bluuebunny/crossref_metadata_2025 huggingface dataset:
The significance of relative loudness in listener identification of pipe organ stops
Authors: Disley Alastair C. , Howard David M.
Published: October 2013

As part of a wider study of the factors affecting listener identification of pipe organ stops, the authors’ previous research [A. C. Disley and D. M. Howard, ‘‘Onset transient significance in listener identification of pipe organ stops,’’ J. Acoust. Soc. Am. 119, 3333, abstract only (2006)] has suggested that the relative loudness of a stop as presented to a listener may have a significant effect on its identification. Recordings of a representative selection of pipe organ stops were played to a group of 43 organists in a formal listening test. Each stop was presented twice, once at its original amplitude and once at an increased amplitude equal for all stops. Listeners were asked to classify each example using nine general categories of organ stops. The results suggest that increased amplitude improves listener recognition for some stops (particularly open flute stops), but produces significant misidentification of other stops, especially those with the greatest difference between the original and increased amplitude. These results suggest that caution should be employed when considering equalizing sample amplitudes in similar listening experiments.
Remembering what’s missing: Modeling phoneme restoration and other complex forms of auditory induction as the triggering of a memory
Authors: Aronoff Justin
Published: October 2013

Although phoneme restoration is often spoken about and modeled as a unique phenomenon specific to language, the effect is very similar to restoration effects found with well-learned musical melodies. This suggests that both of these effects may be the result of a natural characteristic of learned temporal patterns, rather than a special mechanism. A recurrent connectionist network was designed to test this hypothesis. The network contained ten input and output nodes, which represented notes in a melody. The model was trained on sets of simple melodies, each consisting of eight randomly selected sequential notes. To test that the model was learning and identifying the melodies, the network was trained to predict the next note in the tune, which could only be accomplished by maintaining a representation of the sequence of prior notes. After the network learned the melodies, the fourth sequential note in each tune was removed and replaced by activation across all the input nodes, simulating noise. The model demonstrated accurate restoration, performing significantly better than chance, both at the noise position and over the notes that followed, suggesting that sequence learning may be sufficient to explain phoneme restoration and other forms of complex auditory induction.
Perceptual measurement of sound level differences between TV programs and advertisements
Authors: Miyasaka Eiichi , Kamada Takahiro
Published: October 2013

In order to investigate loudness difference between TV program materials and advertisements (CMs) inserted in the materials, sound levels of CMs broadcast in five terrestrial broadcasting stations throughout a day were quantitatively measured and a perceptual experiment was performed. The averaged sound levels (ASLs) and the standard deviations (SDs) of 4262 CMs were concentrated on −7 dB and 5 dB, respectively. Next, three types of CMs with different ASLs and SDs were used to the perceptual experiment. These ASLs and SDs were (−5.1 and 1.8 dB for CM-1), (−7.1 and 4.9 dB for CM-2), (−14.2 and 10.1 dB for CM-3), respectively. The reference sound as a main program material was speech uttered by a NHK female announcer with duration of 10 s. The ASL and the SD were −6.5 and 10.1 dB, respectively. The transformed up-down method was used for 12 observers with normal hearing. The results show that all CMs were perceived at least 3 to 5 dB louder than the reference for a half of the observers. These results show that there is large perceptual discrepancy between CMs and the main program materials. [Work supported by The Okawa Foundation.]
Recent developments and applications of energy finite- element analysis
Authors: Wu Kuangcheng , Vlahopoulos Nickolas
Published: October 2013

The energy finite-element analysis (EFEA) comprises a finite-element-based solution for vibration and acoustic modeling that uses energy-based variables for formulating the governing differential equations. This presentation covers the main points in the derivation of energy finite-element analysis (EFEA), validation through comparison with test data, and case studies that demonstrate the computational capabilities of the EFEA. The EFEA can be applied in various engineering disciplines as it will be demonstrated by the case studies. Thus, it comprises a general purpose simulation method for vibration and acoustic analysis of complex systems. [Work support for this research is provided by ONR code 331.]
Fast finite-element analysis for damping of automotive structures having elastic bodies, viscoelastic bodies, porous media, and gas
Authors: Yamaguchi Takao , Kurosawa Yoshio
Published: October 2013

A numerical method is proposed to calculate damping properties for automotive soundproof structures involving solid bodies, porous media, and air in two-dimensional regions. Both effective density and bulk modulus have a complex quantity to represent damped sound fields in the porous media. Particle displacements in the media are discretized using a finite-element method. For damped solid bodies, displacements are formulated using conventional finite elements including complex modulus of elasticity. Displacement vectors as common unknown variables are solved under coupled condition between solid bodies, porous media, and gas. Further, by applying an asymptotic method to a complex eigenvalue problem, explicit expressions of modal loss factor for the mixed structures are derived. The proposed methods yield appropriate results for some typical problems and this method diminishes computational time for large-scaled finite-element models. Moreover, it is found that damping can be coupled in the mixed structures. An expression to calculate a share of dissipated energy for each element in mixed structures is also derived. Damping behaviors in sound bridge phenomena are analyzed using the proposed method.
Effects of intermodal timing difference and speed difference on auditory-visual speech perception
Authors: Tanaka Akihiro , Sakamoto Shuichi , Tsumura Komi , Suzuki Yo-iti
Published: October 2013

Previous studies have shown that lipreaders possess insufficient sensitivity to intermodal timing differences and adapt to the timing difference when it is kept constant. This study investigated effects of intermodal timing differences and speed differences on auditory-visual speech perception. We used 20 minimal pairs of Japanese four-mora words such as mizuage (catch landing) versus mizuame (starch syrup). We administered intelligibility tests to younger and older adults. Words were presented under visual-only, auditory-only, and auditory-visual (AV) conditions. Effects of AV asynchrony by time-expanded speech (speed difference) on AV benefit (i.e., the superiority of AV performance over auditory-only performance) were compared with those by intermodal timing shift (timing difference). Results showed that the AV benefit declined as the timing difference and speed difference increased. Results also revealed that younger adults adapted to the timing difference but did not adapt to the speed difference. Older adults adapted neither to the timing difference nor to the speed difference. These results suggest that intermodal lag adaptation in younger adults requires a constant timing difference between a talker’s moving image and voice. These results might be useful for design of a multimodal speech-rate conversion system.
Child voice and noise; acoustic effects of a day at the day-care related to background noise levels
Authors: McAllister Anita , Granqvist Svante , Sundberg Johan , Sjolander Peta , Tronnier Mechtild
Published: October 2013

Several studies have found that high background noise levels are detrimental to health parameters. In particular, this seems to apply to developing voices were future vocal habits are established. Thus, it is important to study vocal function and environmental effects on the developing child voice. This study analyzed the effects of background noise on children’s voices, specifically vocal intensity, and fundamental frequency. The investigated vocal parameters were (1) the relationship of background noise levels to F0 and vocal intensity, (2) F0 and vocal intensity variations over the day, and (3) F0 perturbation variations over the day. Ten 5-year-old children from three day-cares participated, six boys and four girls. The audio signal was recorded by two microphones mounted in front of the subjects’ ears. By adding these signals it is possible to separate the voice from background noise. The material analyzed contained data from three 60-min recordings per child from morning, noon, and afternoon during a normal day at the day-care. Generally high mean background noise levels were found (82.6 dBA). Preliminary results suggest a correlation between high background noise and high F0 and vocal intensity in the children’s voices, particularly for boys. F0 perturbation tends to increase during the day.
Physiologic and behavioral classification of delayed speech: Syllable repetition tasks
Authors: Vick Jennell C. , Moore Christopher A. , Venkatesh Lakshmi , Campbell Thomas F. , Leavy Rusiewicz Heather , Shriberg Lawrence D. , Green Jordan R.
Published: October 2013

As part of a large-scale study that aims to create a data-driven model of speech delay of unknown origin (SD) in preschool children, this investigation analyzed two- and three-syllable nonword repetitions by 50 3- to 5-year-old children, half of whom are typically developing (TD). The syllable repetition task was composed of four different words, bama, bada, bamana, and manaba, selected both for articulatory simplicity and to maximize displacement of the external articulators for kinematic analyses. While children in the TD group were significantly more accurate in both two- and three-syllable repetitions than children in the SD group, neither group was able to repeat three-syllable nonwords with particular acumen. Children in the TD group were 55% accurate, while children in the SD group were only 22% accurate. Measures of repetition accuracy, acoustics, nasalence, external articulatory kinematics, and respiration contribute to a hierarchical agglomerative clustering model that generates hypotheses about potential subclinical categories within the SD and TD groups.
New generation aids for laryngectomy patients
Authors: Czyzewski Andrzej , Odya Piotr , Kostek Bozena
Published: October 2013

The artificial larynx has many disadvantages. The produced speech is monotonous and sounds artificially. In addition, produced speech intelligibility is usually poor. The major problem is a background noise caused by the device. In fact, the artificial larynx is only a simple vibrator, a construction of which has been almost unchanged since the 1950s. The aim of the presented project is to design a new generation of devices for laryngectomy patients. There are two different approaches to solve this task. The first one focuses on the artificial larynx. Some major improvements in the construction of the device might be easily introduced. Hence, the artificial larynx engineered was equipped with a digital processor and an amplifier. The spectral subtraction algorithm for noise reduction was used. In this method, an average signal spectrum and average noise spectrum are estimated and subtracted from each other, thus average signal-to-noise ratio (SNR) is improved. The main problem to be solved was that both the noise and the speech signal have the same excitation source and consequently are strongly correlated for voiced sounds. The second approach uses a PDA (portable digital assistant) to generate synthetic speech. The proposed new generation devices helping laryngectomy patients are presented in the paper.
Modeling aspects of vocal fold oscillations with validation to clinical data
Authors: Bentil Sarah A. , Reed Todd , Allen John S. , Yan Yuling
Published: October 2013

We investigate vocal fold vibrations in patients with muscle tension dysphonia (MTD) and Parkinson’s diseases (PD) using a refined two mass model [I. Steinecke and H. Herzel, J. Acoust. Soc. Am. 97(3), 1874–1884 (1995)] and clinical data obtained from acoustic recordings and high-speed digital imaging (HSDI) of the larynx. The two-mass model with empirical parameters is able to reproduce some of the behaviors revealed from functional, image-based analyses of vocal fold dynamics in both normal and abnormal states. In particular, the glottal area waveform (GAW) is used to describe the vocal fold vibratory behavior, which is derived from HSDI data using an analytical framework developed by the Yan research group [Yan et al., J. Voice 19(2), 161–175 (2005); IEEE Trans. Biomed. Eng., (2006)]. Further, a comparison of the two-mass model predictions and the GAW and acoustic data was performed using signal processing techniques including fast fourier transforms, wavelet transforms, and spectrograms. These analyses deliver a comprehensive quantitative representation of the dynamic characteristics of the vocal folds and may provide new insights into the mechanism of abnormal voice productions such as in MTD and PD. [Research is sponsored by NSF Grant ♯0627992.]
Preshearing effects on viscoelastic properties of collagen-based vocal fold injectables
Authors: Klemuk Sarah A. , Titze Ingo R.
Published: October 2013

A criterion for developing a biomaterial is that its viscoelastic properties correspond to native tissues. This criterion is particularly important for vocal fold tissues. Collagen-based injectables are routinely used to repair paralyzed or scarred vocal folds and restore vocalization, yet reported viscoelastic properties are inconsistent. The present study evaluated the effects of preshearing samples on their viscoelastic measurements. Non-cross-linked bovine collagen (Zyderm IITM) and micronized dermal tissue (CymetraTM) were presheared by extruding directly through a mechanically operated syringe with different needle sizes. Preshear rates ranged from 31 to 5600/s for orifice radii of 1.21 to 0.0955 mm, respectively. Sandpaper-coated, parallel plate attachments on a stress-controlled rheometer were used to measure elastic moduli and dynamic viscosities from 0.01 to 10 Hz. A 3–10-fold reduction in elastic modulus and a 2–4-fold reduction in dynamic viscosity occurred for preshear rates exceeding 1000/s compared to rates of 31 and 131/s. These results suggest a change in the molecular structure and the viscoelastic compatibility of vocal fold injectables when sufficiently presheared. [Work supported by NIH.]
Acoustic and aerodynamic effects of false folds and epiglottis
Authors: Alipour Fariborz , Jaiswal Sanyukta , Finnegan Eileen
Published: October 2013

Excised larynx research usually excludes the false folds in an effort to reveal the true vocal folds. The purpose of this study was to examine the aerodynamic and acoustic effects of the false folds and epiglottis on excised larynx phonation. Canine larynges were mounted over a tapered tube on the excised larynx bench that supplied pressurized, heated, and humidified air. Glottal adduction was accomplished by passing sutures that simulated lateral cricoarytenoid muscle activation. First, the excised larynx with intact false folds and epiglottis was subjected to a series of pressure-flow experiments with tension and adduction as major control parameters. Then, the epiglottis and false folds were successively removed and the experiment was repeated for each condition. The subglottal pressure, EGG, flow rate, audio signal, and sound pressure level were recorded during each experiment. Glottal flow resistance was calculated from the pressure and flow signals, while the EGG signal was used to extract fundamental frequency. It was found that these additional structures have positive contribution to the glottal resistance and sound intensity of the larynx. Also, vocal fold elongation and glottal medial compression caused an increase in the glottal resistance. [Work supported by NIDCD Grant No. DC03566.]
Notes on broadband seabed geoacoustic inversion in shallow water
Authors: Zhou Ji-Xun , Zhang Xue-Zhen
Published: October 2013

With increasing interest in shallow-water environments, where bottom interaction is a dominant effect, there is a need to acquire accurate information about seabed geoacoustic parameters. Due to the difficulties and high costs of directly measuring seabed parameters at low frequencies (LF) in shallow water (SW), seabed geoacoustic inversion from long-range propagation has become an active research area. Much progress on this subject, notably on inversion methods, has been made in the last two decades, mainly through the use of powerful numerical codes and data processing tools. Despite this, seabed geoacoustic inversion often fails to yield systematic LF seabed parameters in a broad band (such as speed/attenuation vs frequency) that are required for seabed geoacoustic modeling or for sonar performance prediction in the SW environment. This paper will discuss several basic concepts and technical issues related to SW geoacoustic inversion, including filter phase shifting, speed-attenuation coupling, hidden depth, and sea-surface effects. It shows that inverting the broadband seabed acoustic parameters is a delicate task that can often be subject to errors. These issues should be considered in designing experiments that test seabed geoacoustic inversion methods or in interpreting experimental results. [Work supported by ONR and NNSF of China.]
Anticipatory motion in piano performance
Authors: Goebl Werner , Palmer Caroline
Published: October 2013

Recent motor control studies of piano performance address the anticipatory movements of pianists fingers. The reported study investigated anticipatory motion under manipulations of performance rate or tempo. It was tested whether faster tempi require larger anticipatory preparation, as suggested in piano pedagogy literature. Sixteen skilled pianists repeatedly performed short isochronous melodies from memory at four different tempi (from 500 to 143 ms inter-onset intervals). A passive 3-D motion capture system tracked the movements of 40 markers on the hand, fingers, and the piano keys. The melodies manipulated both the repositioning of certain fingers (thumb and pinkie finger) on the keyboard and the distance between repeating finger movements (three or six tones). Functional data analysis techniques were applied to the analysis of motion trajectories. Data analyses are presented that indicate that the repositioning movement of a finger toward its next goal (keypress) begins sooner at faster tempi than at slower tempi in the x plane (sideways motion on the keyboard). Applications to piano practice will be discussed. [Work supported by the Austrian Science Fund, NSERC, and the Canada Research Chairs program.]
Optical coherence-based imaging and sensing in biomedicine and biotechnology
Authors: Izatt Joseph A.
Published: October 2013

Optical coherence-based imaging techniques including optical coherence tomography (OCT), optical coherence microscopy (OCM), and spectral domain phase microscopy (SDPM) use low-coherence spectral interferometry to obtain nanometer to micron-scale measurements of structure, motion, and molecular composition in living cells, tissues, and organisms. OCT has become a standard diagnostic tool in clinical ophthalmology and is under investigation for other human diagnostic applications including cancer detection and evaluation of cardiovascular disease. Within the past few years, dramatic technology advances have increased the performance of OCT and OCM systems manyfold, and they are now capable of micron-scale two- and three-dimensional functional and molecular imaging noninvasively in living systems. Applications of these new technologies for high-throughput noninvasive phenotyping and rapid 3D imaging in small animals and developmental biology models is particularly compelling. Related technology advances have enabled the design of highly phase-stable interference microscopes capable of resolving nanometer-scale structures and motions in living cells with ms temporal resolution. These new capabilities are being used to probe cellular internal and external surfaces and their responses to chemical and mechanical stimuli. I will review our ongoing work at Duke in these areas. [Work supported by NIH Grants RR019769, EY013516, EB006338, and EY017393.]
Radial reflection diffraction tomography
Authors: Lehman Sean K. , Norton Stephen J.
Published: October 2013

A wave-based tomographic imaging algorithm based upon a single, rotating, radially outward oriented transducer has been developed. At successive angular locations at a fixed radius, the transducer launches a primary field and collects the backscattered field in a ‘‘pitch/catch’’ operation. The hardware configuration, operating mode, and data collection method are identical to that of most medical intravascular ultrasound (IVUS) systems. The goal is to develop a wave-based imaging algorithm using diffraction tomography techniques. Given the hardware configuration and the imaging method, this system is referred to as ‘‘radial reflection diffraction tomography.’’ Two hardware configurations are considered: a multimonostatic mode using a single transducer as described above, and a multistatic mode consisting of a single transmitter and an aperture formed by multiple receivers. The probe is mounted at the end of a catheter or snaking tube inserted into a part or medium with the goal of forming images of the plane perpendicular to the axis of rotation. An analytic expression for the multimonostatic inverse is derived but ultimately a Hilbert space inverse wave algorithm is used to construct images using both operating modes. Applications include improved IVUS imaging, bore hole tomography, and nondestructive evaluation (NDE) of parts with existing access holes.
Speckle reduction of echographic image by transmitting ultrasonic waves of double peak spectrum
Authors: Akiyama Iwaki , Saito Shigemi , Ohya Akihisa
Published: October 2013

Speckle noise of ultrasonic echography is a major issue. To reduce the speckle noise a method of summation after logarithmic compression of higher order harmonic images is proposed. The feasibility is confirmed by the phantom experiments using a mechanical scanner system equipped with a bi-layered structure transducer of PZT and PVDF. The dynamic range of harmonic components higher than the third, however, is not enough for effective speckle reduction. In this study a method of using difference and sum frequency components as well as harmonic components by transmitting pulsed waves of double peak spectrum is proposed. To compare the dynamic range of those components the numerical simulation of transmitting the ultrasonic waves of 2 and 6 MHz and those of 2 and 8 MHz in absorbing media is carried out. An attenuation coefficient of 1.0 dB/cm/MHz is assumed. As a result, complex beam patterns are formed due to degeneration between second harmonic and difference frequency components of 4 MHz in the former case. In the latter case three focal beam patterns of 4-, 6-, and 8-MHz components within 40 dB in dynamic range are formed. Consequently this method has the potential to apply for the biological tissues. [Work supported by a Grant-in-Aid for Scientific Research.]
Subbottom profiling using synthetic aperture processing
Authors: Schock Steven
Published: October 2013

Using synthetic aperture processing, a chirp sonar generates reflection profiles of the seabed. Synthetic aperture processing is implemented by calculating the vertical change in the positions of acoustic transducers between transmission and using the offsets to time shift the acoustic data so that horizontal reflectors are summed coherently. The calculated changes in the positions of the acoustic transducers between transmission events are based on Doppler velocity log and inertial measurement unit measurements of vehicle motion. Imagery generated from reflection profiles of seabeds with various sediment types are used to quantity the improvement in the spatial resolution and the SNR of reflection profiles as a function of the length of the synthetic aperture. [Work supported by Office of Naval Research.]
Influence of nonlinear propagation on calibration of hydrophone sensitivity using a two-transducer reciprocity method
Authors: Yoshioka Masahiro , Sato Sojun
Published: October 2013

This paper describes the phenomenon by which the measured value depends on the ultrasound pressure in the measurement field when the hydrophone sensitivity has been calibrated using the two-transducer reciprocity method. This dependence is attributable to nonlinear ultrasonic propagation. However, the international standard, IEC 60866:1987, ‘‘Characteristics and calibration of hydrophones for operation in the frequency range 0.5 MHz to 15 MHz,’’ provides a correction factor for diffraction and attenuation derived assuming linear propagation of sound. We attempted to derive the correction factor from the ultrasound pressure distribution calculated using the Khokhlov-Zabolotskaya-Kuznetsov (KZK) equation, which is well known as a nonlinear ultrasonic model. Results demonstrated that the dependence of the calibrated sensitivity value on measured sound pressure is quantitatively explainable. The hydrophone sensitivity can be calibrated properly by compensating for the influence of ultrasonic nonlinear propagation using the KZK equation.
Voice quality of artistic expression in Noh: An analysis-synthesis study on source-related parameters
Authors: Kawahara Hideki , Fujimura Osamu , Konpaku Yasuyuki
Published: October 2013

High-quality recordings of Noh vocalizations, produced by one of the authors, were made in an anechoic chamber, comparing vocal styles and emotional expression, using the same sample text from classical Noh. The recorded signals were subject to analysis, synthesis, and morphing by using STRAIGHT. The morphing procedure generated mean parameter settings for comparing two subtly differentiated voicing styles, traditionally called ‘‘wagin’’ soft vocalization, and ‘‘gougin,’’ strong vocalization, to investigate the relation between signal parameters and communicative functions. STRAIGHT extracts three parameters from recorded speech signals: fundamental frequency, smoothed spectrogram, and a time-frequency aperiodicity map. The fundamental frequency and the time-frequency aperiodicity, in other words, source-related parameters, were systematically manipulated to investigate their role in artistic expression.
Physical modeling vocal synthesis
Authors: Howard David , Mullen M. Jack , Murphy Damian
Published: October 2013

Physical modeling music synthesis produces results that are often described as being ‘‘organic’’ or ‘‘warm.’’ A two-dimensional waveguide mesh has been developed for vocal synthesis that models the adult male oral tract. The input is either the LF glottal source model or a user-provided waveform file. The mesh shape, which is based on MRI data of a human vocal tract, can be changed dynamically using an impedance well approach to allow sounds such as diphthongs to be synthesized without clicking. The impedance well approach enables the mesh shape to be varied without removing or adding elements, actions that cause audible clicking in the acoustic output. The system has been implemented as a real-time MIDI-controlled synthesizer, taking its inspiration from the von Kempelin speaking machine, and this will be demonstrated live as part of the presentation. The system is set up to allow a continuous glottal source to be applied that includes vibrato, and thus the real-time output it currently produces is close to being vocalized. It should be noted, though, that appropriate variation of the lip opening does produce a voiced bilabial plosive, demonstrating the potential for moving towards a full speech synthesis system in the future.
Analysis of acoustic features affecting singing-voice perception and its application to singing-voice synthesis from speaking-voice using STRAIGHT
Authors: Saitou Takeshi , Unoki Masashi , Akagi Masato
Published: October 2013

A singing-voice synthesis method that can be transformed from a speaking voice into a singing voice using STRAIGHT is proposed. This method comprises three sections: the F0 control model, spectral sequence control model, and duration control model. These models were constructed by analyzing characteristics of each acoustical feature that affects singing-voice perception through psychoacoustic experiments. The F0 control model generates a singing-voice F0 contour through consideration of four F0 fluctuations: overshoot, vibrato, preparation, and fine (unsteady) fluctuation that affect the naturalness of a singing voice. The spectral sequence control model modifies the speaking-voice spectral shape into a singing-voice spectral shape by controlling a singer’s formant, which is a remarkable peak of a spectral envelope at around 3 kHz, and amplitude modulation of formants synchronized with vibrato. The duration control model stretches the speaking-voice phoneme duration into a singing-voice phoneme duration based on note duration. Results show that the proposed method can synthesize a natural singing voice, whose sound quality resembles that of an actual singing voice.
A psycho-acoustical analysis of vocal effort by opera singers
Authors: Skirlis Konrad , Cabrera Densil , Connolly Anna
Published: October 2013

A singer’s vocal output will be affected by a number of factors such as the size of the auditorium used for the performance. In this paper, the ‘‘large hall’’ and ‘‘small hall’’ voices of singers were explored in terms of the specific loudness pattern, which represents the distribution of loudness in the auditory system. Recordings of seven professional opera singers were made in an anechoic room and in a small recital hall. Singers were asked to imagine they were in a small hall for one set of recordings and in a large hall for another set. Each singer sang the same song under these two ‘‘conditions.’’ The recordings were then analyzed to determine spectral and temporal changes. Spectral analysis was used to calculate the strength of the singers formant under these conditions, and this was compared with the specific loudness patterns. Results help us to understand why the singer’s formant is so important for vocal projection.
Modeling the propagation of volcanic infrasound with a 3-D parabolic equation
Authors: Lingevitch Joseph F. , Edelmann Geoffrey F. , Drob Douglas P. , Collins Michael D.
Published: October 2013

Active volcanoes are significant sources of infrasound in the 1–4-Hz frequency band and are detectable at ranges of hundreds of kilometers. The propagation is affected by the atmospheric temperature, wind, and attenuation profiles, which are range and time dependent. In this paper, a 3-D atmospheric parabolic equation method that accurately incorporates the effects of wind is applied to modeling volcanic infrasound propagation. A parabolic equation is derived using a narrow angle operator approximation in azimuth and a wide angle approximation in depth to account for wind-induced refraction that can trap or disperse sound energy, leading to large variations of propagation with wind conditions. The atmospheric specification for this model is derived from the attenuation coefficients of Sutherland and Bass [J. Acoust. Soc. Am. 115, 1012–1032 (2004)] and the temperature/wind profiles from the Naval Research Laboratory Ground to Space model [J. Geophys. Res. 108, 4680–4691 (2003)], which provides global atmospheric profiles from the ground to 200 km altitude updated four times daily. Signal amplitude and direction of arrival estimates from the model are compared with measurements of volcanic infrasound at listening stations several hundred kilometers from the source. [Work supported by ONR.]
Infrasound wind noise contributions
Authors: Raspet Richard , Yu Jiao , Webster Jeremy
Published: October 2013

In a recent study [J. Acoust. Soc. Am. 119(2), 834–843 (2006)] predictions of various noise contributions were compared to measured wind noise from various devices and windscreens. The fluctuating stagnation pressure, the turbulence-turbulence interaction pressure, and the self-noise contributions in the inertial range were calculated from the measured wind velocity spectrum and compared to measurements. In that study the bare microphone agreed well with the predicted stagnation pressure spectrum, and a heavily windscreened microphone agreed well with the predicted turbulence-turbulence interaction spectrum. In this paper we extend the analysis and the measurements into the source region of the spectra—well into the infrasound range. In addition we predict the spectral contribution due to the turbulence-mean shear interaction. The prediction displays a steep slope in the inertial range and may dominate at low frequencies. [Research supported by the U.S. Army TACOM-ARDEC at Picatinny Arsenal, NJ.]
Elastic response of exotic ferromagnets?
Authors: Keppens Veerle , Luan Yanbing , Bhattacharya Sriparna , Sergienko Ivan , Mandrus David , Sales Brian
Published: October 2013

Resonant ultrasound spectroscopy (RUS) measurements have been carried out as a function of temperature (2–300 K) and magnetic field (0–2 T) for a series of exotic ferromagnetic materials, including Eu8Ga16Ge30 (Tc≂32 K), EuFe4Sb12 (Tc≂82 K), Au4V (Tc≂46 K), and Yb14MnSb11 (Tc≂53 K). The different materials show remarkable differences in their elastic response near the Curie temperature, revealing a sudden softening at Tc in EuFe4Sb12, and a stiffening of the elastic moduli at Tc in Yb14MnSb11. Ongoing attempts to theoretically model the data using Landau theory will be discussed. [Work was supported by the National Science Foundation under Grant No. DMR0506292. Oak Ridge National Laboratory is managed by UT-Batelle, LLC, for the U.S. Department of Energy under Contract No. DE-AC05-00OR22725.]
Measuring elastic constants of novel materials and thin films using small sample resonant ultrasound spectroscopy
Authors: Maynard J. D.
Published: October 2013

One of the most important undertakings for materials is the measurement of the elastic constants. As derivatives of the free energy with respect to atomic displacements, the elastic constants are closely connected to the thermodynamic properties of the material, including the specific heat, the Debye temperature, the Gruneisen parameter, etc. The elastic constants are a sensitive probe of the lattice environment in which all solid state phenomena occur, particularly in the vicinity of a phase transition. Novel materials to which RUS might be applied are often available only in small samples, and for these cases we have developed a small sample RUS apparatus that can measure bulk samples only a few hundred microns in size and thin films a few hundred nanometers in thickness. Thin films are particularly interesting because substrate induced strains in the lattice may produce dramatic changes in the elastic and thermodynamic properties. In this talk, small sample RUS measurements on the new superconductor magnesium diboride and colossal magnetoresistance materials, in bulk and thin film forms, will be discussed.
Acoustical properties of two-dimensional sonic crystals
Authors: Cervera Francisco , Schwarz Elisabeth , Torrent Daniel , Cabrera Suitberto , Sánchez-Dehesa José
Published: October 2013

The transmittance and reflectance spectra of two-dimensional sonic crystals made of hexagonal arrays of cylindrical bars in water is reported. The balance of sound fluxes is analyzed by studying angle-resolved spectra of transmitted and reflected sound for all directions in space. A characterization of the acoustic band structure along the two high symmetry directions in the lattice (i.e., ΓX and ΓJ directions) in a large region of frequencies will be presented. It is shown that, for example, the shape of the stop band features in different types of acoustical spectra is interlinked and must be studied jointly to understand acoustical phenomena in these materials. Moreover, the dependence of acoustical properties as a function of the number of layers is also studied. The experiments are compared with theoretical simulations based on multiple scattering theory. [Work supported by MEC of Spain.]
Design of resonant ultrasound systems for use with very small samples
Authors: Migliori Ben , Migliori Albert , Betts Jon
Published: October 2013

The design of resonant ultrasound (RUS) systems for small samples, whether used in small-bore cryostats at low temperature and high field or at room temperature, presents several unique challenges. These include alignment and containment of the sample, thermal contraction of construction materials, ease of use, and robustness in the presence of large (\?10 T) magnetic fields. Such problems may be resolved through the use of novel (and often easily obtained) materials as well as unique designs. At the National High magnetic Field Laboratory (NHMFL) we have addressed these problems and built a sub-4 K cryostat RUS measurement system capable of reliably measuring samples ‘1 mm in diameter. Examples of such a cryostat systems, designed for conditions ranging from room temperature and zero field to ‘4 K and \?10 T, as well as explanations of the necessary design parameters and recommended materials are presented. Examples of RUS measurements with this system are presented.
Nondiffractive propagation in sonic crystals
Authors: Sanchez-Morcillo Victor J. , Perez-Arjona Isabel , Redondo Javier , Espinosa Victor , Staliunas Kestutis
Published: October 2013

Sonic crystals are media with a periodic modulation of the acoustic parameters, such as the density and the bulk modulus. They have recently attracted a great interest, because of their potential applications in the control of sound propagation, used as reflectors, focusers, or waveguides. All these properties are related with the dispersion introduced by the crystal anisotropy. We report on the nondiffractive propagation of sound in two-dimensional sonic crystals. It is shown that, for given frequencies and directions of incidence, a narrow sonic beam can propagate without diffractive broadening. Such nondiffractive sonic beams exist in crystals with perfect symmetry and do not require the presence of defects, differently from other waveguiding phenomena reported previously. The cancellation of diffraction has been predicted using the plane-wave expansion method to evaluate the dispersion surfaces of the crystal and the spatial dispersion (isofrequency) curves. It occurs for frequencies and wave vectors for which dispersion curves have zero curvature, denoting a transition between focusing and defocusing regimes. By means of perturbative techniques, a simple analytical expression for the nondiffractive conditions has been obtained. The phenomenon is also demonstrated by numerical integration of the acoustic equations using FDTD techniques. [Support from Spanish MEC, Project FIS2005-07931-C03-01/03, is acknowledged.]
Strobo-photoelastic visualization of Lamb-type waves with negative group velocity in a solid/fluid/solid trilayer
Authors: Yamamoto Ken
Published: October 2013

An elastic wave with negative group velocity transports energy in the direction opposite to the phase velocity. This phenomenon exists in certain modes of Lamb waves propagating in a solid plate, but only over a very narrow range of frequency-thickness products and Poisson’s ratios of a plate material. A minimum value of the negative group velocity of a mode is dominated by the Poisson’s ratio of a material. A similar phenomenon also appears in Lamb-type waves that are guided in a symmetric solid/fluid/solid trilayer. Derivation of the dispersion equation of Lamb-type waves emphasizes the existence of the negative group velocity in certain modes. The existence of negative group velocity depends not only on the modes, frequency-thickness product, and Poisson’s ratio, but also on the fluid layer thickness. The magnitude of negative group velocity is controlled by changing the fluid layer thickness. Direct evidence is presented of the negative group velocity that is obtained by means of stroboscopic photoelastic visualization of pulsed Lamb-type waves propagating in a glass/water/glass trilayer. A time-sequence photoelastic image shows the wave-phase behavior and the pulse envelope propagating in the opposite direction.
An MRI-based time-domain speech synthesis system
Authors: Kitamura Tatsuya , Takemoto Hironori , Mokhtari Parham , Hirai Toshio
Published: October 2013

A speech synthesis system was developed based on Maedas method [S. Maeda, Speech Commun. 1, 199–229 (1982)], which simulates acoustic wave propagation in the vocal tract in the time domain. This system has a GUI interface that allows fine control of synthesis parameters and timing. In addition, the piriform fossae were included to the vocal tract model, resulting in antiresonances in speech spectra at the frequency region from 4 to 5 kHz. The system can produce all the Japanese phonemes using vocal tract area-functions (VTAFs) extracted from 3-D cine-MRI obtained during production of VCV or CVCV sequences for a male speaker. The system can be used to synthesize Japanese sentences with high naturalness and intelligibility by concatenating segmental units and controlling the glottal source using the GUI interface. Since a time-varying VTAF is obtained by interpolating between VTAFs, the dataset size of the system is significantly smaller than that of corpus-based speech synthesizers. The speaker-specific VTAFs and inclusion of the piriform fossae permit us to reproduce speaker-specific spectral shapes, not only the lower formants but also higher frequency regions that contribute to the perception of speaker individualities. [Work supported by NICT, SCOPE-R, and Grant-in-And for Scientific Research of Japan.]
Toward hidden Markov model-based spontaneous speech synthesis
Authors: Akagawa Tatsuya , Iwano Koji , Furui Sadaoki
Published: October 2013

This paper investigates several issues of spontaneous speech synthesis. Although state-of-the-art synthesis systems can achieve highly intelligible speech, their naturalness is still low. Therefore, much work must still be done to achieve the goal of synthesizing natural, spontaneous speech. To model spontaneous speech using a limited amount of data, we used an HMM-based speech synthesizer based on three features: cepstral features modeled by HMMs, and duration and fundamental frequency features modeled using Quantification Theory Type I. The models were trained with approximately 17 min of spontaneous lecture speech, from a single speaker, which was extracted from the Corpus of Spontaneous Japanese (CSJ). For comparison, utterances by the same speaker, reading a transcription of the same lecture, were used to train analogous models for read speech. Spontaneity of the synthesized speech was evaluated by subjective pair comparison tests. Results obtained from 18 subjects showed that the preference score for the synthesized spontaneous speech was significantly higher than that for the synthesized read speech. This implies that HMM-based speech synthesis using actual spontaneous utterances for model training is effective at producing natural speech. Additional subjective evaluation tests were also conducted to analyze the effects of individual features on the impression of spontaneity.
Clustering strategies for hidden Markov model-based polyglot synthesis
Authors: Latorre Javier , Iwano Koji , Furui Sadaoki
Published: October 2013

A speaker-adaptable polyglot synthesizer is a system that can synthesize multiple languages with different voices. In our approach, we mix data from multiple speakers in different languages to create an HMM-based multilingual average voice. This average voice is then adapted to a target speaker with a small amount of speech data from that speaker. Using this adapted voice our system is able to synthesize any of the languages for which it was trained, with a voice that mimics that of the target speaker. The main technical difficulty of this approach is how to cluster the acoustic models to create a voice that sounds equal for all the languages. In this paper we analyze different possible strategies for clustering the HMMs and compared them on a perceptual test. The results showed that questions that refer only to the phonetic features and ignore the original language of the sounds produced voices with more foreign accent but that sound clearly more consistent across languages. Surprisingly, this foreign accent affected neither the intelligibility nor the naturalness of the synthetic speech. Therefore for polyglot speech synthesis the overall better performance was obtained by clustering strategies based only on the phonetic features of the sounds.
Concept-to-speech conversion for reply speech generation in a spoken dialogue system for road guidance and its prosodic control
Authors: Yagi Yuji , Takada Seiya , Hirose Keikichi , Minematsu Nobuaki
Published: October 2013

Because of advancements of speech and language processing, a number of spoken dialogue systems have been constructed. However, because most of them adopt existing text-to-speech synthesizers to generate output speech, it is rather difficult to reflect all the linguistic information that is obtained during the reply sentence generation. To resolve this situation, a framework must correctly reflect higher-level linguistic information, such as syntactic structure and discourse information, on the prosody of output speech: concept-to-speech conversion, where reply sentences are generated from information (to be transmitted) and converted into speech in a unified process. A spoken dialogue system for road guidance was constructed, and concept-to-speech conversion was realized in the system. The linguistic information of the generated sentence is handled in tag LISP form to retain the syntactic structures throughout the process. Moreover, a new method of sentence generation from concept was realized with this system: it handles a concept in phrase units and aggregates them to form a sentence. It is tested whether the linguistic information could be reflected properly on the prosody of output speech. Results of listening experiments verified the effectiveness of our proposed method.
Decision-tree-based F0 quantization for hidden Markov model-based speech coding at 100 bit/s
Authors: Itogawa Yoshihiro , Zen Heiga , Nankaku Yoshihiko , Li Akinobu , Tokuda Keiichi
Published: October 2013

A decision-tree-based quantization scheme for a very low bit rate speech coder based on HMMs is described. The encoder carries out HMM-based phoneme recognition and then recognized phonemes, state durations, and F0 sequence are quantized, Huffman coded, and transmitted. In the decoder, sequences of mel-cepstral coefficient vectors and F0’s are generated from the concatenated HMM-using the HMM-based speech synthesis technique. Finally, a speech waveform is synthesized by the MLSA filter using the generated mel-cepstral coefficient and F0 sequences. In the previous system, we train an MSD-VQ codebook for each phoneme for F0 quantization. Although this scheme can quantize F0 sequences efficiently, to achieve a better speech quality, larger codebook sizes are required. It leads to an increase in the bit rate of the system. To avoid this problem, we cluster F0 sequences using phonetic decision trees and then train a codebook for each leaf node. In the encoding and decoding, codebooks to be used can be determined by tracing the decision tree. It allows us to use smaller codebook sizes since the number of codebooks can be augmented without increase in bit rate. A subjective listening test result shows that the proposed scheme improves the quality of coded speech.