. For a 12 ms period, we interleaved tonebursts (generating force) between successive laser pulses/monocycles. Iris displacements averaged 26.5 and 76.7 μm photoacoustically versus 24.9 and 71.3 μm by pulse/echo at 60 and 100 W cm-2 force, respectively. For the retina, the displacements were 14.6 and 25.3 μm photoacoustically versus 12.4 and 25.8 by pulse/echo. Pulse/echo displacements were more difficult to discern due to their broadness and anatomic non-specificity. The photoacoustic signal is advantageous because of its broadband character and because the photoacoustic signal is only generated by specific molecules, which in the retina would correspond to melanin in the pigment epithelium. As little is known about retinal elasticity, this technique offers an avenue toward investigation of this tissue property.
Real-time monitoring of high-intensity focused ultrasound ablations with photoacoustic technique.
Authors: Cui Huizhong , Yang Xinmai
Published: July 2011

High-intensity focused ultrasound (HIFU) has been widely used in clinics as a non-invasive technique for soft tissue ablation. Real-time monitoring of HIFU process is required in order to perform HIFU. A photoacoustic (PA) imaging system was used to monitor HIFU ablation process in this study. Single-element, spherically focused ultrasonic transducers with center frequencies of 5 and 10 MHz were used to generate a HIFU field and detect the PA signals in beef kidney during HIFU treatments, respectively. A 25-micro-diameter thermocouple was used to measure the temperature rise during the treatment as well. Thermal dose, which was used to indicate the coagulation of soft tissue, was calculated with the temperature measurement from the thermocouple. Detected PA signals were therefore related to the coagulation of soft tissue during HIFU through thermal dose calculations. In addition, PA signals from beef kidney coagulating under a constant temperature was obtained to show the changes of PA signals under a constant temperature during soft tissue coagulation. We demonstrate that thermal dose is a more appropriate way to monitor in the treatment process instead of monitoring temperature, since the monitoring of temperature change through the detected PA signals during HIFU ablation may not be feasible.