bluuebunny/crossref_metadata_2025 huggingface dataset:
Speech recognition using hidden Markov models with multiple-track polynomial regression functions as nonstationary states
Authors: Aksmanovic M. , Deng L.
Published: October 2005

The formulation of the hidden Markov model (HMM) has been successfully used in automatic speech recognition for almost two decades. In the standard formulation, the individual states in the HMM are each associated with a generally distinct but stationary stochastic process. This makes the standard HMM inadequate for representing the nonstationary property of the many speech segments intended to be described by the HMM-state statistics. A generalized HMM has been developed to overcome this inadequacy by introducing state-dependent polynomial regression functions on time that serve as the time-varying means in the HMM’s Gaussian output distributions [e.g., L. Deng, Signal Process. 27, 65–78 (1992)]. Recently, Aksmanovic and Deng extended the above model so that the state-dependent nonstationary process contains multiple tracks of the polynomial functions. This new parametric class of nonstationary-state HMMs has been implemented and evaluated. Experiments on fitting models to speech data, on limited-vocabulary word recognition, and on phonetic classification demonstrated advantages of the nonstationary-state HMMs over the traditional stationary-state HMMs. Details of the model implementation and of the experimental results will be described. In particular, the focus will be on comparisons between uses of single-track and multiple-track regression functions defined within the HMM states, and on comparisons among uses of varying orders of the state-dependent polynomial regression functions.
Context-dependent acoustic subword modeling for connected digit recognition
Authors: Lee Chin-Hui , Chou Wu , Juang Biing-Hwang , Rabiner Lawrence R. , Wilpon Jay G.
Published: October 2005

Accurate and robust connected digit recognition is essential for a wide range of telecommunication services. Based on training and testing using only clean network digit data, and using the same whole-word model architecture as in the TI/NIST connected digit testing, the string error rate increased from less than 1% to more than 5%. The performance degraded even further when evaluated on data collected with different network conditions. Most of the observed errors were caused by changing channel characteristics, highly variable digit pronunciations, and inadequate modeling of cross-digit coarticulation. Results are presented for a number of context-dependent whole-word and subword modeling techniques developed to overcome some of the above problems. The most effective one is a new acoustic subword modeling approach that assumes that each digit model consists of three parts, namely, head, body, and tail subword units. Multiple heads and tails are also allowed, one for each of the 11 possible preceding and following digits and the background. Cross-digit coarticulation is modeled by connecting the pair of digits through the corresponding tail and head units. Testing on about 12 000 digit strings, collected from five regions, this new model architecture reduced the string error rate to under 2%.
On the automatic classification of pitch movements
Authors: Bosch L. F. M. ten
Published: October 2005

The design of an algorithm that classifies pitch movements is discussed. The algorithm consists of two steps: (a) a training phase, which is based on a labeled training corpus of 249 grammatical Dutch sentences, and (b) the recognition phase. The IPO labeling system [J.’t Hart etal., APerceptualStudyofIntonation. AnExperimental-phoneticStudytoSpeechMelody (Cambridge U.P., Cambridge, 1990)] is used. The setup of the algorithm is based on (A) the extraction of two time-varying features (pitch, and a feature called vowel strength), followed by (B) a search for characteristic movements over time in the resulting feature set, (C) a decision procedure similar to a linear discriminant analysis, and (D) the use of an intonation grammar. Without using (D), a classification rate of 81% is attained by using the database of 249 semispontaneous sentences, spoken by more than 40 male and female voices. Invoking the grammar [step (D)], at least 6% of the remaining 19% errors can be resolved by disambiguating phonetic candidates. The approach toward the automatic recognition of pitch movements is essentially different from the approach followed in speech segment recognition systems based on hidden Markov modeling. This difference will be discussed. Although the algorithm is developed for Dutch speech material and for the labels as defined in the IPO intonation labeling system, the method is generally applicable to other languages and to other (consistent) intonation labeling systems.
Sample size necessary to provide acceptable reproducibility in laboratory hearing protector attenuation testing
Authors: Franks John R. , Merry Carol J.
Published: October 2005

The ANSI standard method for determining real-ear attenuation at threshold requires that no less than ten subjects be used and that each subject be tested no less than three times each with ears occluded and with ears open. Attenuation is calculated as the difference between each paired open and occluded threshold. The grand mean attenuation is calculated for all 30 attenuations. The standard deviation calculation is based on the difference between each attenuation and the grand mean. The ISO standard requires no less than 16 subjects with no less than one calculation of attenuation. From an interlaboratory study using the subject-fit method, it was possible to evaluate the number of subjects and number of repeated measures per subject necessary to reach a desired confidence interval and level of accuracy. The results indicated that the number varies with the type of hearing protector and center frequency of the testing noise. To reach ±3 dB for an earmuff, as few as 11 subjects with one attenuation test per subject was adequate, while for one of the earplugs more than 75 subjects with two attenuation tests per subject were necessary. Implications of these results on laboratory testing strategies will be discussed.
Measuring the performance of circumaural hearing protectors and communication head sets at low frequencies
Authors: Ryan J. G. , Shaw E. A. G. , Brammer A. J. , Zhang G.
Published: October 2005

An enclosure has been constructed to evaluate the performance of active noise reduction (ANR), circumaural head sets, and hearing protectors, at frequencies below 800 Hz. It consists of a massive (5-kg), flat base plate against which one cup cushion is sealed by spring pressure. The sound pressure within the cup is recorded by a microphone flush-mounted in the base plate. A 20-cm-diam aluminum tube, 20 cm in length, is attached by annular flanges to the base plate at one end and, at the other, to a 15-cm-diam loudspeaker, which drives the enclosed volume so formed. The excitation of cross modes within the tube is suppressed by a 2.0-cm cylindrical annular layer of felt, in contact with the inner wall of the tube. The performance of the measuring system will be described and results reported for the attenuation of an analog ANR head set. [Work done in collaboration with the Defence and Civil Institute of Environmental Medicine.]
A new explanation for the double resonance of a pure vapor bubble
Authors: Mao Yi , Crum Lawrence A. , Roy Ronald A.
Published: October 2005

The second resonance frequency f0(2) of a pure bubble appeared first in R. D. Finch and E. A. Nippiras’ theoretical analysis [J. Acoust. Soc. Am. 53, 1402–1410 (1973)]. However, no one has been able to observe a vapor bubble oscillating at f0(2). A new analysis based on numerical calculations shows that f0(2) is physically unstable. By analogizing bubble oscillations with those of a mass-spring system, it is found that the resonance frequency F(fd) depends on the driving frequency fd and the real bubble resonance f0(2) is a solution of F(fd)=fd. The behavior of F(fd) near fd=f0(2) shows that the bubble tends to shift its oscillation frequency away from f0(2) (repulsive); whereas F(fd) near the ordinary resonance f0(1) is attractive. A theory developed by the authors for directly calculating the resonance frequency and the damping constant without using the mass-spring analogy gives only f0(1). Therefore, we are forced to the conclusion that f0(2) results from an improper mass-spring analogy and is not physically observable. [Work supported by ONR.]
Vapor bubbles revisited
Authors: Finch Robert D.
Published: October 2005

Mao etal. [J. Acoust. Soc. Am. 93, 2379 (1993)] have revived interest in the subject of the dynamics of vapor bubbles, including the issue of ‘‘double’’ resonances. Some earlier work on this topic will be summarized, and a possible application will be discussed.
A new shift factor for the characterization of the dynamic properties of viscoelastic materials
Authors: Nashif Ahid D.
Published: October 2005

During the course of characterizing the dynamic properties of a variety of viscoelastic materials, over the last 10 yr, the Arrhenius equation was found to be valid for describing the shift factor over the glassy, transition, and rubbery regions of the material. Based on this observation, the work described was initiated to define the activation temperature in terms of the dynamic behavior of viscoelastic materials. It will be shown, both empirically and analytically, that the activation temperature is a unique function of the dynamic behavior of viscoelastic materials, which can be accurately determined once the properties of the material have been measured for any frequency but over a wide temperature range. Two advantages exist for using this unique expression to describe the shift factor instead of the previously used ones. The first is that it eliminates the cumbersome and sometimes questionable ways of curve-fitting the data. The second, which is more important, is that by using this unique function to reduce the data results in a smooth set of points without discontinuities; otherwise, the measured data are questionable, and their validity should be investigated.
Characterization of candidate standard polymers
Authors: Reader Wayne T. , Emery N. Scott , Pena Lois S.
Published: October 2005

Investigators actively engaged in the measurement of the moduli and loss factors of viscoelastic polymers are well aware of the diversity of results obtained by different measurement techniques—even though the specimens are fabricated to be identical. To alleviate this perennial problem, the ASA has sponsored working group S2WG79 to identify several polymers for use in calibration of the multitude of apparatuses being used to characterize dynamic moduli and loss factors. Criteria used to select candidate polymers include ready availability, property stability, bondability to mounting blocks, versatility to cover wide frequency and temperature ranges, manufacturing repeatability, and, most important, rheological simplicity. Three candidate polymers have been proposed and distributed for initial evaluation. Evaluations of how these candidates performed on a Polymer Laboratories DMTA will be presented, and their suitability for standard materials as determined by the selection criteria will be addressed.
Measurement of changes in the bulk material properties of solid rocket motor propellant
Authors: Nici Rosario , Bond Leonard J.
Published: October 2005

Solid rocket motor propellant’s bulk viscoelastic material properties change with age and service history. Safety considerations preclude the use of propellant after an estimated service life. An ultrasonic method that characterizes the material by tracking attenuation and velocity, as a function of frequency, over time is presented. No record of service history was assumed. A simple numerical simulation, based on the Kelvin–Voigt model, guided by a scalar expression, matched experimental attenuation over the limited transducer frequency band and provided a viscoelastic damping coefficient, which is a measure of viscosity. Ultrasonic nondestructive evaluation was used on a sample of inert propellant at the 5th and 11th months after casting. A matched pair of 50 kHz transducers, wavelength in the material of 3.2 cm, were used in 120 tests. The preliminary experimental results indicate a decreasing attenuation and associated viscoelastic damping coefficient as time increases. Further testing will be used to ascertain an empirical relationship between attenuation, viscoelastic damping coefficient, and Young’s modulus over time, and their relationship to the safe life for motors.
An optimized finite-difference scheme for high-loss viscoelastic materials
Authors: Hayner Mark A. , Fricke J. Robert
Published: October 2005

In an effort to attenuate helical wave energy in cylindrical shells and reduce its coupling to the surrounding fluid, various constrained layer shell designs are being investigated. The shell skins being considered have an effective loss factor η of order unity. As part of this effort, an efficient time-domain finite-difference scheme is being developed with the ultimate objective of constructing 3-D models of such shells. Optimization of relaxation time spectra for low-η materials, done by Blanch etal. [Technical Report, Rice University, Dept. of Geology and Geophys. (1993)], is extended to high η by matching both the real and imaginary parts of the complex modulus with experimental data. To improve scheme accuracy further, the matching is done again after discretization to minimize the effects of numerical dispersion and dissipation. Data has been matched for a material with η=0.5 over one frequency octave using a Crank–Nicholson scheme with two relaxation times. The general applicability, accuracy, and efficiency of this numerical optimization process are subjects of continuing research. Comparison between numerical simulation and analytical solutions are given for wave propagation in a constrained layer flat plate. [Work supported by ONR.]
Long vertical arrays in acoustical sounding of the ocean
Authors: Voronovich A.
Published: October 2005

Using long vertical arrays (LVAs) in ocean acoustics enables one to measure an S matrix describing transformations of modes in inhomogeneous waveguides. The S matrix allows one not only to calculate the sound field for any configuration of sources and receivers but also to proceed to the solution of the inverse problem (IP)—retrieving inhomogeneities from acoustical data. The wave-type IP is studied for the 1-D case, which differs significantly from the multidimensional situation. For instance, in the 2-D case, the IP can be solved based only on the submatrix describing forward scattering as a function of frequency. The 2-D irregularities in the acoustic waveguide could be reconstructed as in the 1-D problem with the help of the layer-peeling algorithm via backscattering data. But this approach is of little practical use because backscattering is usually weak. Practical formulation of the deterministic IP in ocean acoustics should exploit some final-dimensional description of the media and use all available data about the S matrix. For example, using phases of diagonal elements measured at only two close frequencies leads to the Munk–Wunsch scheme of modal ocean tomography. In the general case, a numerical-evolution-type algorithm is proposed to solve this nonlinear problem in the spirit of the invariant imbedding approach. [Work supported by NRC and PPSIO of Russia.] a)On leave from the P. P. Shirshov Institute of Oceanology, Moscow, Russia.
Renormalization of propagation in layered media with rough interfaces
Authors: Berman David H.
Published: October 2005

There are now a number of good approximations for plane-wave scattering from rough interfaces separating two half-spaces. This work will describe how these approximations can be used when there are two or more rough interfaces. The technique to be developed is based on matching approximate scattering states, expressed in terms of half-space scattering amplitudes, across source planes. Boundary conditions at interfaces are satisfied by the scattering states. It will be shown how the resulting description can be renormalized so that the mean field is described in terms of effective reflection coefficients at the interfaces, which are distinct from coherent half-space scattering amplitudes. It will also be shown that the method of smoothing gives a first-order approximation to a fully renormalized mean field. Expressions for effective modal attentuations will also be derived.
Spectral integral formalism of wave propagation in a range-dependent waveguide
Authors: Tang Dajun , Guo Yue-Ping
Published: October 2005

This work is the study of acoustic wave propagation in a range-dependent waveguide, where either the interfaces or the medium parameters are functions of horizontal dimensions. For a range-independent, i.e., horizontally stratified, problem where the wave equation can be solved using separation of variables, the well-established wave-number spectrum formulation has been proved to be a powerful technique. When such a spectrum is known, the modal structure and its continuous component will completely determine the wave behavior in the waveguide. More importantly, knowledge of the spectral information can be used in inversion techniques to estimate environment parameters through acoustic probing. At the ASA Ottawa meeting [Tang and Guo, J. Acoust. Soc. Am. 93, 2284(A) (1993)], Tang and Guo presented the spectrum formalism and an application of the formalism to an ideal wedge. Here the approximate spectral formalism is further developed for more general weakly range-dependent problems. Special attention is paid to the ‘‘transition regions’’ of the waveguide where the discrete part of the spectrum borders the continuous part of the spectrum. Near such regions, different parts of the spectrum interact strongly with each other, resulting in apparent changes of the acoustic fields. Some examples will be discussed to elucidate the advantages of this approach.
Future directions in impulsive sound sources
Authors: Rynne Edward F.
Published: October 2005

While impulsive acoustic sources have long been used by the geophysical community for underwater exploration, sonar applications have been relatively uncommon. Recent work in the area of electric discharge devices has led to both a better understanding of the physics of this class of impulsive sound generator and to new devices, which may in turn lead to an expanded role for such technology in sonar applications—in particular, the observation that the electric arc commonly associated with sparker sound sources represents a wasteful and unnecessary complication. Proper electrode design and control of the electric discharge can eliminate the arc, leaving only a steam bubble, and can thereby enhance the low-frequency performance of such a device. Insight into the performance and potential of such devices is, in part, a result of improved computer modeling capabilities of the nonlinear processes associated with impulsive devices, as well as on the use of high-speed data acquisition in interpreting experimental results. This is especially true for determining the effects of interactions in arrays of bubble sources. Beyond the electric discharge sources, understanding of the coupling of energy from the bubble to the sound field suggests improvements for chemically driven sound sources as well. An update of work on these impulsive devices and the modeling efforts that support them is presented. The performance of some recently developed devices and the potential for future development will be discussed.
Development of a magnetostrictive sonar transducer using high-temperature superconducting coils
Authors: Joshi C. H. , Voccio J. P. , Lindberg J. F. , Clark A. E.
Published: October 2005

A low-frequency, dual-piston sonar transducer has been designed, built, and tested. This transducer uses a terbium–dysprosium magnetostrictive rod in combination with high-temperature superconductive (HTS) coils as the driver element. Terbium–dysprosium, when operated at cryogenic temperatures, has the highest known magnetostrictive constant. In this transducer, two HTS coils, driven by a combined dc and ac signal, excite the magnetostrictive element. The resultant motion is transmitted through the vacuum cryostat enclosing the driver assembly to two head masses, which transmit the sound to the surrounding environment. The combination of the high magnetostriction and the absence of electrical dissipation in the HTS coils results in very high overall system efficiency. Modeling of this type of transducer shows overall efficiencies of 80% or more can be obtained. The driver assembly is cooled by a compact Stirling Cycle cryocooler. Performance testing in air and water show self-resonance occurs at 520 and 430 Hz, respectively. The overall system design, as well as measurements of acoustic output and efficiency, will be presented. [This project was funded by The Naval Undersea Warfare Center, New London, CT, through the Small Business Innovative Research (SBIR) Program.]
Giant electrostrictive response of polymers
Authors: Scheinbeim Jerry I. , Newman Brian A. , Ma Zhen Yi
Published: October 2005

A new class of electroacoustic materials has been discovered at the Polymer Electroprocessing Laboratory, Rutgers University, based on the enormous electrostrictive response available in certain classes of polymeric materials. Further, many of the properties of these materials, including the dielectric constant and elastic modulus, can be tailored for specific end-use applications. The new material is ‘‘created’’ by applying a very large (≳10 MV/m), dc bias electric field using a superimposed ac driving signal to produce the required acoustic response. The measured thickness response dT, which is electrostrictive in nature, is proportional to the slope of the strain versus the applied electric field curve and is, therefore, proportional to the applied electric field up to saturation. Extremely high dT values (dT≳10 Å/V) can be accessed at high enough dc bias fields (≳20 MV/m).
Development of electrostrictive ceramic compositions for high-power sonar transducers
Authors: Ting Robert Y.
Published: October 2005

New emphasis on active sonar applications requires the development of compact, high-power sonar projectors. For low-frequency use, one desires to be able to drive the transducer elements at high electric field levels in order to achieve large strains. The conventional lead–zirconate–titanate (PZT) ceramics have been widely used for the design and fabrication of sonar projectors, but these ceramics suffer the shortcomings of strain limitation and nonlinear properties under high drive. In recent years, the Navy has been developing new electrostrictive ceramics of the lead–magnesium–niobate family, because these electrostrictive ceramics can produce much greater strains than the PZTs. In particular, a new class of lead–magnesium–niobate lead–titanate (commonly called PMN–PT) solid solution compositions has been found to offer the combined advantages of high-strain, low-dielectric, and hysteresis losses. The development of these materials is summarized, and the test results from prototype transducers fabricated using the new electrostrictive PMN–PTs are also presented. [Work supported by ONR/T.]
Reflections and diffractions from facet-wedge constructions: Acoustic experiments and theory
Authors: Li Saimu , Chu DeZhang , Clay C. S.
Published: October 2005

The higher-order diffractions of transient pressure signal from a combination of hard finite facet wedge of 45 deg and an infinite plane have been studied experimentally and theoretically. Experiments used a spark point source with a separated receiver at the same side of the wedge. The reflected pressure signals were well separated in time domain. Higher-order diffractions from the apex and troughs of the wedge were observed. The studies were concentrated on the reflections from the facets and the diffractions. The experimental geometry is different from Medwin etal. [J. Acoust. Soc. Am. 83, 1794–1803 (1988)]. Biot and Tolstoy’s exact wedge solution [J. Acoust. Soc. Am. 29, 381 (1957)] and extensions were used to compare experimental data and theory.
Simulation of propagation and scattering in a waveguide with a combined parabolic equation/extended boundary condition model
Authors: Zingarelli R. A.
Published: October 2005

By using the parabolic equation for range-dependent waveguide propagation, the extended boundary condition model for object scattering, and a new method for coupling the two models, basin-scale active acoustic simulations have been performed. The transformations and algorithms, as well as some results, will be presented. [Work supported by ONR.]
Loudness growth of complex stimuli in normal-hearing listeners
Authors: Lee Lidia W. , Humes Larry E.
Published: October 2005

This paper examined whether an excitation-pattern model of loudness could adequately describe the growth of loudness for complex stimuli presented to normal-hearing listeners in quiet and in noise. The loudness-growth functions were obtained for three synthesized steady-state vowels (/a,i,u/), each with two talkers (male: F0=120 Hz; female: F0=200 Hz), and for several pure tones (125–4000 Hz). All stimuli were presented, at random, from 2 to 90 dB SPL (in 4-dB steps) in quiet and in spectrally shaped noise. Magnitude estimation were used to measure the loudness of each stimulus. These data are used to evaluate the predictions of an excitation-pattern model [E. Zwicker and B. Scharf, Psychol. Rev. 72, 3–26 (1965)] and a simpler modified power-law model [L. Humes and W. Jesteadt, J. Acoust. Soc. Am. 85, 1285–1294 (1989)]. [Work supported, in part, by NIA.]
Word discrimination in various multitalker noises
Authors: Letowski Tomasz , Tornatore Andrea , Clark Jeff , MacBeth Barbara
Published: October 2005

A large number of different multitalker noises (MTNs) is used in research and clinical applications. These noises differ in acoustical, semantic, and linguistic properties and, thus, may have different masking effectiveness. The authors compared acoustical characteristics and masking effectiveness of several MTNs recorded by the authors and by others. Both spectral and temporal differences have been observed among various MTNs. When noises of similar acoustic characteristics have been compared, the gender of the talkers but not the speech material affected masking effectiveness of the MTN.
Characterization of viscoelastic materials using the resonant modes of a ‘‘free-free’’ bar
Authors: Brown David A.
Published: October 2005

The technique of using the resonant modes of a ‘‘free–free’’ bar to determine the elastic properties of the bar material will be described. A bar can be electrodynamically and selectively excited in three independent vibrational modes by mounting small coils of magnet wire on each end of the sample. When the sample ends are placed in a magnetic field, a current through one coil will induce vibrations that can be detected by monitoring the emf produced in a second coil. The elastic shear modulus is determined by measuring the frequency of the torsional mode. The Young’s modulus is determined from the frequency of either the longitudinal or flexural mode. The loss tangent, tan δ, of the material can be obtained by measuring the quality factor of each resonance. Results from a sample using a transfer function technique will be compared with results obtained using a phase-locked loop (PLL). The transfer function technique is convenient, since the voltage across the detection coil is proportional to the velocity of the sample and the current delivered to the excitation coil is proportional to the force delivered to sample. Their ratio is proportional to the impedance of the material. The lock-in method is attractive because it offers a means to continuously monitor the properties of the material.
Resonance apparatus for characterization of viscoelastic materials
Authors: Hartmann Bruce , Lee Gilbert F. , Lee John D.
Published: October 2005

A resonance apparatus for the characterization of viscoelastic materials will be described. Measurements have been made of modulus values as low as 0.1 MPa and as high as 70 GPa and loss factors from 0.003 to 2.5. Material properties are determined as exact (numerical) solutions of the wave equation with no simplifying assumptions. In particular, the loss factor is not calculated from the half-power point and is valid for large values of loss factor. By bonding the sample to the driver, clamping errors common with other instruments are eliminated. For any given measurement temperature, the frequency range of the measurements is about 1.5 decades centered at about 1 kHz. Depending on the sample length and modulus, data can be obtained from 100 Hz to 25 kHz. Once measurements have been made as a function of temperature, the data are shifted to form a master curve and fitted to an analytical model. Illustrative data will be presented for a high-loss and a low-loss polyurethane. [This work was sponsored by the Independent Research Program of the Naval Surface Warfare Center.]
Servohydraulic applications for vibration damping measurement
Authors: Biegler Kirk K.
Published: October 2005

In recent years, the range of performance has been expanded for servohydraulic test systems utilized for vibration damping measurement. The tests now include: (1) temperature, frequency, and amplitude effects on dynamic characteristics, (2) fatigue effects on dynamic or static properties, (3) resonant frequency determination, (4) stress relaxation after impact, (5) tearing energy analysis, and (6) static deflection measurement. The performance capabilities of servohydraulic test equipment readily available today will be reviewed. Performance capabilities include the range of loading amplitudes, rates, and frequencies and also the automation capabilities that relate to the ease of use and versatility of data analysis currently available. Some of the most recent improvements will be highlighted to show the extended range of tests that servohydraulics can achieve.
Aerodynamic and acoustic changes following intensive voice therapy for patients with Parkinson’s disease: A case study
Authors: Dromey Christopher , Ramig Lorraine O. , Scherer Ronald
Published: October 2005

A 50-year-old patient with Parkinson’s disease received a 4-week course of intensive voice and respiratory therapy aimed at improving functional communication through increased laryngeal adduction. Aerodynamic and acoustic data were gathered on two occasions before treatment, two after the completion of therapy, and at 6- and 12-month follow-up sessions. There were increases in subglottal pressure and laryngeal airway resistance after the treatment. Maximum flow declination rate (MFDR) also increased, as did the sound pressure level of utterances that the subject produced at a ‘‘comfortable’’ effort level. Fundamental frequency variability and conversational intensity levels were higher post therapy. Video-stroboscopic examination before and after treatment showed improvements in adduction and a decreased level of supraglottic hyperfunction. Possible physiological mechanisms are discussed to account for these changes. [Work supported by NIH Grant No. R01 DC01150-03 and NIDCD Grant No. DC00976.]
Spectral myoelectric analysis of thyroarytenoid activity in normal and spasmodic dysphonia subjects
Authors: Roark Rick M. , Dowling Eric M. , DeGroat Ronald D. , Watson Ben C. , Schaefer Steven D.
Published: October 2005

Time-synchronous digital recordings of thyroarytenoid (TA) myoelectric activity (via percutaneous hooked-wire electrodes) were obtained for 10 normal control and 11 spasmodic dysphonia subjects during performance of five vocal tract tasks of increasing motoric complexity: (1) quiet breathing, (2) Valsalva maneuver, (3) whispered ‖i‖, (4) voiced ‖i‖, and (5) Beepbeepwenttheheap. Three-dimensional power spectral density functions, PSD(f,t), were determined for a selected segment of the signals, and measures of median frequency, mean frequency, bandwidth, and center frequency of PSD(f,t) were derived for each sample point. Statistical median, mean, standard deviation, maximum, minimum, and mode of the PSD measures were computed to compose feature vectors for each signal. Discriminate analysis and maximum likelihood classification tests were applied to obtain global discrimination measures for the two subject populations as a function of task performance. Results of the analysis are reported that reveal a task effect. Graphical presentations of TA myoelectric and speech acoustic signals with accompanying time–frequency PSD plots are given. Comparisons are made among incidents of spasm and nonspasm, posing questions into physiologic meaning and basis of spectral- versus amplitude-defined laryngeal muscle spasms. [Work supported by NIH Grant No. DC00410 and NSF Grant No. MIP9203296.]
Comparison between clinician-assisted and computer-assisted methods for obtaining the voice range profile
Authors: Wong Darrell , Hensley Susan , Titze Ingo , Ramig Lorraine O. , Milder Martin
Published: October 2005

A study was made comparing two methods for obtaining the voice range profile, a clinical tool used to establish a subject’s range of intensity and fundamental frequency. One method used a clinician with a pitch pipe to elicit and motivate the subject. The other method used an instructional videotape and a computer. The computer was completely automated, providing visual feedback of the voice range profile, and evaluating each phonation objectively using the criterion of fundamental frequency stability, intensity stability, and duration. The computerized method used a cepstrum algorithm to estimate fundamental frequency, and the intensity was measured by calculating the variance of the signal. The results presented here compare the two methods and discuss the benefits and drawbacks of each approach. Benefits of the computerized method include real time visual feedback of phonatory effort, objective acceptability criterion, and no requirement to phonate at pitches in any particular order. A drawback to the computerized method is its sensitivity to subharmonics. [Work supported by NIH Grant No. DC000-387.]
Speech input for dysarthric users
Authors: Chang Hwa-Ping
Published: October 2005

One practical aim of this research is to determine how best to use speech recognition techniques for augmenting the communication abilities of dysarthric speakers. As a first step toward this goal, the following kinds of analyses and tests have been performed on words produced by several dysarthric speakers: a closed-set intelligibility test based on Kent etal. [J. Speech Hear. Disord. 54, 482–499 (1989)]; an open intelligibility test; critical listening and transcription; acoustic analysis of selected utterances; and an evaluation of the recognition of words by a commercial speech recognizer. The data from one speaker have been examined in detail. The analysis and testing have led to several conclusions concerning the control of the articulators for this speaker: production of obstruent consonants was a particular problem (only 30% of syllable-initial obstruents were produced with no error), whereas sonorant consonants were less of a problem (70% correct). Of the obstruent errors, most were voicing errors, but place errors for alveolars (particularly fricatives) were also high, and these consonants were produced inconsistently, as inferred from acoustic analysis and from low scores from the recognizer for words with these consonants. In comparison, vowel errors were less prevalent. Implications for the use of a speech recognizer for augmenting this speaker’s communication abilities are discussed.
Acoustic analysis and synthesis of pathological voice qualities
Authors: Long Christopher , Bangayan Phil , Alwan Abeer
Published: October 2005

An analysis-by-synthesis approach was adopted to classify the acoustic and perceptual features of three pathological voice qualities: breathy, strained, and rough. One hundred and sixty waveforms of the vowel /open aye/ spoken by female and male subjects with pathological voice qualities were obtained from the VA Hospital in West LA. The temporal and spectral features of the waveforms were studied and the results were used in synthesizing the utterances using the Klatt formant synthesizer. Preliminary results on 30 breathy and strained voices indicate that the perception of ‘‘pathological’’ breathiness is mainly related to: (1) a large open quotient of the glottal waveform (OQ) and (2) the amplitude of aspiration noise (AH) relative to that of voicing (AV) with female voices exhibiting a larger (AH–AV) than male voices. For some voices, it was also necessary to introduce extra poles to the vocal-tract transfer function to achieve a better spectral match. Synthesis of strained voices required a lower OQ than that needed for normal voices and, in some cases, amplitude and/or frequency modulation was introduced to achieve a better match in the time domain. The synthetic voices were judged perceptually by clinicians to be of high quality. The results will be discussed in terms of the effects of different vibratory patterns of the vocal folds on the acoustic speech waveform.
A sparse boundary-element method for scattering from a rigid body of revolution
Authors: Rosen Erik M. , Couchman Luise S. , Canning Francis X.
Published: October 2005

A sparse boundary-element method is used to calculate the scattered acoustic field of a rigid body of revolution. The N×N full matrix resulting from the discretization of the Helmholtz integral is transformed to a sparse N×N matrix using the impedance matrix localization (IML) method. When applied to integral equations for wave phenomena, the IML method produces a sparse matrix by partitioning the scatterer into several regions, over which the response is represented by basis functions with directional radiation patterns. The number of important physical interactions is reduced to a small number over each region, leaving a large number of matrix elements with relative magnitudes less than 10−4 which can be approximated by zero. Used in conjunction with existing BEM codes, the IML method reduces the memory and storage requirements from N2 to approximately 100N, while the solution time is O(N) compared to O(N3) for the full untransformed case. The resulting sparse matrix can be approximately inverted leading to a convergent solution in less than five iterations. Results are given for scattering from a hemispherically endcapped cylinder of L/a=110 for ka=1 to ka=17.
Predicting bistatic target scattering from monostatic data
Authors: Schenck Harry A.
Published: October 2005

Characterizing bistatic scattering from a target for many angles of incidence is expensive and time consuming even in ideal laboratory conditions, and essentially infeasible in an operational environment. Consequently, extending the value of monostatic data by using it to estimate the bistatic target strength is extremely desirable. Previous efforts to do this have severe limitations or restrictions in their applicability. A new technique that combines numerical modeling and experiment will be described. The crux of this method is to rely on modeling to supply the far-field propagator function that predicts the scattering in any direction when the surface values are known, and to determine the surface values by least-squares approximation from knowledge of the monostatic scattering and the principle of reciprocity (which requires the scattering matrix to be symmetric). An example will be shown in which the complete three-dimensional scattering field was estimated for a rigid axisymmetric body given only the monostatic target strength at angles in one plane from 0 to 180 deg. Rules that relate the number of monostatic observations needed to the frequency and the complexity of the target will be described. The extension of this technique to nonrigid scatterers will be discussed.
Localization performance is enhanced under bimodal conditions of the auditory and visual spatial channels
Authors: Costantino Brian L. , Perrott David R.
Published: October 2005

A sequential localization task compared the effects of unimodal versus bimodal spatial acuity in the auditory and visual perceptual systems. The four subjects were tested in a two-alternative forced-choice, three down–one up adaptive paradigm in which two 200-ms signals were presented sequentially with a 1000-ms inter-stimulus interval. Tests were administered with a 620-nm light emitting diode at a luminance level of 200 ml and a 1.0-kHz high-pass noise at 57 dB (A-weighted). The minimum bimodal angle (MBA) threshold was compared to that of the minimum visible angle (MVA) and minimum audible angle (MAA) thresholds and found to increase performance significantly over the single mode localization tasks, except at 0-deg azimuth that equaled the MVA for most subjects. The implications of these results are discussed.
Performance analysis of transient acoustic wave modeling using higher-order WKBJ asymptotics and symbolical manipulation
Authors: Verweij Martin D.
Published: October 2005

The method of modeling transient acoustic waves in continuously layered media using a combination of higher-order WKBJ asymptotic representations and the Cagniard–De Hoop method is addressed. The coefficients of the WKBJ asymptotic representations satisfy a recurrence scheme, which is easily dealt with by applying symbolical manipulation. Moreover, an efficient use of the Cagniard–De Hoop method of inversion is achieved [M. D. Verweij, J. Acoust. Soc. Am. 92, 2223–2238 (1992)]. The performance of this method is discussed on the basis of numerical results for a series of configurations. In particular it is shown how the method performs if transcendental parameter profile functions are replaced by least-square-fit polynomials. From the results it is concluded that the method performs best for low-order polynomial parameter profiles.
The effects of level and frequency region on the recovery of overshoot
Authors: Overson Gary J. , Bacon Sid P.
Published: October 2005

The term ‘‘overshoot’’ refers to the finding that the threshold for a brief signal can be higher when that signal is presented shortly after masker onset than when it is delayed by a few hundred ms. Overshoot can be reduced or eliminated by presenting a stimulus (precursor) prior to masker onset. The ‘‘recovery of overshoot’’ can be examined by varying the delay between the offset of the precursor and the onset of the masker. The purpose of experiment 1 was to evaluate this recovery at two (masker and precursor) levels, one relatively high and one relatively low. Overshoot was first measured as a function of masker level in order to choose two fairly disparate levels that produced nearly equivalent amounts of overshoot. Recovery functions were then measured at those levels. In general, there was little difference in the recovery at the two levels. In experiment 2, the precursor was sectioned into two bands, one on either side of the 4.0-kHz signal frequency. Recovery was measured separately for each band, while the other band was presented continuously. There was a tendency for the recovery to depend more upon the band above the signal frequency, suggesting that that region may be most important for overshoot. [Work supported by NIDCD.]
Modulation detection interference (MDI) for randomly fluctuating envelopes
Authors: Mendoza L. , Hall J. W. , Grose J. H.
Published: October 2005

Modulation detection interference (MDI) was measured in normal-hearing listeners for stimuli with randomly fluctuating envelopes. The stimuli were generated by modulating pure-tone carriers by dc-shifted low-pass noises (cutoff=10 Hz). The target carrier was 1000 Hz and the interferer carrier 2250 Hz. The target could be presented either alone (baseline condition) or in conjunction with the interferer that was either gated synchronously with the target or was presented continuously. The interferer could be unmodulated, comodulated with the target, or modulated independently from the target. Results indicated substantial MDI in the presence of the interferer both for comodulated and independently modulated envelopes. The differential gating effects will be compared to those for sinusoidal amplitude modulations. [Work supported by the NIDCD R01-DC00418.]
Some effects of modulator phase on the detection of a signal in a sinusoidally amplitude-modulated tonal complex
Authors: Hicks Michelle L. , Bacon Sid P.
Published: October 2005

Subjects were required to detect a 1000-Hz signal in the presence of a masker that consisted of a 1000-Hz (on-frequency) component alone or that component and six flanking components (500, 600, 700, 1300, 1400, and 1500 Hz). The on-frequency component typically was sinusoidally amplitude modulated at 10 Hz and the signal was presented in a dip in the envelope of the on-frequency component or in three successive dips. In experiment 1, the flanking components could lower (comodulation masking release or CMR) or elevate (across-channel masking or ACM) threshold, depending upon whether they were modulated in phase with the on-frequency component, or staggered in phase. CMR was observed for both signal types, although it was smaller for the three-burst signal. ACM was only observed for the three-burst signal. These differences were primarily due to lower thresholds in the presence of the on-frequency component alone for the three-burst signal. In experiment 2, thresholds were measured as a function of the modulator phase of the flanking components. The on-frequency component was unmodulated or 100% amplitude modulated (modulator phase of 0°). For both monotic and dichotic presentations of the flanking components, thresholds increased and then decreased as the modulator phase of the flanking components varied together from 0 to 360°. [Work supported by NIDCD.]
The effects of cross-spectrum envelope correlation on detection of a signal band: A comodulation detection difference (CDD) experiment
Authors: Rappold Patrick W. , Carter John R.
Published: October 2005

Most CDD and comodulation masking release (CMR) studies employ only two conditions; a correlated condition and an uncorrelated condition. Temporal envelopes of noise bands are identical in the correlated condition and degree of envelope correlation is the same (r=1.0) across trials within this condition. In the uncorrelated condition envelopes of the noise bands are not identical, but degree of envelope correlation fluctuates across trials within this condition. Thus, reported thresholds for correlated conditions represent the average threshold at a single degree of correlation, whereas reported thresholds for uncorrelated conditions represent the average threshold across various degrees of envelope correlation. In this study thresholds are determined for signalbands centered at 2000 Hz, in the presence of cuebands centered at 1500 Hz, at specific degrees of temporal envelope correlation (r=1.0, 0.75, 0.60, 0.50, 0.40, 0.25, 0.0, −0.25, −0.35). Thresholds at r=1.0 and 0.75 are about 10 dB higher than thresholds at r=0.25, 0.0, and negative degrees of correlation. Thresholds at r=0.50 are about 5 dB less than thresholds at r=0.75 and above, and about 5 dB greater than thresholds at r=0.25 and below. Thresholds at r=0.60, 0.50, and 0.40 are nearly equivalent. These findings suggest that at least three conditions (ranges of correlation) should be employed in CDD, CMR, and other studies investigating the influence of envelope correlation.