bluuebunny/crossref_metadata_2025 huggingface dataset:
When the shear modulus approaches zero: Fluids don’t bend and Scholte leaves the room
Authors: Odom Robert I. , Sylvester Donna L. G. , McHugh Caitlin P.
Published: October 2013

The 4×4 linear system of differential equations describing the propagation of the displacements and tractions in an elastic layered medium becomes singular as the shear modulus of the elastic medium approaches zero. There are a number of approximate ways to handle this singularity in order to impart numerical stability to the computation of the elastic waves in a layered medium. For example, one can impose an irrotational constraint on the displacements or introduce a massive elastic interface (MEI). Both of these ways of handling the weak shear strength are approximate, but avoid the need for singular perturbation theory [Gilbert, 1998]. Scholte waves are interface waves that propagate along the interface between an elastic solid and a fluid. They have nodes near or on the interface and decay exponentially into the bounding media. Scholte waves do not occur at the boundary between fluids. As the shear speed in the bounding elastic medium approaches zero, the Scholte waves disappear from the spectrum. We investigate this disappearance by applying singular perturbation theory to the coupled fluid-elastic system. Among other things, we will address the rate in wave-number space at which the Scholte waves disappear from the spectrum.
Field from a point source above a layered half-space; theory and observations on reflection from the seabed
Authors: Holland Charles W.
Published: October 2013

L. M. Brekovskikh’s book Waves in Layered Media has provided decades of graduate students and researchers alike with a comprehensive and enormously useful reference. One topic from that work, reflection from a point source above a plane-layered medium (the seabed), is discussed here. Both theoretical underpinnings and observations of reflection from a homogeneous halfspace, a transition layer with smoothly varying density and velocity profiles, and discrete layered media are considered for various shallow water sediment fabrics. [Work supported by the Office of Naval Research and NATO Undersea Research Centre.]
FLIP (Floating Instrument Platform): A major Fred Fisher contribution to ocean science
Authors: Spiess Fred N. , Pinkel Robert , Hodgkiss William S. , Hildebrand John A. , D’Spain Gerald L.
Published: October 2013

Frederick H. Fisher, a loyal and zealous member of the Acoustical Society of America, was an imaginative and effective developer of new techniques for research in both laboratory and seagoing acoustics. Most notable among his contributions was his work in bringing into being and enhancing the usefulness of the spar buoy laboratory, FLIP, from its inception in 1960. Not only did Fred use FLIP in his own research, its existence and many of its ancillary capabilities constituted a base for the seagoing research of others. The authors of this paper have benefited from FLIP’s unique capabilities, starting with long-range sound propagation studies in the 1960’s and 1970’s. FLIP’s stability and deep draft structure provided the platform for development of acoustic Doppler techniques for the measurement of ocean currents. Most recently, FLIP has been involved in studies of marine mammal vocalizations and use of multielement arrays to investigate details of shallow-water propagation. Fred’s initial studies of sonar bearing accuracy, for which FLIP’s construction was funded, and his dedication to advancing FLIP’s ability to contribute to ocean science, constitute a legacy that is being utilized today, more than 40 years after FLIP’s launching.
The effects of fluctuating interaural cues on the segregation of speech in rooms
Authors: Brungart Douglas , Iyer Nandini
Published: October 2013

Spatial separation is known to improve the segregation of talkers in anechoic environments, but relatively little is known about the role spatial cues play in speech segregation in reverberant rooms. One might expect the random disruptions in the interaural time and level differences (ILDs and ITDs) that occur in reverberant environments to eliminate many of the intelligibility benefits that normally occur for spatially separated speech. However, the precedence effect often leads to a robust perception of spatial location even in extremely echoic environments. This can result in an apparent separation between talkers that may facilitate selective attention to the location of the target speech even in a highly reverberant room. Also, random fluctuations in ITD and ILD may lead to differences in the apparent source widths of the target and masking sounds, which might serve as a segregation cue. In this talk, we examine the effects fluctuating interaural difference cues have on the segregation of target speech from competing speech or noise. The results suggest that differences in apparent source width can be used to segregate competing speech signals even when the target and masking signals appear to originate from the same direction relative to the listener.
The intelligibility advantage of clear speech in noise and reverberation backgrounds: Effects of speaking rate
Authors: Krause Jean C.
Published: October 2013

Clear speech refers to a speaking style that is significantly more intelligible than conversational speech for a variety of listeners and backgrounds, including roomlike backgrounds such as noise and reverberation [Payton et al., J. Acoust. Soc. Am. 95, 1581–1592 (1994)]. Although typically spoken more slowly than conversational speech [e.g., Uchanski et al., J. Speech Hear. Res. 39, 494–509 (1996)], talkers can produce clear speech at normal rates with training [Krause and Braida, J. Acoust. Soc. Am. 112, 2165–2172 (2002)]. Whether this form of clear speech (clear/normal speech) provides similar intelligibility benefits to clear (clear/slow) speech in all situations has not been fully characterized, though some results [Krause and Braida, Iranian Audiol. 2, 39–47 (2003)] suggest that the amount of benefit is more dependent on talker and environment. To investigate this possibility for roomlike degradations, the intelligibility advantage provided by clear/slow and clear/normal speech was evaluated by groups of eight normal-hearing listeners at five signal-to-noise ratios (−4, −2, 0, +2, and +4 dB) and three levels of reverberation (0, 0.18, and 0.6 s). Results for four talkers will be reviewed in order to illustrate the effects of speaking rate and speaking style on speech intelligibility in rooms.
Development of networked weatherproof remote sensing embedded Linux system for bioacoustical evaluation
Authors: Kobayashi Hiroki , Kawasaki Yoshihiro , Watanabe Shinichi
Published: October 2013

The purpose of this research is to develop a real-time evaluation technology of bioacoustic diversity in a remote location by networked Linux embedded system. The hardware is designed to operate 24 h a day, 7 days a week, 365 days a year under severe environment, such as extremely high moisture and temperature environment in subtropical forest. Three stages of the hardware development were completed at this point. At first, a wired long-lived bioacoustic monitoring system in subtropical forest in Iriomote Island, Japan was developed and succeeded to operate continuously for the past 2 years based on an unmanned condition with less than 1000 US dollars annual budget. Based on this development, a wired multifunctional monitoring system with infrared cameras and microclimate monitoring sensors was developed and evaluated with less than 4000 US dollars. At last, with FOMA (Freedom of Mobile Multimedia Access) mobile information technology offered by Japanese mobile phone operator NTT DoCoMo, a multifunctional bioacoustic monitoring system with two-way communication system which operates passive and active observation for Iriomote Cat was developed with less than 8000 US dollars. [Work supported by NTT DoCoMo.]
Perception of budgerigar (Melopsittacus undulatus) warble elements
Authors: Tu Hsiao-Wei , Dooling Robert J.
Published: October 2013

The warble song of budgerigar (Melopsittacus undulatus) contains various classes of vocal elements, including contact calls, alarm calls, harmonic calls, etc. Previous work has shown that budgerigars can discriminate different call types according to both acoustic and functional features; however, there is no evidence on how well birds can distinguish subtle differences within and between each category. Using operant conditioning and psychophysical methods, budgerigars were trained to discriminate among different warble elements. In general, birds were able to discriminate quite easily between all warble elements, even those with high spectral cross correlations. Humans, on the other hand, had difficulty distinguishing among warble elements that were similar to one another. The relative discriminability of warble elements within and across categories is used to more precisely define the functional categories of warble elements in understanding vocal communication in this species. [Work supported by NIDCD DC000198 to R.J.D. and P30 DC004664.]
Comparative analysis of head-banging behavior in the subterranean termites Coptotermes formosanus and Reticulitermes flavipes using high-speed imaging
Authors: Fink Tom , Gui Lichuan , Wang Yong , Cao Zhonghua , Jaiswal Adarsh , Tahaineh Orwa , Hasse Roger , Seiner John , Lax Alan
Published: October 2013

Disturbed soldiers of Coptotermes formosanus (FST) readily head-bang (HB) only in their galleries and carton nests while Reticulitermes flavipes (Rf) will also readily HB in an artificial chamber. The head-banging motion was examined using the Photron FASTCAM-ultima APX video camera at 10000 frames per second. Termites were placed in a 0.25-in. diameter hole in a wood block and induced to HB by touching them with a small brush. A PCB Piezotronics 352B accelerometer was used to detect HB-induced vibrations. These vibration signals were then received by the Photron Multi-Channel-Data-Link apparatus, which then superimposes the vibration waveform on the video frames corresponding to the head movement. Thus, movement of the head can be correlated with the vibrations produced. Time between successive HB is about 70 ms or more in FST vs 40 ms or less in Rf despite the much larger head of Rf. Upward and downward head velocities in FST are, respectively, 100 and 240 mm/s, while the respective values for Reticulitermes are over 200 and about 400 mm/s. Acceleration for FST is about 400 m/s vs up to 700 m/s2 for Reticulitermes. Both species may show head rebounding after the initial contact with the substratum.
Optical imaging of neural activity in the right and left guinea pig auditory cortices evoked by repeated sound
Authors: Hosokawa Yutaka , Horikawa Junsei
Published: October 2013

Processing of spectral and temporal information in the right and left auditory cortices of guinea pigs anesthetized with ketamine and xylazine was investigated using an optical imaging technique with a voltage-sensitive dye (RH795). Tonotopical organization in the primary (AI) and the dorsocaudal (DC) fields in the left and the right auditory cortices were compared, as visualized by superimposition of the areas activated by tones at 2, 4, 8, and 16 kHz. In 65% of 31 animals, the distance between 2- and 16-kHz isofrequency bands was longer in the left than in the right AI; it was shorter in the left than in the right DC. Repetition rate transfer functions (RRTFs) in AI measured with repeated sounds (420 Hz repetition rates) were low pass, with a sharp drop-off above 10 Hz. In contrast, RRTFs in DC were bandpass with a peak at 8 or 10 Hz. The cutoff frequencies of RRTF were different for each of the left and right cortices.
The effects of source distance and orientation on lateral line stimulation patterns and source localization by Lake Michigan mottled sculpin
Authors: Coombs Sheryl , Patton Paul
Published: October 2013

Visually deprived, Lake Michigan mottled sculpin respond to small dipole sources (sinusoidally vibrating spheres) with an initial orienting response and subsequent approach and strike behaviors. Lateral line canal neuromasts, but not superficial neuromasts, are required for the initial orienting response. Computational (potential flow) models reveal that stimulation patterns across lateral line canals vary in complex ways when the orientation (axis of vibration) and distance of the dipole source are varied. The body location where lateral line stimulation is maximal (the excitatory peak) might seem to provide a reliable cue of source location. However, this cue is (1) confounded by the relative amplitude, number, and body location of excitatory peaks when source orientation is varied and (2) degraded by both an increase in peak width and a reduction in peak amplitude when source distance is increased. Behavioral measures of orienting accuracy for equally detectable sources at two distances (3 and 6 cm) and horizontal orientations (parallel or orthogonal to the long axis of the fish) reveal that localization abilities are degraded by increases in source distance. Although varying source orientation does not produce dramatic deficits in orienting accuracy, more subtle effects were observed that may relate to information encoded in spatial excitation patterns.
Thermal effect of microbubbles in the focused ultrasound field
Authors: Kaneko Yukio , Iida Naoyuki , Takagi Shu , Matsumoto Yoichiro
Published: October 2013

High-intensity focused ultrasound (HIFU) has been developed for the treatment of tumor, and additionally the medical applications with microbubbles such as ultrasound imaging have attracted much attention. In the field of the bubble dynamics, it is known that the microbubble is an energy converter of ultrasound mechanical energy to heat when subjected to an acoustic field. The goal is the tissue heating enhanced by microbubbles. In this study, the relationship between the heating effect and the behavior of microbubbles is analyzed. The temperature rise was measured by a thermocouple and a thermal liquid crystal sheet, and the bubble behavior was simultaneously observed with a camera. The ultrasound frequency is 2.2 MHz and the intensity is 100–2000 W/cm2. The temperature rise became larger as the number density of microbubbles (Levovist) around the focal region increased. As the number density became too large, the conversion efficiency from ultrasound energy to heat became smaller because the shielding effect of ultrasound by the existence of too many bubbles became larger. From these results, it is indicated that it is important to choose the optimal conditions of microbubbles and ultrasound in order to control the heat deposition from microbubbles.
The use of homemade microbubbles in detection of organ bleeding and enhancement of high-intensity focused ultrasound therapy
Authors: Luo Wenbo , Zderic Vesna , Vaezy Shahram
Published: October 2013

We investigated the feasibility of using homemade microbubbles in detection of internal bleeding, and for optimization of high-intensity focused ultrasound (HIFU) therapy. Homemade microbubbles were produced by sonicating albumin and dextrose saline solution with the presence of perfluorocarbon gas. Microbubbles were injected intravenously to reveal rabbit kidney and liver injuries. HIFU treatments were performed on liver lacerations for hemostatis and on intact liver for lesion production, with or without homemade microbubbles. The mean size of homemade microbubble was 4.7 um (range 1.2–7.1 μm). The concentration of microbubbles was 3.1*108 bubble/ml. Kidney and liver injuries were revealed by mosaic color flow at the injury sites. In treatment, the average hemostasis times normalized with the bleeding rate were 39 s/ml*min for HIFU only and 21 s/ml*min for HIFU+microbubbles treatments. The presence of microbubbles reduced 44 percent (p<0.05) time in producing the first formation of the blood coagulum. The HIFU lesion depth was reduced by 35 percent because microbubbles shield the postfocal region from HIFU exposure. Homemade microbubbles oscillate and collapse in ultrasound field, generating mosaic pattern thus revealing the bleeding site. It also shows potential in promoting HIFU hemostasis and shielding deep-tissue regions from HIFU exposures.
Finite-element modeling of thin-film damping in micromachined microphones employing diffraction based optical readout
Authors: Hall Neal A. , Okandan Murat , Degertekin F.L.
Published: October 2013

Micromachined microphones with diffraction-based optical displacement detection have been presented in detail previously [J. Acoust. Soc. Am. 118(5), 3000–3009 (2005)]. In addition to providing superior diaphragm displacement detection sensitivity over miniature capacitive sensors, optical approaches have the advantage of removing mechanical design constraints on the microphone backplate perforation architecture as well as constraints on the diaphragm-backplate gap thickness. Taking advantage of this freedom to tailor design the frequency response function and thermal noise characteristics of miniature high-performance microphones requires a sophisticated damping model to navigate this new design space. A finite-element model in ansys based on the modal projection method is employed to study the dynamics of new optical microphone structures. The model extracts the frequency-dependent resistance and stiffening characteristics of the film using modal displacement profiles of the diaphragm. Simulated frequency response functions and thermal-mechanical noise limits agree well with those measured on fabricated structures. Most notably, 1.5-mm-diameter diaphragm structures with under 1 μPa/√Hz thermal noise and over 20-kHz bandwidth have been successfully designed, fabricated, and characterized. [The authors would like acknowledge the IC Postdoctoral Research Fellowship Program.]
Design optimization of a microelectromechanical piezoresisitive microphone for use in aeroacoustic measurements
Authors: Homeijer Brian , Griffin Ben , Nishida Toshi , Cattafesta Lou , Sheplak Mark
Published: October 2013

A microelectromechanical systems (MEMS)-based piezoresistive microphone optimimum design is presented, with a focus on improving the minimum detectable pressure over many current technologies without sacrificing bandwidth. This microphone design addresses many of the problems associated with previous piezoresistive microphones. Here, a novel nonlinear circular composite plate mechanic model was employed to determine the stresses in the diaphragm, which was designed to be in the compressive quasibuckled state. With this model, the inherent in-plane stresses that occur in the microelectronic fabrication process can be used to increase the sensitivity of the device. Ion-implanted doped silicon was chosen for the piezoresistors and a fabrication recipe was made which minimizes the inherent noise characteristics of the material. The piezoresistors are arranged in a Wheatstone bridge configuration with two resistors oriented for tangential current flow and two for radial current flow. A lumped element model was created to describe the dynamic characteristics of the microphone diaphragm and the cavity/vent structure. The geometry for this device was optimized using a sequential quadratic programming scheme performed using the aforementioned novel device characteristics. Results indicate a dynamic range in excess of 120 dB for devices possessing resonant frequencies beyond 120 kHz.
Ingoing and outgoing waves in time-domain simulation of wind instruments
Authors: Ducasse Eric
Published: October 2013

An extension of a recently published work [Ducasse, J. Acoust. Soc. Am. 112, 3031–3041 (2002)] is presented. Two-port modeling is an efficient tool for time-domain simulation of sound propagation in wind instruments. It requires the separation of ingoing and outgoing waves contributing to the acoustic field in any cross section of the air column. This separation is achieved by using a multimodal approach, where uncoupled traveling modes appear in the case of cylindrical waveguides: the propagation of the plane mode is straightforwardly modeled by a delay line and the impulsional response of any other mode is the sum of a delayed Dirac pulse and a closed-form dispersive term. Although mode coupling occurs in acoustic pipes with nonuniform cross section, the scattering matrix can be directly obtained. When a multimodal approach is not adequate, the time-domain responses of complex elements are numerically calculated by introducing either nonreflecting boundary conditions (NRBCs) or perfectly matched layers (PMLs) in a finite-element method. The examples of (i) the reflection matrix of a bell, as seen from the body, and (ii) a piece of pipe containing an obstacle (e.g., the register hole) illustrate the method.
Motion of brass players’ lips during very loud playing
Authors: Bromage Seona , Campbell Murray , Stevenson Samuel , Chick John , Gilbert Joël
Published: October 2013

When a crescendo is played on a trumpet, horn, or trombone, the timbre changes markedly at the fortissimo level: there is a dramatic increase in the relative amplitudes of the high- frequency harmonics, and the resulting sound is often described as ‘‘brassy.’’ Shock-wave generation in the air column of the instrument as a consequence of nonlinear steepening of the wavefront is the generally accepted explanation of this phenomenon. It has, however, also been suggested that saturation of the lip opening could play a role in the timbre change. This study compares the motion of the lips at lower dynamic levels and during extremely loud playing, using a high-speed digital camera and instruments with different sizes of mouthpiece. Curves illustrating the variation of the open lip area as a function of time at different dynamic levels are presented, together with waveforms and spectral analyses of both the radiated sound and the pressure in the mouthpiece. The results suggest that there is no qualitative change in the time-dependent behaviour of the opening area of the lips when the instrument enters the brassy regime.
Control parameters inversion using genetic algorithms applied to numerical impedance synthesis for woodwinds
Authors: Perichon Laura , Carriere Olivier , Hermand Jean-Pierre , Meyer Matthias , Guillemain Philippe
Published: October 2013

Woodwind sound synthesis in real time has been achieved using different computational techniques, especially the so-called numerical impedance model based only on acoustical variables. The impedances of the excitation mechanism and of the resonator are approximated by IIR filters and related by a nonlinear coupling. The control parameters can be reduced to a number of three including the length of the resonator, the mouth pressure, and a mask parameter. In this paper we present preliminary inversion results for the latter two parameters in the case of a clarinet. The inversion algorithm is based on differential evolution metaheuristics with a number of prior conditions on the correlated parameters. A cost function based on time-domain correlation has been implemented, allowing us to obtain better results in terms of signal matching than with conventional descriptors extraction. The technique has proved its effectiveness on the sustained part of synthesized sounds and a regularization method is presented to allow us to invert attack and decay parts as well as real analogically recorded signals. Results will be of main interest for tuning control parameters in real-time model-based synthesis.
Quiet heights and pulse tails: The modal structure of near ground to near ground sound propagation in the nocturnal duct
Authors: Waxler Roger
Published: October 2013

On clear nights over land a temperature inversion forms, creating an acoustic duct in the first few hundred meters of the atmosphere. Assuming a flat, locally reacting ground surface and a stratified atmosphere, the sound field can be expressed as a superposition of independently propagating vertical modes [J. Acoust. Soc. Am. 112 (6), 2540–2552 (2002) and 115 (4), 1437–1448 (2004)]. The modes account for refraction due to sound-speed gradients as well as for the compliance and resistance of the ground. For typical nocturnal ducts and ground surfaces one finds two types of modes: a surface mode and a set of higher modes. The surface mode is more slowly propagating and more rapidly attenuated than the higher modes. The higher modes’ amplitudes all have a sharp minimum several meters above the ground. Two consequences of this modal structure have been observed: At long ranges, narrow-band signals have a range-independent quiet height at which sound levels are reduced by 10 to 15 dB from the levels on the ground [J. Acoust. Soc. Am. 119 (1) 86–95 (2006)] while impulsive signals develop a bandlimited low-frequency tail [J. Acoust. Soc. Am. 88 (1), 455–461 (1990)].
The use of subsegmental information in sentence comprehension with or without formant transitions by normal-hearing and hearing-impaired listeners
Authors: Lee Jae Hee , Kewley-Port Diane
Published: October 2013

Using a noise replacement paradigm, Lee and Kewley-Port [J. Acoust. Soc. Am., 119, (2006)] examined sentence comprehension by young normal-hearing (YNH) and elderly hearing-impaired (EHI) listeners when sentences were processed to present only subsegmental information (i.e., either steady-state or formant transitions) in sentences processed in four different ways. Results showed that correct word responses by EHI listeners were more affected by the type of information in the four conditions compared to YNH listeners. To compare the use of subsegmental information between listener groups who were carefully matched for audibility and age, new analyses of correctly identified phonemes and different word-error patterns were made for both correct and incorrect word responses. Correlation analyses examined the relations among correct and incorrect word responses, phoneme scoring, hearing thresholds, and age. Despite a larger number of incorrect answers by EHI listeners, word-error patterns, as well as an ability to extract phonemes using subsegmental information, were similar between YNH and EHI listeners. Analyses for EHI listeners showed correct word responses were strongly correlated with correct phoneme identification and with hearing thresholds, but not with age. [Work supported by NIHDCD-02229.]
Bayesian inversion of propagation and reverberation data
Authors: Nielsen Peter L. , Dosso Stan E.
Published: October 2013

A Bayesian matched-field inversion approach to infer geoacoustic and scattering properties of the seabed is applied to simulated propagation and reverberation data received on a towed horizontal array. The approach is based on the method of fast Gibbs sampling (FGS) of the posterior probability density to estimate uncertainties in both geoacoustic and scattering parameters for broadband acoustic data in realistic shallow-water environments. The FGS is linked to an acoustic propagation model that simultaneously provides complex acoustic pressure at short propagation ranges and long-range reverberation intensity. The inversion algorithm is initially applied to long-range reverberation data alone to assess the geoacoustic information content of reverberation in terms of marginal posteriori probability densities for the environmental parameters. A reduction in uncertainty for the extracted geoacoustic and scattering parameters is demonstrated by a simultaneous inversion of the propagation and reverberation horizontal array data.
Experimental investigations of the influence of room acoustics on the teacher’s voice
Authors: Kob Malte , Kamprolf Anja , Neuschaefer-Rube Christiane , Goldschmidt Oliver , Behler Gottfried
Published: October 2013

Teachers belong to the group of professional voice users who often suffer from voice disorders. One reason for a significantly increased prevalence of voice problems can be poor room acoustical conditions in the classrooms. In this study, four rather reverberant and loud classrooms in a primary school in Aachen were analyzed using measurements of the reverberation time, T20, and the speech transmission index, STI. About half of the school’s teachers were investigated with respect to their voice status by using phoniatric, logopedic, and objective voice analysis methods. The prevalence of voice problems in this group was found to exceed previous studies where subjective voice quality was rated. In a second part of this joint project the change of voice quality during the teachers’ working day was analyzed. Two of the four rooms were acoustically optimized. Members of two groups of teachers with and without voice problems were recorded before and after teaching in either one of the acoustically poor rooms or one of the newly renovated rooms. The preliminary results indicate changes of the voice quality in most subjects with respect to one or more voice parameters. Further studies shall prove the significance of the room influence.
Auscultation in high-noise environments using hybrid electromechanical and ultrasound-Doppler techniques
Authors: Houtsma Adrianus J. M. , Curry Ian P. , Sewell John M. , Bernard William N.
Published: October 2013

Medical evacuation of battlefield casualties or traffic accident victims by air typically takes place in very noisy environments. Auscultation of patients, e.g., to support chest intubation or to detect a pneumothorax, is therefore difficult or impossible to perform. A conventional acoustic stethoscope will not function very well in background noise levels beyond 80 dB. Electronic stethoscopes, in combination with mechanical impedance-matched transducer designs, can extend this range to about 90 dB, but this is not enough for helicopter noise levels that can reach 110 dB. The use of an ultrasound transmitter and receiver, however, provides an essentially noise-free auscultation channel since transportation vehicles do not produce acoustic energy at ultrasound carrier frequencies of 2–3 MHz. Clean and noise-free heart and breath sounds have been obtained in broadband noise fields of intensities as high as 120 dB. A hybrid stethoscope has been developed that allows auscultation by ultrasound-Doppler as well as electromechanical means. Pros and cons of making Doppler sounds subjectively similar to conventional sounds by nonlinear signal processing will be discussed, as well as potentially functional and meaningful aspects of Doppler signals that are not found in conventional stethoscope sounds.
Color and pulsed Doppler signatures of vascular bleeding
Authors: Luo Wenbo , Zderic Vesna , Vaezy Shahram
Published: October 2013

To develop ultrasound-based methods for detection and localization of internal vascular bleeding, we investigated the bleeding signatures using color and pulsed Doppler. Under ultrasound guidance, femoral arteries (n=13) and veins (n=12) in seven pigs were catheterized with 6, 9, and 11 F catheters. Color and pulsed Doppler were acquired from the injury site and the surrounding positions after withdrawal of the catheter. The bleeding rates were 3.8–38 ml/s. Color Doppler revealed that the injured vessels were in spasm. The checkered color pattern at the injury site indicated flow turbulence. Pulsed Doppler spectrum of the color jet (extravasated blood) represented a unique arterial bleeding pattern with increased baseline velocity (from 0 up to 25.8 cm/s) and elevated systolic velocity (from 23.2 up to 62.5 cm/s) at the puncture site, as compared to the normal arterial pattern. Venous bleeding showed a narrow-band vibration signal at the bleeding site. Local blood turbulence contained multiple velocity components which were illustrated as checkered color in color Doppler. Pulsed Doppler spectrum showed increased baseline due to turbulence and elevated velocity due to narrowed vessel. In future studies pulsed Doppler radio-frequency data will be used to analyze bleeding patterns for detection and localization of internal bleeding.
A hybrid approach for simulating clarinetlike systems involving the lattice Boltzmann method and a finite difference scheme
Authors: da Silva Andrey Ricardo , Scavone Gary
Published: October 2013

The lattice Boltzmann (LB) method is becoming an increasingly important technique for the simulation of problems involving the interaction between acoustic and fluid fields. New LB algorithms allow the representation of moving boundaries at low Mach numbers. In this work we present the implementation of a hybrid model involving one such LB algorithm coupled to a finite difference scheme to simulate the fluid-structure interaction at the reed of a two-dimensional clarinetlike system. The results of the simulation for different input pressures and reed parameters are compared with the results provided by the related literature. [The first author was supported by CAPES (Brazil).]
Vortex sound and the flute
Authors: Bamberger Andreas
Published: October 2013

The flute is investigated for its aeroacoustical properties based on the vortex sound theory. Particle image velocimetry (PIV) is used for quantitative flow determination of the jet-edge interaction. The Endoscopic-PIV offers a nonobstructive view of the system over all phases. The evaluation of the source term through the vorticity is done according to M. Howe (1975). The acoustic flow across the embouchure is determined by excitation through the foot. The flute is operated near 1200 Hz with various jet speeds. Finally the acoustic radiation power in the far field is determined to be compared with the source terms. The following findings are presented: (i) The space integrated and time averaged power of the coherent source terms turns out to be positive, i.e., emitting acoustic energy. (ii) There is a dominant contribution near the labium. (iii) The source term power compares favorably with the far field power, which is well below 1% of the total input power. The results demonstrate that the vorticity of the upper and lower shear layers of the jet cancels to a great extent, except for regions where the transverse amplitude of the jet is large.
Synthetic percussion
Authors: Bilbao Stefan
Published: October 2013

The widespread availability of fast small computers has removed many of the restrictions that governed the development of sound synthesis methods in the past, and, in particular, those based on physical modeling of musical instruments. Even relatively complex models can now produce sound output in something approaching real time, if not real time itself. As an example, various models of strings and 2-D percussion instruments, simulated using direct time domain methods, are presented here. Various issues are discussed, including interconnections of various vibrating objects, extensions of percussion instruments involving acoustic ‘‘preparation,’’ the modeling of nonlinearities, multichannel output, algorithmic complexity, and operation counts, as well as current work towards developing real time implementations. Sound examples are presented.
Physical modeling of the piano
Authors: Giordano N.
Published: October 2013

As part of an ongoing project we have constructed a computational model of the piano. This model uses Newton’s laws to compute the motion of the hammers, strings, soundboard, and room air, yielding the sound pressure that reaches a listener. The model is based heavily on corresponding experimental studies of hammers, strings, and soundboard properties. To date we have focused on the modern piano, but our basic modeling approach can also be used to investigate how changes in the instrument would change the resulting tones. After reviewing the model, we describe studies of the sound produced by models of historical pianos, i.e., the type of piano made in the late 1700s or early 1800s, and compare those tones with the tone of a modern instrument.
Numerical dispersion of finite difference method
Authors: Takasawa Yoshimitsu
Published: October 2013

The partial frequencies of the transverse vibrations of a string, which has stiffness ε and is assumed to be hinged at both ends, form an inharmonic series and obey the equation fn=f0n(1+π2εn2)1/2, where f0 is the fundamental frequency. For musical synthesis, the equations of a physical model can be simulated in the time domain using numerical methods. The most straightforward approach can be said to be the finite difference method. The equations are formulated in discrete form using discrete positions iΔx and discrete time steps nΔt. In the finite difference method, two problems must be taken into account. The first is numerical stability: it must be roughly γ>N. Here, γ=fe/2f0, fe=1/Δt is the sampling frequency, and N=L/Δx is the segment number of the string. The second is the numerical dispersion. It has been shown that some undesirable dispersive effects might be present in the solution if a finite difference scheme is used. This numerical dispersion can be formularized as the equation fn≊f0n[1+π2{ε−1/12(1/N2)−1/γ2)}n2−π4/6{ ε/N2−1/60(1/N4−1/γ4)}n4]1/2,n≤N−1.
Experimental determination of the correction for noise levels assessed in front of a facade
Authors: Memoli Gianluca , Kephalopoulos Stylianos , Paviotti Marco , Licitra Gaetano
Published: October 2013

The assessment of noise levels, in the proximity of a building or on its facade, is a requirement of the European Environmental Noise Directive 2002/49/EC concerning environmental noise produced by road and railway traffic, airports, and industries. One of the problems that concerns both theoretical models and measurements is to estimate the correction due to reflections. Several works in the past have approached this problem, now stated by international standards, which suggest using a +3 or +6-dB correction in function of the microphone position. A complete set of measurements have been performed in the framework of the IMAGINE project for road traffic, in different urban situations, in order to determine the real intereference pattern in terms of coherence length. A careful treatment of the acquired data highlighted a dependence of corrective factors on the distance from the building facade that cannot be neglected. Practical solutions and uncertainties also will be discussed in this work.
The role of noise audits in characterizing and reducing consumer product noise
Authors: Bowen David L.
Published: October 2013

Consumer products often share some common traits that impact both the measurement techniques that can be employed, as well as the noise control options that are available. These traits include their relatively small size (except for the special case of automobiles), low profit margin, and quick time to market. Their small size implies that existing units or prototypes can usually be set up for acoustical measurements inside a modestly sized reverberation room. The low profit margin implies that design solutions, rather than adding ‘‘acoustical’’ materials, are the preferred route to a quieter product, but the quick time to market can sometimes work against this approach. This talk will focus on a sound power based ‘‘noise audit’’ technique that is useful for characterizing and rank ordering the potential noise sources in a product, in order to devise the most effective noise reduction strategies. Case studies will be used to illustrate the use of this engineering technique applied to the particular case of consumer products, where the design trend is often towards lighter, stiffer components and faster rotating parts, all of which tend to increase noise.
Evaluation of noise intensity level using an eccentric press during the production of metallic closure: Findings according to different positions surrounding the press
Authors: Zannin Paulo H. T. , Ansay Samuel S.
Published: October 2013

This study was performed to evaluate the differences in noise level produced by an eccentric press, during the manufacturing of metallic clo- sures for steel cans. The primary objective of the study was to establish a rank of sound-intensity level and sound-power level as well as the global noise level emitted by this particular press. Ten different areas in the surface surrounding the press were evaluated regarding both the sound power and the sound intensity of each measurement. Each site was measured twice to increase accuracy. The ten areas sampled were sweeping using intensity probe. We have found the surface area to demonstrate the lowest noise level was at S1 and the highest one was at S10 (p value 0.0005). In conclusion, our study demonstrates adequately the global noise level as well as in specific sites, indicating areas requiring special treatment for noise reduction in this environment.
Noncontact acoustic manipulation using a standing wave field in air
Authors: Kozuka Teruyuki , Yasui Kyuichi , Tuziuti Toru , Towata Atsuya , Iida Yasuo
Published: October 2013

A noncontact manipulation technique is necessary to develop micro-machine technology, biotechnology, and new materials processing. The authors have developed an acoustic manipulation technique to transport particles in water [T. Kozuka, Proceedings of WCU2003, Paris, 483–486 (2003)]. Although it is more difficult to generate a strong sound field in air than in a liquid, many researchers are actively studying the trapping of particles, droplets, and aerosols in air using ultrasound. Nevertheless, many of the resultant studies only trap and observe objects; they do not transport them. This paper describes an advanced manipulation technique to transport small objects in air using that scheme. A standing-wave field was generated by two sound beams (40 kHz) using bolted Langevin transducers. The beams’ axes cross each other. Expanded polystyrene chips were trapped at nodes of the sound pressure in the crossing sound beams. The trapped position was shifted by changing the phase difference of the two sound beams. In addition, transportation at constant speed of the trapped target was realized by a slight difference in ultrasonic frequency between the two sound beams. This system is also applicable to liquid droplets and aerosols in air.
Direct computational method of including piriform fossae and nasal cavity in a time-domain acoustic model of the vocal tract
Authors: Mokhtari Parham
Published: October 2013

Frequency-domain simulations of the human vocal tract (VT) have previously shown the importance of including the piriform fossae, which impart a pole and two zeros in the 4–5-kHz frequency range and thereby contribute to speaker individualities. The literature has also shown that time-domain simulation of VT acoustics can result in high-quality synthesis naturally including interactions between the time-varying glottal area and the supraglottal VT. In the present work, the time-domain model of [S.Maeda, Speech Commun. 1, 199–229 (1982)] was extended to include both left and right piriform fossae as side-branches connected to the main VT, in addition to the nasal tract and sinuses. Departing from Maeda’s original implementation owing to the complexity of including more than one side branch, the variables representing acoustic pressure and volume velocity at the piriform fossae and nasal tract junctions were analytically eliminated, and the resulting large system of linear equations were solved simultaneously at each simulation sample. This direct method runs at only a few times real-time on a 1.8-GHz notebook PC, while achieving a more natural sound quality in speech synthesis and control over timbral (or voice quality) features that contribute to each speaker’s individuality. [Work supported by NiCT and SCOPE-R of Japan.]
Measures of intraoral pressure pulse shape during stop consonants
Authors: Lucero Jorge C. , Koenig Laura L.
Published: October 2013

To maintain voicing during a stop consonant, adult speakers typically perform articulatory actions that increase supraglottal volumes and slow the rate at which of intraoral pressure (Pio) rises. In an influential study, Müller and Brown [in Speech and Language: Advances in Basic Research and Practice 4, edited by N. Lass (Academic, New York, 1980), pp. 318–389] proposed a quantitative measure of Pio pulse shape, based on the difference between (a) the slope from pressure at closure to average pressure and (b) from average pressure to pressure at release. We recently applied these techniques to data from women and child speakers of English and observed some shortcomings in Müller and Brown’s measurement technique, such that the slope difference angle did not clearly reflect the actual pressure pulse shape. The purpose of this paper is to explore new methods of analyzing intraoral pressure contours, based on fitting a mathematical model to the pressure pulse shapes. It will report our progress and compare our results with the previous technique.
Comparison of vocal tract shaping patterns derived from articulatory fleshpoint data and MRI-based area functions
Authors: Story Brad H. , Bunton Kate
Published: October 2013

A recent addition to the University of Wisconsin x-ray microbeam (XRMB) database [Westbury, UW-Madison, 1994] (XRMB) is a speaker from whom MRI-based vocal tract area functions have been previously obtained. The purpose of this study was to compare principal components derived from collections of vocal tract shapes for vowels obtained with these two different techniques. For the XRMB data, midsagittal cross-distance functions representative of each of 11 vowels were first obtained from tongue and jaw pellet positions and the hard palate profile of the speaker. A principal components analysis was then performed on the set of cross-distance functions. Results indicated that the first two orthogonal components (referred to as modes) accounted for more than 90% of the variance. The shape of these two modes was shown to be similar to those previously determined from the same speakers set of MRI-based area functions. The XRMB-based modes were then used to develop a database-lookup technique where mode coefficients could be determined for time frames throughout a series of vowel sequences. Finally, a procedure is described for using the time-dependent mode coefficients obtained from the XRMB data as input for an area function model of the vocal tract. [Work supported by NIH R01-DC04789.]
A study of manual articulatory feature-based transcription of conversational speech
Authors: Livescu Karen , Chi Xuemin , Lavoie Lisa , Bezman Ari , Borges Nash , Yung Lisa
Published: October 2013

This study investigates the manual labeling of speech, and in particular conversational speech, at the articulatory feature level. A detailed transcription, including subtleties such as overlapping or reduced gestures, is useful for studying the great pronunciation variability in conversational speech. This type of labeling also facilitates the testing of automatic feature classifiers, such as those used in articulatory approaches to automatic speech recognition. For this study, approximately 100 utterances drawn from the switchboard database have been transcribed using eight articulatory tiers rather than the traditional single phonetic tier. The tiers include: place and degree for up to two constrictions, nasality, glottal state, lip rounding, and vowel quality. Two transcribers have labeled this set of utterances in a multipass strategy, allowing for correction of errors. Preliminary analysis shows a high degree of intertranscriber agreement. Further analysis of the data is being performed to address a number of questions, such as: How quickly and reliably can this type of transcription be done? What are its advantages and disadvantages relative to purely phone-based transcription? What characteristics of the utterances correspond to high or low transcriber agreement? What can be learned from the data regarding articulatory phenomena such as reduction and asynchrony?