bluuebunny/crossref_metadata_2025 huggingface dataset:
Dual-pulse lithotripter accelerates stone comminution and reduces cell injury in vitro
Authors: Sokolov Dahlia L. , Bailey Michael R. , Crum Lawrence A.
Published: October 2013

Peak acoustic pressures and cavitation generated in shock wave lithotripsy (SWL) appear to contribute to both desired stone comminution and undesired injury to surrounding renal tissue. Our dual pulse system, comprised of two opposing, confocal lithotripters and generating simultaneous, converging shock pulses, localizes and intensifies the peak pressures and cavitation. Comparison of cavitation damage to aluminum foil shows an 8-cm stripe of pits produced by a single-pulse lithotripter and a 1-cm stripe of deep pits produced by the dual-pulse lithotripter. 100 dual pulses generated at 15 kV comminuted gypsum stones placed at the geometric focus F2 into 8 times as many fragments and significantly reduced hemolysis in dilute blood 2 and 4 cm off F2 when compared to 200 single pulses generated at 18 kV. Thus the dual-pulse lithotripter enhanced comminution and reduced injury while cutting treatment time in half. Additionally, when cavitation was suppressed by placing the stones in glycerol, the improvement in comminution was reduced to only a twofold increase. This result indicates that the localized and intensified cavitation is the dominant mechanism in the accelerated comminution produced by the dual-pulse lithotripter. [Work supported by NIH Grants Nos. P01-DK43881 and R01-DK55674.]
Lithotripter shock wave interaction with bacteria
Authors: Loske Achim M.
Published: October 2013

The worldwide success of extracorporeal lithotripsy and shock wave treatment of some orthopedic diseases is not doubted. In the future, shock waves could also be used as a method for in vivo drug delivery and gene transfer into target cells; however, basic research is still necessary to understand shock wave interaction with biological tissue, bacteria, and cells. The destructive effects of ultrasonic waves on micro-organisms and the reduction in renal infections after extracorporeal shock wave lithotripsy led to the idea of studying the effects of lithotripter shock waves on bacteria. Independent of possible applications to medicine, this could lead to a new, nonthermal food preservation method. The bactericidal effect of underwater shock waves on Escherichia coli ATCC 10536 and O157:H7, Listeria monocytogenes L8, and Salmonella enterica servovar typhimurium suspensions in isotonic saline solution were studied. Shock waves of about 55 MPa were generated using an electrohydraulic generator. Results indicate that pressure variations, shock wave-created cavitation, and the radiation resulting from the high-voltage spark significantly reduce the viability of these micro-organisms. An analysis of variance revealed that the bactericidal action of shock waves seems to depend on multiple-factor interactions, which vary depending on the type of bacteria.
Numerical simulation of shock and bubble dynamics in shockwave lithotripsy
Authors: Colonius Tim , Tanguay Michel
Published: October 2013

Theoretical evaluation of the efficacy of stone comminution (and potential for tissue damage) during shockwave lithotripsy requires knowledge of the complex stress fields associated with both the incident focussing shock and the dynamics of cavitation bubbles that it induces. While simple models from geometrical acoustics and subsequent modeling of spherical bubbles in isolation (Gilmore equation) can provide estimates, high-speed photography in vitro reveals a far more complex flow with bubble number densities that are sufficiently high such that collective effects associated with a cloud of bubbles are important. This talk will describe a modeling effort aimed at estimating stresses from these complex lithotripter generated flow fields. We compute the time-dependent, compressible, ensemble-averaged two-phase flow equations with a finite-difference scheme. Detailed modeling of the dynamics of bubbles (on the microscale) and high-order weighted essentially nonoscillatory shock-capturing schemes are employed. The model is compared to hydrophone and passive cavitation detection measurements, as well as qualitative comparison with high-speed photography. Finally, we explore collective bubble mechanisms ranging from defocusing and shielding of the stone (for high bubble densities in the focal region) to enhanced stresses due to concerted cloud collapse in a dual-pulse lithotripsy configuration. [Work supported by NIH P01 DK-43881 and NSF under grant CTS-9979258.]
Dynamical response of a bubble submitted to two following shock waves
Authors: Jullien Marie-Caroline
Published: October 2013

A numerical study of the dynamical response of a bubble submitted to two following shock waves is reported. After the passage of a shock wave, a micron-size bubble expands enormously, reaching millimeter size, and then inertially collapses; this is the so-called cavitation phenomenon. The influence of the passage of a second shock wave on the bubble inertial collapse control is investigated. This control may have a considerable impact for drug delivery or gene transfer, by the use of extracorporeal shock wave lithotripsy, which is a technique already used in medical treatment for kidney stones. It is shown that the dynamical response of the bubble qualitatively depends strongly on the forcing shape. Furthermore, for a given forcing shape, the dynamical response depends on both the delay and forcing amplitude ratio between the two applied shock waves. [This work has been done in collaboration with Ruediger Toegel, Claus-Dieter Ohl, and Detlef Lohse (Twente University).]
Measurement of the acoustic response of a wind instrument with application to bore reconstruction
Authors: van Walstijn Maarten , Campbell Murray
Published: October 2013

Reconstruction of a bore from measured acoustic response data has been shown to be very useful in studying wind instruments. Such data may be obtained in different ways; directly measuring the frequency–domain response of an acoustic bore has some distinct advantages over directly measuring time–domain data (for example, by pulse reflectometry), but so far has been unsuitable for producing input data for deterministic bore reconstruction algorithms, due to the limited accuracy at high frequencies. In this paper a method is presented for large-bandwidth measurement of the input impedance of a wind instrument using a cylindrical measurement head with multiple wall-mounted microphones. The influence of the number of microphones and the types of calibration impedance on the accuracy will be discussed, and bore reconstructions derived using this technique will be compared with reconstructions obtained using pulse reflectometry. [Work supported by EPSRC.]
Acoustic input impedance measurements on brass instruments
Authors: Pyle Robert W.
Published: October 2013

Measurement of the acoustic input impedance of a brass instrument can reveal something about the instrument’s intonation, its reasonable playing range, its tone color, and perhaps whether the mouthpiece used for the impedance measurement is appropriate for the instrument. Such measurements are made at sound-presssure levels much lower than those encountered under playing conditions. Thus, impedance measurements may offer the only feasible way to infer something about the playing characteristics of instruments, typically museum specimens, that are too rare or too fragile to be played. In this paper the effects of some of the available choices of sound source and stimulus signal on measurement accuracy will be explored. Driver–transducer nonlinearity, source impedance, signal-to-noise ratio, and any necessary signal processing will be discussed.
Pulse reflectometry as an acoustical inverse problem: Regularization of the bore reconstruction
Authors: Forbes Barbara J. , Sharp David B. , Kemp Jonathan A.
Published: October 2013

The theoretical basis of acoustic pulse reflectometry, a noninvasive method for the reconstruction of an acoustical duct from the reflections measured in response to an input pulse, is reviewed in terms of the inversion of the central Fredholm equation. It is known that this is an ill-posed problem in the context of finite-bandwidth experimental signals. Recent work by the authors has proposed the truncated singular value decomposition (TSVD) in the regularization of the transient input impulse response, a non-measurable quantity from which the spatial bore reconstruction is derived. In the present paper we further emphasize the relevance of the singular system framework to reflectometry applications, examining for the first time the transient bases of the system. In particular, by varying the truncation point for increasing condition numbers of the system matrix, it is found that the effects of out-of-bandwidth singular functions on the bore reconstruction can be systematically studied.
Body vibrational spectra of metal flute models
Authors: Hurtgen Clare M. , Lawson Dewey T.
Published: October 2013

For years, flutists have argued over the tonal advantages of using different precious metals for their instruments. Occasionally, scientists have entered the fray and attempted to offer an objective point of view based on experimental measurements. However, their research often involved actual instruments and performers, ignoring variations in wall thickness, craftsmanship, and human consistency. These experiments have been conducted using a variety of methods; all have concluded that wall material has no effect on tone. This paper approaches the question using simple tubular models, excited by a wind source through a fipple mouthpiece. The amplitude and phase of the harmonic components of the body vibrational signal were measured with a stereo cartridge. Results demonstrated the existence of complex patterns of wall vibrations in the vicinity of a tone hole lattice, at frequencies that match significant harmonics of the air column. Additionally, the tube wall was found to expand in a nonuniform or ‘‘elliptical’’ manner due to the asymmetry of the tone holes. While this method is somewhat removed from direct musical applications, it can provide an objective, quantitative basis for assessing the source of differences among flutes. [Work financed by two Undergraduate Research Support grants from Duke University.]
Development of Mexican standard on hearing protection devices
Authors: Salomon Daniel P. , Razo Jose N.
Published: October 2013

A working group was established by the Secretaria del Trabajo (similar to OSHA) to work on a standard whose objective was to protect workers from noise by trying to predict the performance of HPDs as personal protection equipment in the workplace. Various standards were analyzed including ISO4869, AS-NZ 1270, and ANSI. All evaluate attenuation as a performance descriptor. Experimenter-Fit methods tend to overrate field performance due to very different field versus laboratory conditions. Subject-Fit methods result in lower attenuation values which are more likely to be achieved by groups of workers, even if inadequately motivated and trained. Subject-Fit methods encountered large resistance, including laboratories’ difficulties in finding naive subjects, commercial interests of manufacturers, risk of overprotection, etc. Having settled for an Experimenter-Fit method, the intent is to do more to help protect workers’ hearing through other means available. A thorough, yet fairly simple, HPD Selection and Usage Guide was prepared as an Informative Appendix. Relevant information should be provided to the user with the Packaging. NRR wars should be avoided. The standard includes a list of contents, instructions, and warnings that must be included. A sample label containing key information is detailed in another Informative Appendix.
The subjective testing of hearing protectors and issues of interlaboratory variability—The Australian/New Zealand experience
Authors: Williams Warwick
Published: October 2013

Over the past few years the proposal to use ‘‘inexperienced’’ subjects to test the attenuation of hearing protectors has been controversial, particularly in the USA and in connection with the development of draft ISO 4869–7 Acoustics–hearing protectors–Part 7: Subjective method for measurement of sound attenuation–Subject-fit method. Much of the discussion centers around the belief that by using ‘‘inexperienced,’’ ‘‘unassisted’’ test subjects attenuation results will be unrepresentative of the ‘‘true’’ figure. Both Australia and New Zeland have been using subjective testing of hearing protectors with inexperienced, unassisted test subjects for many years. The results have been positive and the conjectured difficulties have not eventuated. Any ‘‘unusual’’ results have been readily explained. The Australian acoustic community has no difficulty with the test methodology. Further it is the opinion of Occupational Health and Safety workers that ‘‘inexperienced,’’ subjective testing most closely represents attenuation values typical in the well informed workplace.
Hearing protection: Limits to attenuation
Authors: Berger Elliott H. , Kieper Ronald W. , Gauger Dan
Published: October 2013

With louder and louder weapon systems being developed and military personnel being exposed to steady noise levels approaching and sometimes exceeding 150 dB, the interest in greater amounts of protection is evident. When the need for communications is included in the equation, the situation is even more extreme. New initiatives are underway to design improved hearing protection, including active noise reduction (ANR) earplugs and perhaps even cancellation of head-borne vibration. With that in mind it may be useful to explore the limits to attenuation, and whether they can be approached with existing technology. Data on the noise reduction achievable with high-attenuation foam earplugs, as a function of insertion depth, will be reported. Previous studies will be reviewed that provide indications of the bone-conduction (BC) limits to attenuation that, in terms of mean values, range from 40–60 dB across the frequencies from 125 Hz to 8 kHz. Additionally, new research on the effects of a flight helmet on the BC limits, as well as the potential attenuation from deeply inserted passive foam earplugs or communication earplugs, worn with passive earmuffs, or with active-noise reduction (ANR) earmuffs, will be also examined.
Do sound restoration hearing protectors provide adequate attenuation for gunfire noise?
Authors: Franks John R. , Murphy William J.
Published: October 2013

Sound restoration earmuffs have an electronics package installed in a passive earmuff shell or earplug body. The electronics package consists of a microphone placed on the outside of the ear cup or earplug and a limiting amplifier driving a loudspeaker placed inside the ear cup or the earplug. The devices are designed to provide unity gain or better to low-level signals and to shut off when sound exceeds a given level. The effects of several sound restoration earmuffs and two nonlinear orifice earplugs on gunfire for several different weapon types were analyzed. In general, the nature of the electronics packages was such that the studied devices provided the same attenuation when turned on as when turned off. However, as with passive devices, a single protector, be it by earplug or earmuff, appears to be inadequate for gunfire when more than just a few shots are fired. Sound restoration earmuffs are best used with a well-fitted earplug, since the electronics can compensate for both the insertion loss of the earmuff and the earplug. However, nonlinear orifice earplugs may not provide sufficient protection for extended sessions of target practice.
Industrial warning signal detection under augmented hearing protection devices
Authors: Casali John G. , Robinson Gary S.
Published: October 2013

The results from several experiments concerning the effects of passive, Amplitude-Sensitive Sound Transmission (ASST), and Active Noise Reduction (ANR) HPDs on vehicle backup alarm detection by normal hearers are reviewed. Masked thresholds under a Peltor electronic ASST earmuff and an AEARO orifice-based ASST earmuff were found not to differ significantly from thresholds under conventional earmuffs of similar volume to the ASST devices, in pink noise of 75, 85, and 95 dBA. A separate analysis using a KEMAR manikin indicated that subjects’ chosen settings of the Peltor ASST earmuff gain control did not significantly increase the noise exposure dose over the earmuff’s amplifier-off setting. In low frequency-biased ‘‘red’’ noise conditions (averaged across 85 and 100 dBA), masked thresholds under a Bose ANR earmuff and an AEARO foam earplug were significantly lower (better) than those under a Peltor passive earmuff having weaker low frequency attenuation, but there were no differences among the 3 devices in pink noise, nor at 85 dBA in either noise. Detection thresholds in 85 dBA noise were 3.2–4.5 dB lower in the 3 protected conditions than in the unprotected condition, attesting to the fact that HPD use can facilitate signal detection under certain conditions for normal hearers. [Work supported by U.S. Bureau of Mines (now NIOSH), Bose Corp., Aearo Corp.]
Sound propagation in straight, conical, exponential and real external ear canal
Authors: Fortkamp Alexsander , de Lima Washington J. N. , Gerges Samir N. Y.
Published: October 2013

Mathematical and numerical modeling of the noise attenuation of hearing protectors (HP) is a powerful tool for optimizing the HP design. The human ear canal has a complex geometry and therefore can be modeled by numerical methods. Below the cut-off frequency, plane wave propagation can be modeled by simple geometric. In this paper straight, conical, exponential, and real ear canal geometry are considered. The Webster equation is used to model the sound propagation inside these canals. In this paper a comparison of sound field propagation obtained from a plane wave model of the simple geometries and the finite elements model of the real ear are presented.
Radiative transport for elastic waves
Authors: Papanicolaou George
Published: October 2013

Radiative transport equations for elastic waves in random media can be derived from first principles in a certain regime of physical parameters. They account for both longitudinal and transverse waves, as well as polarization. Transport boundary conditions can also be derived from first principles. An important application of transport theory is to the analysis of the deep coda of elastic waves, where some rather interesting phenomena of energy distribution emerge. A reference to work in this area is by Ryzhik et al. [Wave Motion 24, 327–370 (1996)]. Another reference, more directly related to seismics, is Ryzhik et al. [Bull. Seismol. Soc. Am. 86, 1107–1115 (1996)].
Anisotropic diffusion of elastic waves
Authors: Turner Joseph A. , Yang Liyong
Published: October 2013

The scattering of elastic waves in heterogeneous, anisotropic media is relevant for ultrasonic and seismic waves for media including textured polycrystalline media, fiber-reinforced composites, and media containing oriented cracks. Elastic waves propagating through such media lose energy due to scattering from the heterogeneities of the material. In anisotropic media, this scattering attenuation is a function of propagation direction. Here, the case of transverse isotropy is examined. The problem is formulated in terms of the Dyson and Bethe–Salpeter equations for the mean and covariance of the anisotropic Green’s function, respectively. Solution of the Dyson equation, using the first-order smoothing approximation (FOSA), gives attenuations for shear horizontal, quasicompressional, and quasishear waves as a function of direction and frequency. The FOSA is also used to reduce the Bethe–Salpeter equation to an elastic radiative transfer equation. In the limit of many scattering events, this equation then reduces to an anisotropic diffusion equation. The primary result is the diffusion tensor given in terms of the various scattering functions that characterize the transfer of elastic energy between the different wave types of directions. The results are discussed in terms of applications to textured metals and media with aligned cracks. [Work supported by NSF and DOE.]
Performance of a small, low-lift regenerator-based thermoacoustic refrigerator
Authors: Poese Matthew E. , Garrett Steven L.
Published: October 2013

A regenerator-based thermoacoustic refrigerator [Swift, Gardner, and Backhaus, ‘‘Acoustic recovery of lost power in pulse tube refrigerators,’’ J. Acoust. Soc. Am. 105(2), 711 (1999)] has been constructed. It is capable of moving about 5 W across a 40°C temperature span. The machine operates with air at atmospheric pressure and is driven by an off-the-shelf electro-dynamic loudspeaker capable of producing peak-to-mean pressure ratios up to 12%. The thermal core of this research device contains an exhaust-side shell and tube heat exchanger (with water as the secondary heat transfer fluid), a regenerator made of 88 annular stainless-steel screens, and a constantan wire electrical heater that applies a measurable heat load to the cold side of the regenerator. An annular latex diaphragm is placed over the cold side of the regenerator to stop time-averaged mass flow through the regenerator and insulate the cold side [Gedeon, ‘‘DC Gas Flows in Stirling and Pulse Tube Cryocoolers,’’ in Cryocoolers 9, edited by R. G. Ross (Plenum, New York, 1997)]. Detailed measurements of heat load, temperature span, and exhaust heat flux will be presented and compared to DeltaE. [Work supported by ONR.]
Computational and experimental investigation of minor losses in high amplitude acoustic resonators with varied cross section
Authors: Doller Andrew J. , Boluriaan Said , Atchley Anthony A. , Morris Philip J.
Published: October 2013

Minor losses in high amplitude oscillatory flows through changes in resonator cross section are not well understood. The application of numerical simulation in the study of these losses has shown promise over the past few years. A computational model has been developed based on a time accurate numerical simulation of the full Navier–Stokes equations on parallel computers. A rectangular clustered mesh is used to capture flow field details in the boundary layers and in the vicinity of changes in cross section. Results of simulations are compared to measurements of acoustic and time-averaged pressures at multiple points along the length of resonators and are compared at multiple drive amplitudes. The resonators consist of two sections of straight brass pipe of different diameters and joined through either steplike or conical couplers. They are driven at the end of the large diameter pipe with a rigid piston. The end of the small tube is rigidly terminated. [Research supported by ONR.]
The cochlear outer hair cell motility must be based on a tonic change in the membrane voltage: Implications for the cochlear amplifier
Authors: Allen Jont B. , Fahey Paul F.
Published: October 2013

It has been widely assumed that the function of the OHC is to increase the sensitivity and frequency selectivity of the cochlea via a phasic OHC voltage, which controls the soma length. This action is called the cochlear amplifier. According to this view the length of the OHC is assumed to follow the stimulus to the upper frequency limit of hearing, in a phasic manner (cycle by cycle), adding power at the signal frequency. We propose an alternative view that the OHC controls the dynamic range in a parametric, or tonic manner, via the cells axial stiffness. In this case the change in gain seen by the IHC does not require a phasic response at high frequencies. The OHC could mediate a fast acting gain control, via impedance changes, that follows the OHC membrane tonic voltage envelope. Given a level-dependent change in dynamic range (i.e., dynamic range compression), the tuning and sensitivity would necessarily change. Our analysis and conclusions are based upon a re-interpretation of existing mammalian outer hair cell (OHC) studies using a generalized admittance matrix formulation of the OHC, that relates the plasma membrane voltage and current to the soma axial force and velocity.
A measure of internal noise based on sample discrimination of level differences
Authors: Jesteadt Walt , Nizami Lance , Schairer Kim S.
Published: October 2013

In a sample discrimination task, subjects are presented with stimuli drawn from each of two overlapping distributions and asked to identify the distribution from which the stimuli were drawn. Lutfi [J. Acoust. Soc. Am. 86, 934–944 (1989)] has shown that performance in this task is nearly ideal when the stimuli are single sinusoids differing in level, for distributions with a mean difference of 5 dB and standard deviations of 5 dB. Theoretically, as the mean difference and standard deviation of the distributions are decreased, however, performance should increasingly deviate from ideal, yielding an estimate of internal noise. To test this hypothesis, intensity discrimination was measured for seven subjects’ sinusoids drawn from distributions that had mean differences ranging from 0.1–2.2 dB, in steps of 0.3 dB. The standard deviation of each distribution was always half the mean difference, resulting in uniform ideal performance. Estimates of internal noise obtained by fitting a function to the observed d-prime values agreed with estimates obtained from psychometric functions measured using the same differences in level. Analyses of decisions on individuals trials indicated that the two observation intervals were weighted equally. [Work supported by NIDCD.]
Effects of the finite speed of sound in—and surgical modification of—the cochlea
Authors: Talmadge Carrick , Tubis Arnold
Published: October 2013

Most analyses of the mechanics of the cochlear to date have involved the assumption that the cochlea is a rigid chamber filled with incompressible fluids. However, in vivo measurements of the cochlear fluid pressure or basilar membrane motion necessarily must be performed by first surgically modifying the cochlea through removing part of the cochlear bone. This modification violates the assumption of a rigid chamber, and can significantly modify the mechanics of the cochlea. Because of anatomical constraints, many of these measurements are performed near the base of the cochlea. For the species that are used (guinea pig, gerbil, chinchilla, etc.), measurements near the base necessitate the use of stimulus tones with frequencies that are too high for the assumption of an incompressible fluid to be valid. In this paper, we discuss the modifications to the cochlear mechanics necessary to accommodate both the finite speed of sound and the surgical modification of the cochlea. We show that both of these modifications can easily be incorporated into numerical cochlear model codes. Semianalytic solutions based the WKB method are also discussed.
Dichotic sound interference products, sound localization cues, and wave interaction at a neural level
Authors: Vassilakis Pantelis
Published: October 2013

The possibility of sound interference products arising from neural wave interaction is reexamined through five dichotic experiments. It is argued that interference percepts do not arise unless sound waves interact physically. Experiment 1 examines the possibility of beating for diotic sine stimuli. In this case, diplacusis would introduce beating if neural wave interaction could result in interference products. No beating was observed. Experiments 2a, 2b, and 2c examine the effect of dichotic phase changes on sine stimuli in terms of loudness, direction, and stereo-image-width. It is shown that such phase changes lead to perceptions incompatible with the interference principle and consistent with claims of binaural phase differences as sound-localization cues. Experiment 3 uses two-component dichotic stimuli. The results indicate that, when two sines with slightly different frequencies are presented dichotically, the constantly shifting phase between them affects a constantly shifting localization of the two-component complex stimulus. Depending on the frequency difference, this shift may be interpreted as a perceived rotation of the sound and/or a timbral fluctuation, often confused with loudness fluctuation. The findings do not support the claim of sound interference products arising from neural wave interaction. They are consistent with sound localization studies and indicate that the dichotic beats reported previously must have been a misidentified rotating sensation.
Computation and analysis of a nonlinear nonlocal cochlear model with applications to multitone interactions in hearing
Authors: Xin Jack , Qi Yingyong , Deng Li
Published: October 2013

A nonlinear nonlocal cochlear model of the transmission line type is studied to capture the multitone interactions and tonal suppression effects. The model can serve as a module for voice signal processing. It is a one-dimensional (in space) damped dispersive nonlinear partial differential equation (PDE) based on mechanics and phenomenology of hearing. The elastic damping is a nonlinear and nonlocal functional of basilar-membrane displacement. The initial boundary value problem is solved with a semi-implicit second-order finite difference method. Numerical results are shown on two-tone suppression from both high-frequency and low-frequency sides, consistent with known behavior. Suppression effects among three tones are demonstrated by showing how the response magnitudes of the two fixed tones are reduced as the third tone is varied in frequency and amplitude. Qualitative agreement with existing cat auditory neural data is observed. The model is thus simple and efficient as a processing tool for voice signals. Mathematical analysis of global well-posedness of the model PDE and the existence of multitone solutions will also be shown via the method of a priori estimates and fixed point theory. [Work partially supported by ARO and NSF.]
The effect of middle ear stiffness changes on noise hearing thresholds in presence of a high level background pure tone
Authors: Aazh Hashir
Published: October 2013

In this study the interaction of two different sounds, loud low frequency pure tone and noise in threshold level was evaluated during air pressure changes in order to determine the effects of middle ear stiffness changes on noise threshold, with presence of a high level background pure tone. Forty-two normal hearing subjects after otoscopy, audiometry, and tympanometry participated in this experiment. During the presence of a background pure tone (220 Hz, 90 dB), the thresholds of wide band, low pass filtered, and high pass filtered noises were measured (in dB SPL) in three ear canal air pressure conditions (0, +300, −300 dapa) through a probe set. The pump manometer system of an AZ7 impedance audiometer was used to change the air pressure in an external auditory canal. The means of noise thresholds were significantly greater in 0 dapa than +300 and −300 dapa. But there was no significant difference between noise thresholds in +300 and −300 dapa. By increasing the middle ear stiffness, the transmission of background loud tone will decrease and ear overload is omitted, then the noise hearing thresholds will be better.
Design of an analog cochlear implant
Authors: Flores Joel , Becerra Jorge , Ramrez Patricia L.
Published: October 2013

The present work describes the development of cochlear implants, an effective methodology in the treatment of severe to profound hearing loss. A group in the acoustic section from ESIME Zacatenco is investigating the development of cochlear implants in Mexico. Cochlear implants are known to have had encouraging results for subjects with postlinguistic deafness; however, one of the great drawbacks of these implants is their cost. For this reason, this work described here counts on using interdisciplinary personnel to accomplish the construction of a cochlear implant of the low-cost analog type, without losing sight of its objective: the best possible restoration of hearing.
Spectral finite elements combined with energy flow techniques: Hybrid approach for the energy flow parameters evaluation
Authors: Ichchou Mohammed , Jezequel Louis
Published: October 2013

Energy flow parameters involved in many energy flow techniques (statistical energy analysis, or its local formats) are mainly deduced from pure analytical basic theories. However, in practice, application of SEA or alternatives to complex realistic structures suffers from the choice of subsystems and needed relevant inputs. This paper proposes a procedure which reuses existing reduced finite element modeling of the structural component in view of energy flow parameters identification. Precisely, extraction and analysis of dispersion curves for typical structures, wave numbers, and energy velocities, are first concerns in this contribution. The dispersion curves extraction is based on a spectral problem and uses a reduced finite element model. Properties of eigensolutions of the posed spectral problem are first demonstrated and some remarkable aspects in term of energy flow parameters are discussed. Definitions of energy velocity, needed for modal densities expressions associated with the spectral finite element model, are presented in depth. Ultimately, extension of the proposed formulation, in order to deal with coupled complex structural components is given. This leads to the numerical evaluation of diffusion matrix (reflection and transmission parameters), often needed in order to define coupling loss factor or equivalent energy transfer quantities. Some numerical applications are finally presented.
Summary of energy flow measurements and calculations made on the INCE standard test structures
Authors: Hambric Stephen A.
Published: October 2013

In 1996 a series of standard test structures was conceived and manufactured by members of the Institute of Noise Control Engineering (INCE) [Cuschieri, Burroughs, and Carroll, Evaluation of Structure-Borne Noise Prediction Techniques Review, Proceedings of Noise-Con 98, April 1998, pp. 315–320]. The structures include a Lexan T-shaped beam and two ribbed panels of identical geometries but different materials: aluminum and lexan. In subsequent years, a wide variety of investigators from the U.S. and around the world have conducted experimental and numerical studies on the test structures, particularly on energy flow parameters, such as power input, power dissipation, and power flow. Most of the studies have been performed at low frequencies, and have shown phenomena such as the conversion of flexural wave power to longitudinal wave power at the T-beam joint and the nature of the structural intensity fields in the ribbed panels. Measurements and computations compare well. At high frequencies, Statistical Energy Analysis (SEA) techniques have shown that energy tends to become trapped in the drive leg of the T-beam. SEA studies on the ribbed panels show that the Lexan panel transmits less energy across the ribs than the aluminum panel does due primarily to differences in material loss factor.
Active minimization of acoustic energy density in enclosed sound fields
Authors: Sommerfeldt Scott D. , Faber Benjamin M.
Published: October 2013

Minimization of acoustic energy density has been investigated for active noise control applications for enclosed sound fields. The standard approach of minimizing the squared acoustic pressure has been shown to often lead to localized control of the sound field, which may be undesirable. It has been shown that minimizing the energy density often leads to improved global attenuation of the field, since quantities dependent on both the pressure and particle velocity are minimized. This penalizes active control solutions in which the pressure is minimized and the particle velocity is increased, as often occurs when simply minimizing the squared pressure. Minimizing the acoustic energy density also requires multiple microphones to be integrated as part of the error sensor. However, in many cases the number of microphones can be reduced by strategically placing the sensor in a location where one component of the particle velocity is assumed to be negligible. Active control results will be shown for an enclosed sound field to compare the reduction obtained using energy density with that obtained for minimizing squared pressure. In addition, the attenuation obtained using a full three-dimensional energy density sensor will be compared with the control achieved using an error sensor with fewer microphones.
Effects of cross-sectional variation in T-beam structures
Authors: Coronado-Matutti Alberto , Sampaio Rubens
Published: October 2013

Structural modification of geometry has been proven to be very effective when it was necessary to improve the dynamic performance of some structures [A. J. Keane, J. Sound Vib. 185(3), 441–453 (1995)]. This technique has the advantage of achieving good performance even without the heavy viscoelastic treatments or the high-tech, high-cost active vibration control schemes. In this paper a T-beam structure [R. P Szwerc, C. B. Burroughs, S. A. Hambric, and T. E. McDevitt, J. Acoust. Soc. Am. 107, 3186–3195 (2000)] of uniform cross section is modified in order that the power transmitted to some point, due to an external excitation, is minimized. The modification of geometry is made only in one of the legs of the T-beam. The structure will be modeled combining the finite element method and the spectral element method. The dynamic response of the structures (initial and modified configurations) will be carried between 0–350 Hz; the first ten resonances are placed in this frequency range. After defining a specific region to be optimized via genetic algorithms, the options to minimize the frequency response will be shown; in some cases it is possible to move one or more resonances, in others only a reduction on the response between the nonresonance frequencies is permitted.
Speech intelligibility training using automatic speech recognition technology
Authors: Kewley-Port Diane , Dalby Jonathan , Burleson Deborah
Published: October 2013

This paper describes two computer-based speech training systems that have been developed over the past decade of research. This research began at Indiana University and has been continued and commercialized at Communication Disorders Technology, Inc. The Indiana Speech Training Aid (ISTRA) uses a speaker-dependent speech recognizer and is designed for use by speech-language pathologists treating clients with articulation disorders of various sorts. The HearSay system employs a speaker-independent recognizer and is designed to improve the segment-level intelligibility of learners of English as a second language. Both systems employ minimal-pairs contrast training and share the underlying assumption that effective speech training requires intensive individual practice accompanied by evaluative feedback. The key technological achievements of ISTRA and HearSay are the methods employed for providing valid feedback on the intelligibility of the client’s speech. This paper will describe these methods and report the results of new training studies that demonstrate that computer-based speech training can be an effective method for improving the English speech intelligibility of learners of English as a second language. [Work supported by Grants No. DC00893 and No. DC02213 from the NIH–National Institute on Deafness and Other Communication Disorders.]
String kernels for the classification of speech data
Authors: Goddard Close John Ch. , Martinez Licona Fabiola M. , Martinez Licona Alma E. , Rufiner H. Leonardo
Published: October 2013

Support Vector Machines (SVM) have been applied to a wide variety of classification problems with excellent results. This has to do with their provable generalization ability derived from Statistical Learning Theory. Recently specialized kernels, such as the Fisher kernel and string kernels, have been introduced in an attempt to apply the same SVM framework to sequential data. Notable results have been obtained on classification tasks related to biosequences and text documents showing that the specialized kernels may provide a viable and interesting alternative to other classifiers, such as those using Hidden Markov Models. String kernels are particularly attractive because of their conceptual simplicity and they also furnish insight into the task of sequential data classification. In the present paper string kernels are applied to a new application area, that of automatic speech recognition. In particular, different string kernels are tested on the task of phoneme recognition and the results obtained are compared with those from several other common classifiers. This comparison reveals the potential of string kernels as a simple and feasible alternative to other established methods for certain speech recognition tasks. [Work supported by CONACYT under Project 31929-A.]
Neural-network-based voice-tracking algorithm
Authors: Baker Mary , Stevens Charise , Chaparro Brennen , Paschall Dwayne
Published: October 2013

A voice-tracking algorithm was developed and tested for the purposes of electronically separating the voice signals of simultaneous talkers. Many individuals suffer from hearing disorders that often inhibit their ability to focus on a single speaker in a multiple speaker environment (the cocktail party effect). Digital hearing aid technology makes it possible to implement complex algorithms for speech processing in both the time and frequency domains. In this work, an average magnitude difference function (AMDF) was performed on mixed voice signals in order to determine the fundamental frequencies present in the signals. A time prediction neural network was trained to recognize normal human voice inflection patterns, including rising, falling, rising–falling, and falling–rising patterns. The neural network was designed to track the fundamental frequency of a single talker based on the training procedure. The output of the neural network can be used to design an active filter for speaker segregation. Tests were done using audio mixing of two to three speakers uttering short phrases. The AMDF function accurately identified the fundamental frequencies present in the signal. The neural network was tested using a single speaker uttering a short sentence. The network accurately tracked the fundamental frequency of the speaker.
Diagnostic system for speech articulation and speech understanding
Authors: Czyzewski Andrzej , Kostek Bozena , Skarzynski Henryk
Published: October 2013

With the increase in access to multimedia computers, speech training can be made available to patients with no continuous assistance required from speech therapists. Another function the system can easily perform is screening testing of speech fluency providing directed information to patients who have various speech disorders and problems with understanding speech. The idea underlying the proposed system is a programmed speech therapy training algorithm consisting of diagnostic tools and rehabilitation devices connected with it. The first function the system described in the paper has to perform is data acquisition where information about the patient’s medical history is collected. This is done through electronic questionnaires. The next function is analysis of the speech signal articulated by the patient when prompted by the computer followed by some multimedia tests carried out in order to assess the subject’s ability to understand speech. Next, the results of the electronic questionnaire, the patient’s voice, and the patients’ reactions are automatically analyzed, based on that the system automatically diagnoses possible speech disorders and how strong they are. A large number of schoolchildren were tested with this method. In the paper foundations of applied speech testing method and obtained results will be demonstrated.
Choice of speech features for single-channel vibrotactile stimulators
Authors: Barros Alessio Trindade , Freire Raimundo Carlos S. , Deep Gurdip Singh
Published: October 2013

The choice of speech features for coding of a single-channel vibrotactile stimulation signal was analyzed to obtain a better discrimination between phonemes that compose each viseme of the Portuguese language. Two different speech features were selected for comparison, based on previous investigations [I. R. Summers and D. Gratton, IEEE Trans. Rehab. Eng. 3, 117–121 (1993)]: the amplitude envelope—AE and the zero-crossing frequency—ZRX of the speech signal. A codification scheme (PRP) was proposed, with output based on the switching between AE and ZRX, depending on the value of ZRX. The selected voice items were consonants of each viseme, in the context vCv: ata/ada/ana, apa/aba/ama, aka/aga/alha, asa/aza, afa/ava, acha/aja, and ala/ara, for different combinations of vowels. Histograms for each feature were done for each consonant of a viseme, and were compared in the context of each viseme. It was observed that the output of PRP was the same as AE for most of the consonants, but different for fricatives. PRP allowed better discrimination than AE and ZRX alone. The difficulty of single-channel stimulator with discrimination of fricatives is known in the literature, and the proposed scheme showed better results for this work. [Work supported by CAPES.]
Speech coding for fast speech commands recognition
Authors: Herrera Abel , Gardida Arturo
Published: October 2013

Isolated speech recognition could be more appropriate than continuous speech recognition systems for some word command applications. In this paper, some speech coders are explored that can be useful for fast word command recognition using low memory storage. First, decimated critical bands vectors by frame are used as coding features. The accuracy is around 94% for a set of ten commands; they are used at first step. After that, using a reduced DTW classification scheme, the accuracy for a small subset of possible candidates is above 99%. Another coding scheme is using the Karhunen–Loeve Transform (KLT) combined with multisection vector quantization. The KLT is not a fast transform; however, used at the subword level it gives a fast recognizer with accuracy around 99%. The classifiers in both cases are fast, around a second, and require low memory storage.
Adapted speech recognizer to improve English pronunciation of Hispanics
Authors: Garcia Maria A. , Arechiga Rene O.
Published: October 2013

This work presents advances on an adapted speech recognition system, based on hidden-Markov models (HMMs), to help Hispanics in their pronunciation of English. The main results so far, with English digits, show significant recognition improvements of the adapted recognizer over the not adapted one, both for one speaker (100% vs 94%) and for groups of them (96% vs 74%). The adaptation uses maximum likelihood linear regression (MLLR). Two databases were used in the experiments. The TIMIT database to train the recognizer models with English native speakers, and the Latin-American Spanish database to adapt and test the adapted and not adapted recognizers. The project, of which this work is a part, is an English pronunciation tutor, whose main parts are this adapted speech recognizer, a pronunciation evaluator (phonetic and prosodic) and a dialog manager. English speech recognition systems are trained with native speakers, and most of the recognition errors are attributed to the interference of foreign accents. This is why it is necessary to adapt the models to take into account the characteristic features of a given population; in this case Hispanics. (To be presented in Spanish.)