bluuebunny/crossref_metadata_2025 huggingface dataset:
Changes in acoustic characteristics of /r/ following production training using electropatatography
Authors: Schmidt Anna
Published: October 2013

Visual feedback using electropalatography (EPG) is an effective method for changing production of speech sounds in children. Differences in tongue placement and general acceptability of production have been demonstrated with EPG training for tongue to palate contact. Little is known, however, about changes in the spectral characteristics of speech sounds following training. The current study examined pre- and post-EPG patterns for /r/ in prevocalic, intervocalic, and postvocalic positions in sentences with 5 weeks of intensive EPG training. In particular, pre- and post-spectral characteristics of /r/ were examined for differences pre- and post-treatment. Subjects were four children (aged 9–10). These children exhibited distorted /r/ despite several years of traditional speech therapy but learned to produce perceptually acceptable /r/ with EPG. Spectral characteristics for /r/ productions in the same sentences by four typically developing children were used for comparison. Differences will be discussed in terms of acoustic characteristics as well and changes in tongue position.
Acoustic features of speech after glossectomy
Authors: Ishimoto Yuichi , Suzuki Noriko
Published: October 2013

The tongue serves an important role in speech production. Speech disorders due to tongue resection often occur and cause communication difficulties after surgery, according to the extent of the resected area. Such patients typically require appropriate speech therapy and palatal augmentation prosthesis. However, diagnosis of the speech in such patients has been based largely on subjective evaluation by clinicians. Acoustic features in glossectomee have yet to be clarified, and there is no objective estimation method for speech intelligibility. This study aims to clarify acoustic features associated with speech disorders after glossectomy and to develop a computerized evaluation system. The speech data were analyzed using the improved cepstrum method and compared with varying degrees of speech intelligibility. The vowel /i/ was selected as the speech sample because it is the high front vowel. The results showed that spectral envelopes of the Japanese vowel /i/ with low speech intelligibility exhibit two peaks in the second formant (F2) region from 1500 to 3000 Hz. This suggests that the spectral features in the F2 region of the vowel /i/ are important clues to determine speech intelligibility.
Spasmodic dysphonia in Japan: 1486 Botulinum toxin injections for 260 spasmodic dysphonia patients
Authors: Kumada Masanobu , Kobayashi Takeo
Published: October 2013

Botulinum toxin injection is a very effective treatment for spasmodic dysphonia. We experienced 1486 injections for 260 patients in past 15 years in Japan. Here we show statistics of these injections. We use type A toxin that is produced by Clostridium Botulinum. This toxin affects the neuro-muscular junction as a blocker of release of Ach, which leads to inversible parasis of a muscle. Mainly, we inject the toxin paracutaneously on the neck to the vocalis muscle using a partly covered needle that works as an EMG electrode. Mainly we have four options: 2.5 i.u. unilaterally (38% of 256 patients), 2.5 i.u. for each laterality bilaterally (16%), 5.0 i.u. unilaterally (29%), and 5.0 i.u. for each laterality bilaterally (5%). Its effectiveness persisits for 17.3 weeks on average. The longest effectiveness we have ever experienced is more than 1 year. A small percentage of the injections are not effective; it can be thought that the injected toxin does not reach the neuromuscular junction even when the needle is in the muscle: neuromuscular junctions contribute a part of the muscle and EMG signals do not tell us where these junctions are. Mainly, the side effects are breathy hoarseness and misdiglutition, which persist a week or so.
Numerical simulation of sound originated from the vocal tract in soft neck tissues
Authors: Otani Makoto , Hirahara Tatsuya , Adachi Seiji
Published: October 2013

A nonaudible murmur (NAM), a very weak speech sound produced without vocal vibration, can be detected by a special NAM microphone attached to the neck, thereby providing a new communication tool for use with functional speech disorders. The microphone is a condenser microphone covered with soft-silicone impression material that provides good impedance matching with soft tissues that are found in the neck. The NAM detected with the NAM microphone, however, is insufficiently clear because higher-frequency components are suppressed severely. A production mechanism of the NAM as well as transfer characteristics of the NAM in soft tissues of a neck should be clarified to improve the NAM clarity. As a first step, sound propagation from the vocal tract to the neck surface is simulated using a head model scanned with magnetic resonance imaging and finite difference method. Results show that soft-meat-tissue-conducted sound has fewer higher-frequency components than an air-conducted sound. The decay of transfer characteristics is approximately −10 dB/oct in the audible range, which is roughly parallel to the spectral characteristics of the measured NAM. [Work supported by SCOPE-S.]
Computer graphics facial animation of Japanese speech using three-dimensional dynamic visemes
Authors: Hirayama Makoto J.
Published: October 2013

Speech production makes not only acoustic signals but also visual images of the face especially for the jaw, lips, teeth, and tongue. To reproduce realistic facial images during Japanese speech, a three-dimensional computer graphics model of Japanese visemes was made. These visemes were extracted from the video database of images around lips during speech captured by two high speed (up to 300 frames per second) cameras [M. J. Hirayama, Proc. ICPhS 2003 (2003), pp. 3157–3161]. Most of the visemes are created as static shapes. They are for five vowels, semi-vowels, and some consonants. For explosives by labials (/p/ /b/ /n/) or tongue (/t/ /d/ /n/), dynamic information, that is, multiple shapes and timing information were assigned for each viseme. By placing these visemes onto a time axis at the timing of phonemes of a sentence, then interpolating shapes in between by using a spline interpolation technique on speech articulators’ motion graphs, computer graphics animation was made by ray-tracing rendering. [A part of this work was supported by Japan MEXT Academic Frontier Project.]
A photoglottographic (PGG) method with external lighting and sensing
Authors: Honda Kiyoshi , Maeda Shinji , Hans Stephane , Crevier-Buchman Lise
Published: October 2013

Photoglottography (PGG) is a moderately invasive technique to observe glottal movements and vocal-fold vibrations. A light guide is inserted into the pharynx and transglottal light is detected by a photosensor placed on the outer skin below the glottis, or conversely the sensor is placed inside the pharynx and the subglottal cavity is illuminated from outside with a light source. This report evaluates a noninvasive PGG method, in which the hypopharynx is illuminated with a light guide placed on the neck skin at the level between the hyoid bone and thyroid cartilage and a photosensor on the skin near the cricoid cartilage. The light passed through the skin illuminates the lumen of the hypopharynx and the photosensor detects indirect (diffusion) light through the glottis during glottal adduction. Choosing an appropriate location for the light source and photosensor, glottal adduction pattern is observed for voiceless stops and fricatives in low as well as high vowel context. Thus, our diffusion-light PGG potentially overcomes the tongue-position dependency of the detected signals, which has been a major caveat of the conventional PGG methods.
An MRI-based 3-D tongue atlas for tongue modeling
Authors: Wu Chao-Min , Goh Han Tiet
Published: October 2013

The main objective of this study is to build an MRI-based 3-D tongue atlas with an established available spatial transformation technique. The subjects for the MRI data were eight male and eight female college students (19–28 years old) who are native speakers of mandarin with Taiwanese accent without speech disorders. The oral MR images (axial: TR, 400 ms; TE, 10 ms, FOV 24×24; image matrix, 256×256 for 35 slices with 2-mm thickness) were acquired using a GE SIGNA 1.5-T scanner in the University Hospital of Chung Shan Medical University. The axial MR images of the human tongue were first segmented with snakes active contour method, then the 3-D tongues of each subject were reconstructed with morphology-based gray-level interpolation. Finally, these 3-D tongues were spatial transformed into a 3-D tongue atlas with thin-plate spline method. The 3-D tongue atlases for male and female subjects and morphometric studies, such as subject-to-subject, subject-to-atlas, and male-to-female, of these two atlases will be presented. Implication of these data related to tongue modeling will also be discussed. These information will be used as the basis for future research on tongue modeling and underlying physiological mechanism of normal and disordered speech production.
Direct measurement of glottal volume velocity using high-speed, stereoscopic, particle imaging velocimetry
Authors: Berry David A. , Neubauer Juergen , Zhang Zhaoyan
Published: October 2013

Despite its central importance as the glottal source in several long-standing models of voice production, the glottal volume velocity signal has been relatively little studied. In particular, few direct measurements of the glottal volume velocity signal have been made. One common method for estimating the glottal volume velocity signal has been linear inverse filtering of the oral acoustic output. However, a major assumption of linear inverse filtering, and of the linear source-filter theory of speech production in general, is that the glottal source and the vocal tract are independent of each other. The accuracy of this assumption has been questioned, particularly at extremes of fundamental frequency and intensity. In order to test the limits of linear inverse filtering, direct, instantaneous measurements of glottal volume velocity have been made using high-speed, stereoscopic, particle imaging velocity on a self-oscillating, physical model of vocal-fold vibration. These direct measurements are contrasted with results from linear inverse filtering.
Tongue muscle fiber tracking during tongue protrusion and rest
Authors: Shinagawa Hideo , Murano Emi Z. , Zhuo Jiachen , Gullapalli Rao P. , Landman Bennett , Prince Jerry L. , Stone Maureen
Published: October 2013

Diffusion tensor imaging (DTI) has recently become a promising tool to investigate not only the nerve structure of the brain, but also integration of the peripheral muscle structure. The purpose of this study was to represent the functional deformation of tongue muscles with DTI by examining two conditions, tongue protrusion and rest. Three normal volunteers participated in the study. They were recorded at rest and wearing a custom made tongue retaining device (TRD), often used for patients with sleep apnea. The TRD held the tongue immobile in protruded position. In both positions, the tongue muscle structure and functional deformation could be visualized. In particular, the deformation of the genioglossus (GG) muscle, which is generally fan shaped, was clearly visible. The fibers of GG anterior muscle were more vertical with the TRD. The DTI technique has the potential to reveal not only myoarchitecture but also functional deformation of the tongue in vivo. With a little ingenuity, it might be possible to conquer the MRI sensitivity to small motion artifacts or improve tongue muscle stabilization to track the muscle fiber architecture during sustained speech. [Work supported by JSPS and NIH.]
Efficient use of a priori data in sediment inversions through the use of null space
Authors: Potty Gopu R. , Miller James H. , Lin Ying-Tsong , Lynch James F.
Published: October 2013

We present improved inversions for sediment parameters by incorporating a priori information about the environment. This improvement is achieved by projecting a desirable solution into the null space of the inversion and including this null space contribution along with the standard non-null space contribution. We use singular value decomposition (SVD) to define the null space of the inversion and elucidate our projection method. The desirable solution, which is projected into the null space, is constructed based on previous data from cores, geophysical surveys, and historic data. This approach introduces user bias into the solution; the projection onto the null space supplies a safety net by showing which aspects of this bias are justified by the data. The user bias can be considered as additional data which, when incorporated, can lead to meaningful solutions. Effectiveness of probing the null space will be compared to the more conventional nonlinear inversion schemes. This approach will be tested using field data collected as part of the Shelfbreak Primer experiment. [Work supported by ONR.]
Robust source localization and geoacoustic inversion in the Haro Strait Primer
Authors: Jain Rashi , Michalopoulou Zoi-Heleni , Tolstoy Alex
Published: October 2013

Gibbs sampling, a Markov chain Monte Carlo technique, has been shown to be apowerful tool for geoacoustic inversion and source localization. By providing estimates of posterior joint distributions, it offers a global optimization route for multidimensional estimation that reports uncertainty and covariance in addition to point estimates. In this work, Gibbs sampling is applied for extracting time delays from recorded time series during the Haro Strait primer experiment. Employing time delay estimates and using a linear approximation to the inverse problem and then regularization, estimates are obtained for source and receiver location and some environmental parameters. Simultaneously processing receptions at all three vertical line arrays for localization of each source reduces ambiguities in the estimation process. Similarly, using received signals corresponding to several sources, array element localization for a single array, a difficult problem for this particular data set, becomes more precise. Multiple data sets are used in the inversion and consistent results validate the robustness of the approach. Estimated bathymetry is in agreement with bathymetric maps for the region. [Work supported by ONR.]
Geophysical parameter inversion in a range-dependent environment
Authors: Seong Woojae , Lee Keunhwa , Kim Kyungsup , Kim Seongil
Published: October 2013

Matched-field inversion technique is applied for estimation of geophysical parameters of the ocean bottom in a range-dependent shallow water. In the experiment (MAPLE-4), conducted off the coast of the East Sea during May 2005, narrow-band multitone cw acoustic data were obtained from the towed moving source along a weakly range-dependent track, from 2 to 18 km apart from the L-shaped receiver array. In the inversion, complex density model based on Biot model is used to invert for parameters including porosity and permeability. Inversion results are compared with existing geological survey data. In addition, the effect of range dependency resulting from the seafloor slope and the existing bottom intrusion is examined.
On the use of acoustic particle velocity fields in adjoint-based inversion
Authors: Meyer Matthias , Hermand Jean-Pierre , Smith Kevin B.
Published: October 2013

Following the recent interest in the use of combined pressure and particle motion sensors in underwater acoustics and signal processing, some general aspects regarding the modeling and multipath phenomenology of acoustic particle velocity fields in shallow water environments have been studied. In this paper we will address a number of issues associated with the incorporation of vector sensor data (pressure and particle velocity) into adjoint-based inversion schemes. Specifically, we will discuss the ability of a semi-automatic adjoint approach to compute the necessary gradient information without the need for an analytic model of the adjoint particle velocity field. Solutions to the forward propagation of acoustic pressure are computed using an implicit finite-difference parabolic equation solver while the particle velocity is calculated locally at each grid point. Some numerical examples of vector sensor inversion results are provided. [Work supported by Royal Netherlands Navy.]
Sensitivity analysis of a geoacoustic parametrized model and its application to inversion of seabed properties
Authors: Liu Jin-Yuan , Chu Chung-Ray
Published: October 2013

The work first aims to analyze the parametrized geoacoustic model proposed by Robins [J. Acoust. Soc. Am. 89, 1686–1696 (1991)], in which the density and sound speed distributions vary with respect to depth as a generalized-exponential and an inverse-square function, respectively. The model contains a set of parameters that, by appropriate selection, may fit well the realistic geoacoustic variations [E. L. Hamilton, J. Acoust. Soc. Am. 68, 1313–1340 (1990)]. By choosing the plane-wave reflection field as an objective function, each model parameter is carefully analyzed to determine its range and sensitivity. Then, numerical simulation is employed to establish an inversion procedure, in conjunction with the application of acoustic wave reflection from a nonuniform seabed. Finally, a field experiment is designed and implemented to estimate the seabed acoustic properties based upon the model parameter inversion.
Characterization of sediment dynamics in an estuary environment using acoustic techniques
Authors: Hermand Jean-Pierre , Perichon Laura , Verbanck Michel
Published: October 2013

In recent years, acoustic-based methods have been developed to characterize the dynamical behavior of loose sediments and bed deposits in very shallow water environments. In this paper, we present preliminary results on the estimation of the dynamic changes in an estuarine environment using data from dual-frequency echosounding at high resolution and contemporaneous hydrological measurements including suspended matter concentration, density subbottom profiling, and data assimilation based on a sediment transport model. Those measurements are being conducted in the lower estuary of the Scheldt (Belgium) at the Sint Anna site where strong tide and season-dependent phenomena can be observed. This allows us to construct a ground-truthed, time-dependent geoacoustic model of the environment, i.e., a characterization of sound speed, density, and attenuation in function of time and depth. Synthetic acoustic data generated by that model will then be used to test inversion methods for monitoring sediment dynamics in real time.
Geoacoustic model for the New Jersey Shelf by inverting airgun data
Authors: Jiang Yong-Min , Chapman N. Ross , Badiey Mohsen
Published: October 2013

This paper describes geoacoustic inversion of airgun data acquired during the SWARM95 experiment. Hybrid optimization and Bayesian inversion techniques were applied to three airgun data sets recorded by a vertical line array. Optimization results are used to show the consistency of the estimates from all of the shots in terms of histograms and standard deviations of the inverted geoacoustic model parameters. The inversion results from the Bayesian approach are used to show the uncertainties of the estimates in terms of marginal distributions, MAP estimates, and credibility intervals. In the Bayesian inversion, full data error covariance matrices were estimated by ensemble averaging the covariance of the residuals of the measured and modeled data of inversions from many shots. The numbers of shots in the ensemble averages were determined by checking the temporal coherence of the signal. Statistical tests were used to test the validity of the assumptions in the Bayesian approach after incorporating full data error covariance matrices. With these inversion techniques, equivalent geoacoustic models with/without shear wave estimates are extracted for this experimental site. The frequency dependence of the p-wave attenuation, and the correlation between the geoacoustic parameters are obtained from the inversion results. [Work supported by ONR.]
Adapting results in filtering theory to inverse theory, to address the statistics of nonlinear geoacoustic inverse problems
Authors: Ganse Andrew A. , Odom Robert I.
Published: October 2013

The intrinsically non-Gaussian statistics of nonlinear inverse problems, including ocean geoacoustic problems, is explored via analytic rather than numerical means. While Monte Carlo Bayesian methods do address the non-Gaussian statistics in nonlinear inverse problems, they can be very slow, and intuitive interpretation of the results are at times problematic. There is great theoretical overlap between recursive filters/smoothers, such as the extended Kalman filter, and methods of linear and nonlinear geophysical inversion. The use of recursive filters in inversion is not in itself new, but our interest is in adapting statistical developments from one to the other. Classic analytic methods in both filtering theory and inverse theory assume Gaussian probability distributions, but newer nonlinear filters do not all make this assumption and are explored for their potential application to nonlinear inverse problems. The similarities and differences between the frameworks of filtering theory and inverse theory are laid out in a series of geoacoustic inverse problem examples. Recent work in nonlinear filters handles non-Gaussian probability densities that are constrained to a particular form, and also derives analytic expressions for higher order moments of these density functions. The application of these developments to geoacoustic inverse problems is addressed. [Work supported by ONR.]
Perceptual compensation for reverberation: Effects of noise-context bandwidth
Authors: Makin Simon J. , Watkins Anthony J. , Raimond Andrew P.
Published: October 2013

Perceptual compensation for reverberation is observed when the reverberation is applied to a test word (from a ‘‘sir’’-to-‘‘stir’’ continuum) and its context (e.g., ‘‘next you’ll get—to click on’’) are varied independently. Increasing reverberation in test words decreases listeners’ ‘‘stir’’ responses, as reverberation ‘‘fills the gap’’ that cues the [t]. Compensation occurs when the context’s reverberation is commensurately increased, and ‘‘stir’’ responses increase back to the level found with minimal test-word reverberation. Compensation is strongest with speech contexts but also occurs with some noise-like contexts, including ‘‘signal-correlated noise’’ that has the wideband temporal envelope of the original speech. Also effective is a wideband noise that is given the temporal envelope seen at the output of a single auditory filter in response to speech. A narrow-band version of this ‘‘auditory-filtered’’ noise is not effective, but when contexts are made by summing of three or five of these bands, their effectiveness increases correspondingly. Compensation appears to be informed by the ‘‘tails’’ that reverberation adds at offsets, so it merely requires contexts with suitable temporal-envelope fluctuations. However, effects seem confined to the context’s frequency region, so the crucial offsets need to be in a wide range of frequency bands. [Work supported by EPSRC.]
Aural localization of speech stimuli
Authors: Way Evelyn
Published: October 2013

Localization error was tested for a variety of signals to answer the question: do humans aurally localize different speech stimuli with a different localization blur? A series of tests was conducted comparing the effect of the sentence length, gender of the talker, and frequency content of the signals on localization. Results were applied to ongoing research into constructing an aurally accurate telepresence system.
Case study for voice amplification in a highly absorptive conference room using a negative absorption tuning by the Yamaha Active Field Control system
Authors: Watanabe Takayuki
Published: October 2013

The Yamaha Active Field Control (AFC) enhances and varies acoustical conditions from the room acoustics using the acoustical feedback in a system, which is properly tuned to the installed room acoustics. The system is primarily used to improve auditory impressions of reverberance, loudness by enhancement of reverberation in time, and energy in a music venue. Here, a negative absorption tuning is demonstrated as beneficial to the presentation room for a simple spoken speech used with a relatively excessive absorption at a middle-sized multiuse conference room. The space occupies 1100 ft2 with a ceiling height of 9 ft. It is designed for demonstration of sound equipment as well as lecture without audio equipment. Since the acoustics in the room are highly absorptive (average absorption coefficient, 0.3), considering a primary use for product demonstration, it causes a flutter echo between longitudinal walls due to less sound energy in a late field. We demonstrate here a system adjustment of the AFC system emphasizing voice amplification as well as an acoustical solution to a defect of a fluttering echo here. Acoustical descriptors such as RT, echo diagram, and SPL are discussed for a tuned room acoustics with the AFC system.
Suppression of speech intelligibility loss through a modulation transfer function-based speech dereverberation method
Authors: Unoki Masashi , Toi Masato , Shibano Yohei , Akagi Masato
Published: October 2013

The concept of modulation transfer function (MTF) can successfully be applied to evaluate the quality of speech transmission in the room acoustics [Houtgast and Steeneken, J. Acoust. Soc. Am. 77, 1069–1077 (1985)]. We previously proposed a speech dereverberation method based on the MTF concept, which consisted of MTF-based power envelope inverse filtering and the carrier regeneration in the filterbank [Unoki et al., EuroSpeech2003 (2003)]. This paper evaluates how the proposed method can suppress the loss of speech intelligibility caused by reverberation, by comparing various methods. We have carried out massive simulations of dereverberation for reverberant speech signals to objectively evaluate these methods. We also subjectively evaluated the methods via listening tests. In these simulations, artificial reverberations were convolved with a clean speech signal in which the impulse responses in the room acoustics can be approximated from the exponential decay as a function of reverberation time with a white-noise carrier. The results of both evaluations show that, in addition to reducing the averaged log-spectrum distortion by about 1 dB, the proposed method reduces the loss of speech intelligibility by about 30%. [Work supported by a Grant-in-Aid for Science Research from the Japanese Ministry of Education (No. 18680017).]
Influence of Deutlichkeit value and reverberation time on improved speech intelligibility in reverberant environments because of steady-state suppression
Authors: Hayashi Nahoko , Hodoshima Nao , Arai Takayuki , Kurisu Kiyohiro
Published: October 2013

To improve speech intelligibility in reverberant environments, Arai et al. proposed ‘‘steady-state suppression (SSS)’’ as preprocessing [Arai et al., Acoust. Sci. Technol. 23, 229–232 (2002)]. In this study, a perceptual experiment under artificial reverberant conditions with simulated impulse responses was conducted to elucidate the effect of the Deutlichkeit (D) value and reverberation time (RT) on improvements of speech intelligibility because of SSS. Artificial impulse responses were simulated with white noise multiplied by a decay curve. The advantage of this method is that the simulated impulse responses have mutually similar frequency characteristics; consequently, we can evaluate them using only the D value and RT regardless of their different frequency characteristics. Two parameters, the energy of the impulse response 50 ms from the direct sound and the attenuation rate of the decay curve, were controlled to obtain several impulse responses having certain D value and RT. Results show that SSS improved speech intelligibility in the conditions of low D value, even if RT was long or short. We could also interpret these results as indicating that processing is effective when the original speech intelligibility is less than 60%. [Work supported by JSPS.KAKENHI (16203041).]
Basic study on angular dependence of ultrasonic scattering from wire phantom mimicking myocardium
Authors: Onodera Teppei , Hasegawa Hideyuki , Kanai Hiroshi
Published: October 2013

It is reported that the ultrasonic scattering from myocardium varies periodically during a cardiac cycle. One of the reasons is considered to be the change of the angle between the ultrasonic beam and the direction of myocardial fibers. Therefore, it is expected that ultrasonic scattering from myocardium quantitatively characterizes the condition of myocardium. To investigate the angle dependence of ultrasonic scattering in relation to the fiber direction, in this study, ultrasonic echoes from a wire phantom (thinner than a wavelength in diameter) that mimics myocardium were measured as a function of the insonification angle. Two ultrasonic transducers of 7 MHz were employed for transmitting and receiving ultrasonic pulses. Focal points of these transducers were set at the same point on the phantom. A custom-made experimental system can change the azimuth and elevation angles while keeping the focal points at the same point. Measured ultrasonic scattering showed a significant change depending on the azimuth angle. Such changes depending on the azimuth angle were decreased by increasing the elevation angle. These preliminary findings support the hypothesis that the change in the angle between the ultrasonic beam and myocardial fibers due to the heartbeat varies the ultrasonic scattering property.
Three-dimensionality and temporal variability of the jet flow in organ pipes
Authors: Yoshikawa Shigeru , Sakamoto Yumiko , Bamberger Andreas , Angster Judit
Published: October 2013

Flow visualization by the particle image velocimetry (PIV) is expected to provide quantitative information based on high resolution from the use of sheetlike laser pulses. This PIV technique is applied to investigate the difference in jet-flow structure of organ pipes when the ear (small plate standing at each side of the mouth) is used or not. The experimental results given by PIV are also compared with those by the hot-wire anemometry. Two-dimensionality and laminar-flow condition tend to be maintained when the ear is used. However, when the ear is not applied, the deviation from laminar flow and the resulting three-dimensionality are apparent halfway between the flue and the edge (re less than 1800). The ear seems to block the entrainment from the mouth side. Moreover, it is shown that the magnitude of jet velocity is significantly reduced at the phase when the jet deflects innermostly to the pipe with the ear and at the phase when the jet deflects outermostly from the pipe without the ear. These 3D and temporal variability of the jet flow should be examined in more detail in the context of harmonic structure of organ tones and vortex-sound theory on flue instruments.
Sound generation in flutelike instruments: Simulation and analysis
Authors: Kuehnelt Helmut
Published: October 2013

Three-dimensional flow-acoustic simulations using the lattice Boltzmann method were used to study the sound generation in flutes and recorders. Theory suggests that the fluctuating Coriolis force in the mouth of the instrument caused by the fluctuating vorticity distribution of the jet is the main source of sound. A numerical experiment was undertaken to clarify this hypothesis for flute instruments. In the mouth of the instrument an equivalent temporal fluctuating force was imposed in a quiescent fluid. Although this force excited fluid dynamic effects in the mouth not present at mean flow, it nevertheless induced a standing wave in the instrument with a sound pressure whose amplitude differed only less than 1 dB from the playing situation. This shows the cardinal role of the Coriolis force for the sound generation in flutes. The temporal and spatial structure of the Coriolis force distribution and the vortex sound power generated will be compared for flute and recorder. Especially the spatial structure differs systematically for the two types of instruments. Implications for the design of flute head joints will be discussed.
Imaging vibrations and flow using electronic speckle pattern interferometry
Authors: Moore Thomas R.
Published: October 2013

Electronic speckle pattern interferometry has been used to measure sub-micrometer displacements and visualize the deflection shapes of harmonically vibrating objects for many years; however, to date the technique has not been widely used within the community of musical acousticians. The cost of the equipment, the need for effective vibration isolation, and the complexity of the hardware are usually cited as the reasons why this technique has not been more widely utilized in the study of musical instruments. To alleviate these concerns, a speckle pattern interferometer has been designed that is relatively inexpensive, easy to assemble, and very robust. Construction of the interferometer will be discussed, and applications to problems in musical acoustics will be presented.
Psychoacoustic experiments with virtual violins
Authors: Fritz Claudia , Cross Ian , Moore Brian C. J. , Woodhouse Jim
Published: October 2013

This study is the first step in the psychoacoustic exploration of why some violins sound better than others. A method was used that enabled the same performance to be replayed on different ‘‘virtual violins,’’ so that the relationships between acoustical characteristics of violins and perceived qualities could be explored. Recordings of real performances were made using a bridge-mounted force transducer, giving an accurate representation of the signal from the violin string. These were then played through filters corresponding to the admittance curves of different violins. Initially, the limits of performance in detecting changes in acoustical characteristics have been characterized. Thresholds were measured for the detection of different modifications of a violins acoustical response, such as a shift in frequency or an increase in amplitude of one or several modes, using a three-interval forced-choice discrimination task. Thresholds were higher for an input of a musical phrase than for a single note, but depended strongly on the choice of the note. The lowest threshold corresponded to a simultaneous shift in frequency of 1.5%, or an increase in level of 3 dB, of several modes, but thresholds appear to be dependent on the musical training of the listeners.
Numerical analysis of the 48-key experimental piano
Authors: Hasegawa Kenji , Nishiguchi Isoharu , Sasaki Masataka
Published: October 2013

An experimental piano has been built as a part of research program for students in the 4th grade. Though the main purpose of the building was to use the piano in vibrational and acoustical experiments, it was also intended to deepen the understanding of the structures of a piano by newly designing and building a piano that was different from existing pianos. The range covered by the piano is 4 oct (key nos. 21–68). After the sizes and arrangements of the strings were determined, the metal frame that sustains the tension of the strings was designed. The deformation and the stress were evaluated by the FEM. In this paper, the design and results of numerical analysis of the strings and soundboard are reported.
An exact point-source starting field for the Green’s function parabolic equation in outdoor sound propagation
Authors: Gilbert Kenneth , Di Xiao
Published: October 2013

A method for exactly representing a point source in a Green’s function parabolic equation calculation is presented. The method is no more difficult to use than standard starting-field formulations and has the several advantages: (1) the point-source solution is essentially exact; (2) only the propagating Fourier components are used; and (3) there is no limitation on the source height, which can be taken to be at z=0, if necessary. The method is derived analytically and examples of the numerical implementation are given. [Research supported by U.S. Army Armament, Research Development and Engineering Center (ARDEC).]
Protocol for measuring dishwasher sound power levels
Authors: Herreman Kevin , Godfrey Richard
Published: October 2013

A leading retailer of home appliances came to the OC acoustics laboratory to evaluate the noise generated by dishwashers sold in their stores. A protocol based on a methodology used in Europe was used. It was modified to account for North American building standards. The standard provides an overall sound power level determination for the complete dishwasher cycle with and without the drying portion of the cycle. As a result customers making a purchasing decision can compare the noise performance of the various products using the same yard stick.
Optimizing seismo-acoustic sonar source array configuration for maximum radiated Rayleigh wave energy
Authors: Johnson Anthony N. , Scandrett Clyde L. , Baker Steven R.
Published: October 2013

A three-dimensional (3-D) continuum mechanics approach to the development of a time-dependent finite element model for optimizing the position and excitation of a seismo-acoustic sonar source array to detect the presence of buried landmines will be presented. Various source configurations will demonstrate the use of constructive and destructive interference, which maximizes the radiated energy of unidirectional Rayleigh waves while suppressing the radiation of body waves. Radiation characteristics are analyzed in a linear, horizontally stratified (isotropic and homogeneous within each layer) half-space with a discrete number of transient seismic sources. Results for Rayleigh wave strengths are presented in both a homogeneous half-space and a layered medium.
Accurate time domain computation of linear wave propagation using Chebyshev collocation and matrix diagonalization
Authors: Huang Lixi
Published: October 2013

Accurate time domain computation of the wave propagation phenomena is of great interest to many fields of studies. The spectral method of Chebyshev collocation is adopted for smooth problems of linear waves. Space differentiation is discretized by Chebyshev derivative matrices and the wave equation is cast as a second-order ordinary differential equation in time domain. The general boundary conditions involving pressure and acoustic particle velocity are discretized and absorbed into the system equations for the unknown vector containing all grid points. The system equations are solved accurately by the method of matrix diagonalization, which involves the finding of the eigenvalues and eigenvectors for the appropriate combinations of the damping and stiffness matrices. For this solution, the time step for stable and accurate computation is only limited by the consideration of round-off errors when the exponential functions are evaluated. Comparison is made with analytical solutions in the one-dimensional case, and good agreement is obtained. The only perceived drawback would be the size of matrices that can be alleviated by the use of domain decomposition approaches. The physical meaning of the eigenvalues derived from the discretization procedure is discussed.
Propagation phase and zeros in the complex frequency plane
Authors: Takahashi Yoshinori , Tohyama Mikio , Miyoshi Kazunori
Published: October 2013

This paper describes the transfer function (TF) including the coherent sound field as the distance between source and observation points increases, from a point of view of the number of zeros on a complex frequency plane. The propagation phase could be estimated according to the linear-regression analysis for narrow frequency-band frequency characteristics of the minimum-phase component [Inst. Electron. Inf. Commun. Eng. Jpn., J89-A(4) 291–297 (2006) (in Japanese with English figures)]. This paper analyzes zeros of the TFs for the coherent and reverberant fields based on the modal and random sound field theory. Consequently the distribution of zeros could be estimated in the transition from the coherent to the reverberant conditions.
Using instabilities in the long wavelength transverse optical mode to generate nuclear fusion
Authors: Naranjo Brian , Putterman Seth J.
Published: October 2013

Various crystals such as lithium niobate or lithium tantalate display a phase transition due to an instability in the long wavelength transverse optical mode. These low-temperature phases develop huge electric fields which can accelerate ions to energies where nuclear fusion is observed [Nature 434, 1115–1117 (2005)]. [Research funded by DARPA, ONR, NSF.]
Using tagged cine-MRI and finite-element method to lower bound the number of independently controllable motor units in the tongue
Authors: Essex-Torcaso Caroline , Levine William S. , Stone Maureen , Murano Emi Z. , Parthasarathy Vijay , Prince Jerry L.
Published: October 2013

The biomechanical structure of the tongue is unusual in that there is no rigid structure, such as a bone, for the muscles to act against. In order to understand these biomechanics and their control, a mathematical model has been created. This model has the form of a nonlinear controllable incompressible elastic structure that undergoes large deformations. The model was used to solve a simplified two-dimensional inverse problem. For a simple speech motion (/i-u/), the trajectories of specific points within the tongue were tracked by means of tagged cine MRI. These points correspond to finite element nodes and were used as input to the simplified model. The remaining unknowns in the model are the set of muscle activations that produce the observed motion. Because there is no such set of activations, the set that best approximates the observed motion in a least squares sense was chosen. The results indicate that there must be independently controllable compartments in SL and GG in order to produce this movement. Three methods were used to improve the representation of the tongue and the resolution of the model: refinement of the number of elements, refinement of the number of muscle activations, and the inclusion of dynamics.
A new aerodynamic critical parameter of the speech production: Phonation threshold flow
Authors: Jiang Jack J. , Tao Chao
Published: October 2013

In this paper, the implications of phonation threshold flow (PTF) as an addition to the aerodynamic parameters of speech production system are studied. PTF, the minimum airflow volume velocity able to sustain stable vocal-fold vibration, could have utility in clinical vocal disease assessment just like phonation threshold pressure (PTP) [I. R. Titze, J. Acoust. Soc. Am. 83, 1536–1552 (1988)]. Furthermore, because glottal airflow can be more easily measured noninvasively than subglottal pressure, PTF could be more suitable for routine clinical assessment than PTP. Theoretical studies indicate that PTF is a sensitive aerodynamic parameter dependent on tissue properties, glottal configuration, and vocal-tract loading. Accordingly, this theory suggests that PTF is reduced by the following: decreasing tissue viscosity, decreasing mucosal wave velocity, increasing vocal-fold thickness, or decreasing prephonatory glottal area. A divergent glottis and low vocal-tract resistance also reduce phonation threshold flow. Finally, the significance of PTF for investigation of the distribution of energy in a vocal-fold vibration system and the potential for application of the PTF parameter are assessed. [Work supported by NIH.]
Estimation of area function for three-dimensional vocal tract using finite-element method
Authors: Mochizuki Keita , Nakai Takayoshi
Published: October 2013

Numerous studies have obtained a vocal tract area function using data obtained from magnetic resonance imaging (MRI). However, this approach often results in increased error between formant frequencies derived by recorded voice and estimated frequencies derived by MRI. Two factors engender error: creating the vocal tract model and estimating the area function from the vocal tract model. Generally, the errors from those two factors are not considered independently. This study is intended to establish precise estimation of the area function automatically from a vocal tract model. Pressure contours and sound intensity in the vocal tract are obtained using the finite-element method (FEM); the vocal tract area function is obtained from them. The precision is assessed by comparing the formant frequencies that are derived using the FEM with our method. Results show that the vocal tract area function can be estimated precisely from three-dimensional MRI.
Temporal structures of articulatory movements: A contrastive study of stop consonants in Japanese, Korean, and Chinese
Authors: Kaburagi Tokihiko , Nakajima Yoshitaka
Published: October 2013

Stop consonants of Japanese, Korean, and Chinese were studied from an articulatory perspective, particularly in relation to the temporal structure of articulatory movements. In general, acoustical features of stop consonants, such as the amount of aspiration, voice onset time (VOT), and so on, are largely controlled by interarticulatory movements of the tract and glottis. In this study, several speakers were instructed to produce stop consonants of their native tongues. Then, states of the articulators were monitored using an electroglottograph (EGG) and pressure and flow sensors. While the EGG tracked vibratory patterns of the glottis, the pressure and flow sensors were used to measure the air pressure inside the vocal tract and the amount of air flowing out of the mouth. It is noteworthy that these languages have different sets of stops: the voiced and unvoiced stops of Japanese; the aspirated, weak, and strong stops of Korean; and the aspirated and nonaspirated stops of Chinese. Temporal structures of interarticulatory movements were analyzed from the observed data. Their respective acoustical consequences were compared among these languages. [Work supported by JSPS and 21st Century COE, Kyushu University.]
Intrusive vowels in Cruceno Spanish
Authors: Kilpatrick Cynthia , McGee Kathryn , Kirby James
Published: October 2013

Intrusive vowels are short vowels appearing within consonant clusters that are not treated phonologically like full vowels. Though reported in many languages, few quantititative studies have examined these vowels. Here, an acoustic study of intrusive vowels in obstruent+tap clusters in the Spanish of Santa Cruz, Bolivia is reported. The results support earlier studies in that the intrusive vowel quality resembles that of the following nucleic vowel, rather than a neutral vowel, and intrusive vowels are significantly longer in clusters with voiced obstruents than in those with voiceless ones. However, results do not fully support all previous phonetic descriptions and related theoretical assumptions. In particular, a significant difference is not found in intrusive vowel length based on variables such as place of articulation of the obstruent, quality of the nucleic vowel, or placement of stress in relation to the cluster. In addition, where previous work finds no significance for position in the word, the present work finds that intrusive vowels are significantly longer in word-medial clusters than in word-initial clusters. The data further suggest that the articulation of obstruent+tap does not require an intrusive vowel, as has previously been claimed, as not all obstruent+tap clusters include an intrusive vowel.