bluuebunny/crossref_metadata_2025 huggingface dataset:
Rollover with high-frequency emphasis
Authors: Rankovic Christine M. , Zurek Patrick M.
Published: October 2005

For individuals having high-frequency sensorineural hearing loss, restoring the audibility of the entire speech spectrum via amplification [maximizing the articulation index (AI)] can require a frequency-gain characteristic with 25 dB or more gain at high than at low frequencies. Contrary to AI predictions, both Skinner [J. Acoust. Soc. Am. 67, 306–317 (1980)] and Rankovic [Ph.D. thesis, University of Minnesota (1989)] have shown that this type of amplification is associated with a reduction in speech intelligibility scores (rollover) when compared to conditions having less relative high-frequency gain. Since normal-hearing subjects also demonstrate this effect, we tried to determine whether the degradation for them is due to the strong and abrupt high-frequency emphasis, or simply the value of gain at high frequencies. Subjects were presented with nonsense syllables in a speech-shaped noise background under two conditions of high-frequency emphasis (both having 45 dB more gain at high than at low frequencies) as well as a flat condition with the same spectral levels at high frequencies, two signal-to-noise ratios, and various overall levels. When results are plotted as a function of high-frequency gain, the flat and high-frequency emphasis conditions demonstrate nearly identical consonant-correct scores, and similar degrees of rollover. This suggests that, at least for normal-hearing subjects, excessive presentation level rather than spectral imbalance is the primary determinant of the rollover effect with strong high-frequency emphasis. [Work supported by NIH.]
Perception of normal and spectrally compressed speech by the skin
Authors: Hurtig Richard R.
Published: October 2005

The development of sensory aids continues to be focused on increasing the number of channels and on alternative processing schemes. The present study compares natural hVd syllables and simple sentences presented with varying degrees of spectral compression [R. Hurtig, J. Acoust. Soc. Am. Suppl. 1 85, S44 (1989)] under vibrotactile and electrocutaneous stimulation conditions. In addition, a comparison of the number of channels used was made. Confusion matrices were constructed from the responses made in a closed set identification task, which followed a subject-paced exploratory learning task. Multidimensional scaling of the stimuli based on their first three formant values produced the standard representation against which the perceptual data could be evaluated. This congruence analysis graphically illustrates the distortions of the perceptual vowel space. An overall shrinkage of the perceptual space is evident across conditions; however, this effect is not uniform. Warping of the space is observed but in different portions of the space for each of the stimulation conditions.
Continuing evaluation of a synthetic Tadoma system
Authors: Rabinowitz W. M. , Henderson D. R. , Reed C. M. , Delhorne L. A. , Durlach N. L.
Published: October 2005

A synthetic Tadoma system has been developed for use as a research tool in studying the highly successful Tadoma method of tactual speech reception [Reed et al., J. Acoust. Soc. Am. 77, 247–257 (1985)]. In the synthetic system, signals recorded from a talker's face are used to control an artificial mechanical face. Recently, an experienced deaf-blind Tadoma user was tested on open-set CID sentence reception; he obtained about 20% keyword identification. Although this score is well below his performance on natural Tadoma (about 85%), it is clearly above chance and was obtained after only about 10 h experience with the synthetic system. Using an improved version of the system, two normal subjects, artificially deafened and blinded, trained on the identification of consonants and vowels, initially using small sets of stimuli with feedback and working up to larger stimulus sets. Asymptotic performance reached values of 64% correct for a set of 24 consonants and 67% correct for a set of 15 vowels and diphthongs, scores similar to that for natural Tadoma. Taken together, these results imply that the synthetic system has captured much of the information available in Tadoma. [Work supported by NIH.]
Adequate sample size for representation of speaking fundamental frequency and its standard deviation in subjects with vocal pathology
Authors: Hamilton Claudia , Stanford Michelle , Trudeau Michael
Published: October 2005

In comparing the speaking fundamental frequency (SFF) for individual sentences to SFF derived from the entire “rainbow passage” (Fairbanks, 1960), Horii (1975) concluded that a sufficient correlation (r = 0.985) existed between sentence two's and the passage's SFF to justify use of only sentence two for SFF analysis. He cautioned, however, that use of single sentences to represent standard deviation (SD) of SFF should be avoided as the resultant correlations with the entire passage were too low. The purpose of the present investigation was to extend Horii's analysis of normally produced voice to voice of subjects with vocal pathologies (14 adult women with mass related lesions to the vocal folds). SFF analysis of these subjects' readings of the “rainbow passage” was completed using a microcomputer-driven speech editor. The results were used to correlate single sentence samples and multiple sentence samples with the SFF and SD derived for the entire passage. The proposed poster will display the correlation matrices. The most salient findings were that for SFF two sentences (4 and 5) comprised the smallest sample with a correlation <0.985 and for SD the entire passage minus the second sentence was the smallest sample to produce such a high correlation.
Speaking without a tongue
Authors: Gerdeman Bernice , Fujimura Osamu
Published: October 2005

A 45-year old male with normal speech underwent a total glossectomy with myocutaneous flap reconstruction for rebuilding the posterior floor of the mouth. Speech therapy was initiated one month post surgery. Several prosthodontic devices were made to help improve speech intelligibility and deglutitory function. After several months of practice with a prosthesis, improvements of specific speech sounds including the vowel [i] were attempted by further revising the design of the prosthesis. The surgically reconstructed area seems to be capable of very limited volitional movements. In assessing the extent of movement of the flap reconstruction, x-ray microbeam recordings were obtained before and after a major revision of the prosthesis. Articulation of selected sounds were considerably improved. Although articulation is not perfect, conversational speech is highly intelligible, including distinguishable alveolar and velar consonants and improvements on vowel production. Further revision of the prosthesis and practicing are being performed. Audio demonstrations will be given and results of simple formal assessment of intelligibility as well as “articulatory” and acoustic assessment will be discussed with interpretation of specific articulation strategies.
Acoustical design of large multipurpose stadiums and arenas
Authors: Wrightson Jack , Johnson Jim
Published: October 2005

The variety of events presented in modern sports arenas has created a situation where a range of optimal acoustical characteristics is expected. These requirements and potential acoustical design approaches will be discussed, taking into account the operational requirements of professional sports and large scale touring shows. A concept of local energy density is presented, which defines the relationship between loudness, reverberation, and liveliness with respect to various event types. The design approach is concerned with maximizing energy densities in spectator areas while achieving low overall reverberation time. Geometric and variable acoustical elements required to achieve the best relationships among loudness, reverberation, and speech intelligibility are described. The introduction of both house and touring sound systems to this environment is considered with regard to reverberation time, specular reflections, and speech intelligibility. The requirement for high SPL loudspeaker systems, including configuration, location, and operational flexibility are discussed. Examples of successful implementation of novel loudspeaker placement are shown. A process for predicting speech intelligibility of sound systems in very large spaces (domed stadiums), accounting for propagation losses, is presented.
Interdependent impact of acoustics, psychoacoustics, electroacoustics, and usage on design of arena facilities
Authors: Smith Howard G.
Published: October 2005

Current and future usage priorities arc explored for large arena facilities. The impact of such usage is a significant factor in determination of acoustic and psychoacoustic criteria appropriate for the type and relative number of projected arena events (basketball, concerts, hockey, etc.). Usage-related parameters that affect acoustics and the sound reinforcement system design are room shape, volume, type and distribution of materials, HVAC ambient noise exterior noise intrusion (overflight), crowd noise for various event types, and associated crowd size. Speech intelligibility, frequency response, peak sound level, uniformity of coverage, time delay effect, and qualitative issues arc discussed in conjunction with these usage related design parameters. Design constraints imposed by the acoustics of large arenas are discussed as they relate to arena usage, operation, and sound reinforcement system configuration. Specific examples will be cited from experience with McNichols Arena (Denver), Arco Arena (Sacramento), and with current design projects such as the San Jose Multipurpose Arena (San Jose).
Statistical models for pole-zero distributions in the complex frequency plan—Reverberant amplitudes and reverberant phase
Authors: Lyon Richard H.
Published: October 2005

Statistical energy analysis (SEA) is a way of estimating transfer function magnitudes in complex, multi-degree of freedom (dof) systems. The system model is a Poisson distribution of poles along the pole line, with residue statistics determined by the mean and mean-square values of the mode shape functions. The model gives the average and variance of the response amplitudes, and the effect of modal overlap on the statistics. On the other hand, the phase of transfer functions is determined by both the poles and zeros of the transfer function, and is important for determining group delay and distortions of waveforms. Since the zeros lie throughout the complex frequency plane, their distribution is two dimensional. The distribution of zeros normal to the pole line is the Cauchy distribution. This modeling is part of “statistical phase analysis” (SPA). The concept of “reverberant phase,” which has many corrolaries to the idea of reverberant amplitudes, arises quite naturally from the model. This paper will expand on these ideas and their potential applications. [Work supported by NTT Human Interface Laboratory, Tokyo.]
A statistical analysis of fluid-loaded structures
Authors: DeJong R. G.
Published: October 2005

A statistical model of the dynamical response of fluid-loaded structures is presented. Special emphasis is given to fluid-loaded, ribbed plates excited by turbulent boundary layer pressure fluctuations. The statistical model is based on the dispersion relationship for an infinite, fluid-loaded plate and the associated wave types. The frequency distribution of modes in a finite structure is represented probabilistically. Justification for the probabilistic model is based on the observed distribution of modes in simple structures with statistical variations of parameters and in complex structures. The dynamical responses of the system are determined by the flow of energy through the system. The other response variables, such as acceleration or pressure, are derived from the energy levels. The model can be generalized to handle orthotropic plates and ribbed cylinders. [Work supported by ONR.]
Modal density and impedance formulations for cylindrical shells and pipes
Authors: Manning Jerome E. , Quartararo Louis R.
Published: October 2005

As shown by Heckl [J. Acoust. Soc. Am. 34, 1553–1557 (1962)], the modal density for a cylindrical shell can be obtained from simplified thin-shell equations, and can be used to obtain the conductance—the real part of the drive point mobility. The use of modal densities to obtain impedance results has long been known and provides a key concept in statistical methods of vibration analysis. This paper expands on Heckl's basic concepts by obtaining more accurate modal densities for cylindrical shells and pipes, and by using these modal densities to determine both the real and imaginary parts of the mobility. The distinction between cylindrical shells and pipes is based on the length-to-diameter ratio. The modal density for pipes, which have a large length-to-diameter ratio, is best determined as a sum of the modal densities for the individual circumferential modes. The use of such a summation results in modal densities with large peaks at each circumferential mode cut-off frequency.
Space-averaged energetics of membranes and plates
Authors: Bouthier Oliver M. , Bernhard Robert J.
Published: October 2005

The space-averaged energy density and structural intensity of plates and membranes are investigated. When small damping is assumed, the structural intensity is proportional to the gradient of the space-averaged energy density. Furthermore, if a hysteretic damping model is assumed and energy is conserved, the space-averaged energy density can be modeled using a second-order partial differential equation. The resulting models of energy and intensity are good statistical predictions of vibration at medium and high frequency. The equations have been verified for several classical membrane and plate problems. [This investigation was supported by the NASA Langley Research Center.]
Frequency domain scattering effects of multiple subsystems
Authors: Igusa Takeru , Min Kyung-Won , Achenbach Jan D.
Published: October 2005

The effect of a large number of subsystems on the harmonic and wide-band response characteristics of their supporting master structure is of interest. A modal approach is used, and attention is on the coupling effects between each master structure mode with n subsystem modes with closely spaced natural frequencies. The mathematics centers around an n + 1 complex eigenvalue problem that is handled by using series expansions and asymptotic techniques. It is found that by making certain assumptions about the subsystem modes, the properties of each master mode becomes shared and distributed among all coupled subsystem modes resulting in a frequency scattering effect. This effect is well known in experimental work, and one contribution of this study is in providing a mathematical solution that makes the physics of the problem transparent. The results show that the subsystems have a smooth effect on the master structure, which makes the problem amenable to statistical methods. Simple expressions were derived for averaged wideband response that quanitifies the damping effect of large numbers of subsystems on the master structure. [Work supported by ONR.]
Improving SEA response predictions with better estimates of input power
Authors: McCarthy Daniel J. , Scott Brian L. , Manning Jerome E.
Published: October 2005

A beam system consisting of six beams of varying lengths was analyzed using both statistical energy analysis (SEA) and a deterministic technique to determine whether an improved estimate of SEA input power would lead to more accurate predictions of vibrational response. The exact input power calculated with the deterministic technique was used to adjust the vibrational energy levels predicted with SEA. Comparisons between the deterministic predictions and the SEA predictions indicate that the use of exact input power does lead to better SEA predictions. The results show that the entire system does not have to be modeled to improve the input power estimate—large portions of the system can be modeled with a single semi-infinite beam. The adverse effect of local vibrational modes on the adjusted SEA predictions is examined.
A study on the effect of system fluctuation on transfer function measurement
Authors: Kinoshita Akio , Lyon Richard H.
Published: October 2005

The influence of system fluctuation on transfer function estimation has been examined in relation to establishing more realistic simulation models for the development of quieter machinery. The major concern here is to give answers to the following questions: (1) Is there any substantial effect on transfer function estimates caused by system fluctuation? (2) If there is, how does it affect measurements taken using FFTs? (3) How can one know whether there is considerable influence or not? (4) And finally, how can one exclude the effects of fluctuations from system identification? Statistical parameters were introduced into a single degree of freedom mass-stiffness-damper model for carrying out computer simulation. It was found that fluctuations could cause overestimation on damping and underestimation on mass and stiffness. Therefore, a system could be misidentified because of deviations in its parameters. This result agrees with experimental data in general. The possibility of recovering correct system parameters in the presence of fluctuations will also be discussed.
Ralph Woollett and the art of transducer design at USNUSL
Authors: Sherman Charles H.
Published: October 2005

Ralph Woollett spent most of his life studying the subject of electroacoustics. He approached the subject with the attitude of a physicist, trying always to understand and to relate what he knew to basic physical laws. But he was also an engineer, analyzing the behavior of new transducer designs and creating new design concepts such as trilaminar flexural transducers and magnetostrictive drive of flexural bars. It was the good fortune of his colleagues that he also devoted much of his time to teaching them his way of looking at electroacoustics and suggesting interesting and relevant problems for them to work on. He was active in ASA and IEEE committees and was a U.S. delegate to the International Electrotechnical Commission's Subcommittee on Ultrasonics. Ralph's untimely, accidental death reminded us of a lesser known side of his personality, for he was an avid flyer and somewhat of an adventurer who was always seeking new experiences and challenges.
The development of low-frequency trilaminar bender bar transducers
Authors: Gonzalez Manuel A.
Published: October 2005

Acoustic underwater projectors using bender bar technology are widely used as sound sources in low-frequency sonar applications. In the early 1980's, the Marine Systems Division of Honeywell in Everett, Washington, under contract with NUSC, initiated a development program to dramatically improve the acoustic performance and reliability of low-frequency bender bar projectors for a submarine sonar application. This contract provided the vehicle for an extremely fruitful interaction with Ralph Woollett, and resulted in the development of the trilaminar bender bar concept now used in the new family of high-power, high-efficiency, bender bar transducers built by Honeywell over the past few years. Ralph made significant contributions to our theoretical understanding and modeling of these transducers. He also set a personal example of technical excellence and integrity that motivated all of us who were involved with him in this project. In this paper, the theory of operation of trilaminar bender bars will be summarized, the major performance benefits that they offer will be presented, and some recent work in the low-frequency transduction area will be described.
Low-frequency hydrophone calibration
Authors: Geleskie Mark E. , Hughes W. Jack
Published: October 2005

Low-frequency underwater acoustic calibration of hydrophones in finite semireverberant tanks has always been desired. However, for a given tank, the physical dimensions determine the lowest frequency measurement possible using standard short-pulse calibrations. A long-pulse technique has been used to extend calibrations from the short pulse limit of 5 kHz down to 1–2 kHz. It is known that if the acoustic impedance of a transducer is high enough so that its radiation impedance can be neglected (such as at very low frequencies) and if the transducer is also small enough so that diffraction effects can be neglected, then the transducer's receiving sensitivity will be the same in air as it is in water. This property is used to verify the accuracy of calibrations made in an acoustic tank at low audio frequencies using both the traditional short-pulse calibration method and the long-pulse technique. This paper compares the air and water calibrations of hydrophones, showing the validity of the long-pulse method and the agreement with the short-pulse measurement technique.
Using stroboscopic photography to study the motion of a plucked cord
Authors: Winters Loren M.
Published: October 2005

The motion of a stretched, elastic cord, plucked midway between fixed ends, has been investigated using stroboscopic photography. A computer was used to trigger up to four electronic flash units in rapid succession. Countdown subroutines were used to set two time intervals: (1) a delay between passage of the center of the cord through an infrared photogate and discharge of the first flash, and (2) the interval between successive discharges of the flash units. Time intervals as short as 20 ̈s could be set, although, in practice, the intervals were in the range of 0.1–1 ms. The cord, which was marked at equal distance intervals, was photographed at equal time intervals throughout the first cycle of motion after being plucked. Transverse and longitudinal displacements of the marks were obtained as a function of time from digitized records of the photographs. Changes in tension were indicated by the separations of the marks. The results for the first eighth cycle can be explained in terms of the superposition of the triangular pulses moving in opposite directions on the cord. For the rest of the cycle, more complex waveforms develop as the tension in the cord changes. The apparatus will be demonstrated and photographs shown.
Accessible methods for measuring the resonance properties of musical instruments
Authors: Corliss Edith L. R.
Published: October 2005

Increased availability of amplifiers that supply precise time delays for introducing “ambience” effects in music reproduction provides a useful resource for the study of musical instruments. The circuitry provides simultaneous comparison of two parts of a resonance decay separated by a precise time interval. From these data, logarithmic decrement may be measured. For single-frequency tones, the figure of merit “Q” at that frequency can be ascertained. Examples of the use of these processes will be shown.
A measurable effect of prolonged vibration on the frequency relation of two important modes in violins
Authors: Hutchins Carleen M.
Published: October 2005

It has been found that in a finished violin, the frequency spacing (ΔF) between the first cavity mode (A1) above the so-called “Helmholtz air” mode and the first large-amplitude mode (B1) of the violin body, particularly the top, provides a measurable clue to its tone and playing qualities. When this ΔF is greater than 65–70 Hz, violins have been rated by players as “harsh” and often “hard to play,” while violins with lower ΔF's are usually preferred. Methods of reducing this ΔF are being explored, particularly in the finished instrument. Prolonged vibration of several violins has caused a decrease of between 10–20 Hz in the frequency of the B1 mode, thus decreasing the A1–B1 ΔF. Experiments showing these changes will be discussed along with implications for long-term violin playing and restoration practices.
Computer-simulated effects of local wood removal on the first mode frequencies of a violin top plate supported at the edges and the soundpost
Authors: Rodgers Oliver E.
Published: October 2005

The acoustical output of the first top plate mode of an assembled violin is usually the largest of any mode in the first 1000 Hz. The frequency of this mode is usually quite near that of the second natural frequency of the air cavity. Since the frequency relationship of these two modes has been found to be an important factor in top plate tuning and in violin quality, it would be desirable to be able to fine tune the top plate mode frequency after the instrument has been assembled and the critical frequencies have been determined. A finite element program has been used to determine the most sensitive top plate areas where wood removal would reduce the top plate frequency. In the program, no out-of-plane motions were permitted at the edges of the plate and at the sound post and no rotations at the end and corner blocks. The most sensitive areas are found to be at the ends of the bass bar, at the upper end of the F hole on the bass bar side, and at the sound post.
Comparisons of computer simulation and the effect of actual wood removal on the first large-amplitude body mode of a violin
Authors: Hutchins Carleen M.
Published: October 2005

Computer simulation of a violin top supported at the edges and with rigid soundpost indicates several areas where the frequency of the first large-amplitude body mode (B1) is particularly sensitive to wood removal. Effects are reported on B1 frequency changes with wood removal in these areas from a real violin top with resulting changes in the assembled instrument.
Musical pitch tracking with a narrowed autocorrelation function
Authors: Brown Judith C. , Zhang Bin
Published: October 2005

An autocorrelation function with narrowed peaks can be obtained by including, in addition to the usual product f(t)f(t + τ), products of the form f(t)f(t + 2τ), f(t)f(t + 3τ),…, f(t)f(t + [N − 1]τ) [J. C. Brown and M. S. Puckette, J. Acoust. Soc. Am. 85, 1595–1601 (1989)]. This narrowing is useful for the identification of the peak position in the presence of nearby frequency components and in some cases for resolution of these components. Algorithms will be presented for obtaining a narrowing of either the usual positive peaks of the autocorrelation function or those of an inverted autocorrelation function [J. C. Brown and M. S. Puckette, Proc. 1987 Int. Conf. Comput. Music, Urbana, IL, 84–88 (1997)]. This latter function has some advantages for use in pitch tracking. These calculations will be applied to a variety of musical examples including the violin, flute, and piano, for the purpose of obtaining the pitch of digitized waveforms. These pitch tracking results will be presented for sampling rates of 10 K and 32 K and relative advantages will be discussed. [Work supported by a Brachman Hoffman Fellowship from Wellesley College.]
Ear-adequate spectral analysis for noise evaluation
Authors: Genuit Klaus
Published: October 2005

In numerous sound situations, sound quality cannot be determined by means of conventional measuring techniques. The frequent discrepancy between subjective judgment of sound events yielded by standard measurement procedures, can be explained by the difference between a sound situation as received by a single omnidirectional microphone, on the one hand, and as received by the human hearing, on the other hand. A new procedure using a dummy head and a special binaural analyzer has been developed for spectral analysis that realizes frequency and time resolution in the entire audio spectral range. This is comparable to the properties of human hearing, which is able to perceive quickly changing time structures and minor frequency differences at the same time. New approaches for ear-adequate noise evaluation in order to judge noise quality can be achieved by means of psychoacoustic properties of human heating and binaural signal processing. The results of these investigations show that human hearing, which has two auditory paths with a complex directional characteristic and a binaural pattern recognition, gives a different judgment of noise quality as compared to conventional measurement methods, especially in complex sound situations with a spatial distribution of several sound sources and a sound-pressure level below 85 dB.
Signal processing techniques for identifying sound quality parameters
Authors: Laux P. , Davies P.
Published: October 2005

Importance is now being placed on the type of sound that machinery makes rather than simply on the noise level it produces. Two reasons for this emphasis are: the adverse affect an unpleasant acoustic environment can have on workers' performance; and the link a consumer makes between the quality of a machine and the sound it makes. The current problem is to find objective measures that reflect this subjective quantity known as sound quality. It is important to define such a quantity based on objective measures so that design objectives or performance specifications can be formulated rigorously. Single number indicators that are in use now include loudness, speech interference level, sound-pressure level (A-E weightings), and impact noise ratings. In the automobile industry, aspects of sound quality such as muddines, roughnes, and linearity have been shown to be correlated with the relative power of harmonic components in a spectrum and also to the statistics of the time history envelope. However, these indicators can yield similar values for signals that sound very different and cause a different subjective response. In order to identify the relationship between different signal components and the sound quality rating, techniques that can alter particular components of signals are needed to generate playback signals for subjective testing. In this paper, signal processing techniques are described that have been used to vary noise signal parameters so that components that contribute to sound quality can be identified.
Microcomputer-based implementation of the speech transmission index for large spaces
Authors: Jacob Kenneth D. , Birkle Thomas K. , Ickler Chris B.
Published: October 2005

The speech transmission index (STI) has been shown to be an accurate predictor of speech intelligibility over a wide variety of conditions [T. Houtgust and H. Steeneken, J. Acoust. Soc. Am. 77, 1069 (1985)], and the computationally more efficient RASTI method is being considered as an IEC standard. Instruments have been developed by others to measure the STI after room construction is complete and the sound system is operating, but the method has not been implemented in a program for predicting sound system performance. Thus the method has not been used to predict the intelligibility of unbuilt or in-process designs. A new microcomputer-based implementation of the STI method will be described along with the results of an experiment designed to verify its accuracy. A database of ten reverberant rooms was used to compare actual intelligibility scores using human subjects to intelligibility scores predicted by the new computer-based STI program.
Acoustical conditions in large stadia and interface of speech/music electronic reinforcement systems thereto
Authors: Dickensheets Kenneth R.
Published: October 2005

Due to the volume and shape of large covered and semicovered stadia, acoustical and noise conditions are often poor. Sound system support requirements for the many types of events held in these stadia have become more rigorous due to increased audience demands and expectations and also because of the nature of event types held therein. The interface of a permanent, music quality, sound reinforcement system to the acoustical conditions of a multi-use, professional football stadium is presented. Measurement and analysis of reverberation time, background noise levels, and spectra and echo conditions are presented along with their effects on the operation and audience acceptance of the sound system. Description of the system, including design details incorporated to maintain source illusion, provide nondelayed field/performance area monitoring and various other functional and performance particulars are presented.
Electrorheological fluids
Authors: Duclos Theodore G.
Published: October 2005

Electrorheological fluids (ER fluids) are suspensions of fine particles dispersed in an insulating fluid. If the properties of the particles and the fluid are selected properly, the resulting ER fluid can have the remarkable ability to flow like a liquid in its normal state, yet freeze into a gel in an electric field. The controllable property allows engineers to design devices that have unique capabilities in the field of motion and vibration control. Tunable mounts, continuously variable clutches, and tunable structures are all devices that can use ER fluids. The current state of the art is such that the ER fluids are being improved to meet the demands of the various device applications. The link between the necessary ER fluid improvements and the device demands is discussed in this paper, as well as the ER fluid physics and the device designs.
Active control of automobile suspensions
Authors: Hayek Sabih I. , Schwenk Allan E.
Published: October 2005

The response of a vibrational system representing one-fourth of a car's suspension system, with a conventional passive shock absorber, is simulated for a unit step displacement input. In order to minimize the transmitted body displacement and body force, an active quarter-car system, with displacement and velocity actuators, is optimized with respect to the same unit step input. An optimum system is designed to include actuators parallel to a conventional shock absorber. The passive quarter-car model is extended to a half-car model with front and rear tires and independent conventional shock absorbers. The front and rear body displacements and body forces are simulated when the “car” encounters a unit step displacement input. Finally, active control of the half-car is optimized with independent displacement and velocity actuators at the front and rear suspensions. A system similar to the active quarter car is designed to include actuators in parallel to conventional shock absorbers.
Features for speaker-independent recognition of noisy and Lombard speech
Authors: Applebaum Ted H. , Hanson Brian A.
Published: October 2005

Additive noise and noise-induced changes in vocal effort (Lombard effect) cause significant loss of performance for recognizers trained on “normal” noise-free speech when the speech is represented by cepstral coefficients or the combination of cepstral coefficients and their first time derivative. The goal of this work is to find a representation of speech that is more robust to the mismatch of test and training noise conditions. In Applebaum and Hanson [EUSIPCO-90, Barcelona], it was shown, for a digits vocabulary, that adding a second derivative of the cepstral coefficients substantially improved recognition rate for a recognizer utilizing perceptually based LP analysis when the derivatives were calculated over long (> 200 ms) regression windows. It was conjectured that these long windows not work well for a more confusable vocabulary. The current study extends this previous work to a 21-word vocabulary consisting of confusable subsets of the English alpha-digits and the words “no” and “go.” Despite the small duration of the discriminating portion of many of the contrasts involved in this vocabulary, the second derivative feature is again found to yield substantial improvement to recognition rate at long regression window lengths. Unlike the previous (digit vocabulary) study, incorporating the third derivative is found to give further improvement. Results are examined separately by noise condition and confusable subset.
Speaker verification—problems in small systems
Authors: Anto Babu P. , Sridhar Chetlur S.
Published: October 2005

The problems encountered for speaker verification in small systems like laboratory PC-based experimental systems are: expensive implementation of complex algorithms and difficulty in obtaining satisfactory results using standard measures. For example, a system built by the authors following some successful works reported earlier failed to prove that the energy envelope is a good measure for verification. To cater to small-scale applications and to avoid uncertainities regarding the applicability of techniques used by others, a new approach is presented in this paper. The main thrust is to use a fixed set of phrases for verification and to use simple and easily computable measures for classification. For each speaker, training sessions are used to store master feature matrices, which are similar to templates. By appropriate weighting, vectors are derived from these matrices for easy comparisons with utterances during test sessions. The features chosen for the matrices are zero crossings, autocorrelation function, energy distribution, and pitch. In order to avoid complex time-warping and segmentation algorithms, the above features are evaluated for voiced sounds and over one or more pitch periods only. Also, the pitch period is determined by a new algorithm using symmetry check. The hardware needed and the highlights of the software and algorithms are presented in detail. Results for a small group of test speakers are also given. This new technique gives reasonably good performance compared to some of the standard techniques.
Reducing context effects in the subjective evaluation of speech quality
Authors: Rossi Mario , Pavlovic Chaslav V. , Espesser Robert
Published: October 2005

This study investigates whether the contextual invariance and subject invariance of categorical and magnitude estimates of speech quality could be improved by introducing a reference system and by normalizing the results with respect to it. One hundred and forty-four subjects were equally divided into 12 experimental groups and tested in various stimulus contexts using either a magnitude estimation (ME) or categorical estimation (CE) procedures. Various normalization procedures were compared to the nonnormalized results with respect to their immunity to the context/group effects. Different listening conditions were created by using various text-to-speech synthesis algorithms and external noise. The reference condition was natural speech. The results indicate that the introduction of a reference system and appropriate normalization algorithms is beneficial. [Work supported by a grant from the EEC Esprit SAM project.] Also at the Speech and Hearing Center, Univ. of Iowa, Iowa City, IA 52242.
Numerical modeling of reverberation from facets in an elastic seabed
Authors: Schmidt Henrik , Gerstoft Peter
Published: October 2005

The field of ocean bottom reverberation has been lacking efficient numerical models for simulation of elastic scattering from subbottom facets. The only available models have been of finite-difference and finite-element types, both of which are extremely computationally intensive due to the spatial discretization requirements, restricting the applicability to very-short-range reverberation problems. An alternative hybrid boundary element-wave number integration approach has recently been developed [P. Gerstoft and H. Schmidt, J. Acoust. Soc. Am. Suppl. 1 87, S83 (1990)] limiting the discretization requirements to the boundary of the facets and thus allowing for both short- and long-range reverberation simulations in a stratified seismoacoustic ocean environment. For a facet inhomogeneity in an otherwise horizontally stratified environment, the total field is expressed in terms of Green's theorem. By choosing a Green's function satisfying the boundary conditions at all horizontal interfaces, only the interface between the irregularity and the layered medium contributes to the surface integral in Green's theorem. A modified version of SAFARI [H. Schmidt and F. B. Jensen, J. Acoust. Soc. Am. 77, 813–825 (1985)] is applied to compute all Green's functions involved in the boundary integral in both the exterior and interior domains. The boundary element equations, expressing the discretized boundary conditions between the two domains, are then solved, and the total reverberant field is again computed by means of SAFARI. Simulations of both forward and backward reverberation from facets such as salt diapirs and Arctic ice keels will be presented, in part in the form of a computer graphics animation of the time development of the scattered seismoacoustic field. [Work produced at the Naval Res. Lab., Code 5160 Super mini-computer Center and supported by ONR.]
Diffuse reverberation from rough, horizontally stratified elastic boundaries in the ocean
Authors: Kuperman W. A. , Schmidt Henrik
Published: October 2005

Reverberation from a rough elastic layered structure is formulated in terms of a self-consistent perturbation approach applied to the Green's function of a smooth elastic waveguide [W. A. Kuperman and Henrik Schmidt, J. Acoust. Soc. Am. 86 (1989)]. The acoustic field is decomposed into mean (specular) and scattered components, the generation of the latter causing the decay of the former. The scattered component is the reverberation in the waveguide, and computed examples are given. The mechanism for scattering from body waves into interface waves and vice versa is included in the formalism. Simulations of the scattered fields demonstrate this coupling between body and evanescent waves at rough interfaces separating elastic media. Reverberation can also be thought of as signal-generated noise and its structure is of interest when receiving signals in their reverberant background. Further simulations are presented highlighting this aspect of the problem.
Measurement and modeling of acoustic waves in the 50- to 600-Hz range at the Hudson Canyon site
Authors: Rogers Andrew K. , Yamamoto Tokuo
Published: October 2005

Measurements of the shallow-water propagation of acoustic waves on the New Jersey continental shelf were conducted by the Naval Underwater Systems Center, September 1988. On the same site, accurate measurements of geoacoustic properties were obtained by Yamamoto et al. (1988) using the bottom shear modulus profiler (BSMP) method invented by T. Yamamoto. The bottom properties extracted by the BSMP inversion method were introduced to SAFARI computer code for modeling of normal-mode acoustic propagation by the author. Favorable agreement has been found between model calculations and results of the acoustic propagation experiment conducted by NUSC. Comparison has revealed findings about the contributions of the compressional and shear wave characteristics of the seabed to the acoustic waveguide. The compressional wave profile has been found to contribute to the characteristics of the acoustic waveguide modal structure, while the shear wave profile has been found to contribute only to the attenuation of the acoustic wave propagation. [This work sponsored by the Office of Naval Research.]
Spatial structure of long-range, low-frequency seafloor reverberation in the Canary Basin
Authors: Berkson Jonathan M. , Facemire Fredrick R. , Wallmark Jay A.
Published: October 2005

Experimental data on range and azimuthal dependence of long-range, low-frequency reverberation from specific topographic scatterers are limited. Such information is important in designing controlled scattering experiments and in determining scattering mechanisms at the seafloor. In an effort to examine the spatial structure of reverberation, data from the BHST experiment, a test for the development of a wide-area bathymetric reconnaissance technique [Schifter et al., Marine Geodesy 10(1), 1–33 (1986); Erskine et al., Marine Geodesy 10(1), 67–78 (1986)] have been analyzed. Seafloor reverberation from deep explosive line array sources (bubble-pulse frequency of 270 Hz) was received on a towed array in the Canary Basin and processed to obtain spatial distribution of scattering features. Scattering strengths of specific features were estimated from the measured reverberation levels using calculated transmission loss and simplified scattering areas for the long-path propagation. Initial results for four bathymetric features at ranges of 200–650 nmi show scattering strengths that exhibit range independence characteristic of incoherent scattering and are independent of azimuth. [Work supported by ONR.]
Three-dimensional finite-difference modeling of acoustic backscatter from rough seafloors
Authors: Burns Daniel R.
Published: October 2005

A second-order, staggered grid, three-dimensional finite-difference model that treats the acoustic wave equation in the water column and the full elastic wave equation in the subbottom was used to generate wavefield snapshots and time-series results for rough seafloor conditions. Deterministic seafloor structures such as fault scarps, channels, and seamounts result in significant acoustic backscattered energy. Trapped energy propagating inside a channel within the seafloor results in continued radiation of acoustic energy back into the water column. Likewise, seismic energy reverberating inside the seamount, as well as guided waves propagating around the seamount, results in additional acoustic backscattered energy. Depending on the angle between the source-receiver plane and the strike of the seafloor topography, seismic and acoustic energy is focused and scattered creating complex signals in which wave types are preferentially amplified or attenuated as a function of azimuth. Snapshots and time series have been generated in a number of azimuthal directions to illustrate these effects. Although the 3-D finite-difference model cannot handle broadband, long-range wave propagation at this time, it is capable of answering many critical questions regarding the role of the seafloor and subbottom elastic properties on the backscattered acoustic wave field. [Work supported by ONR.]
Parabolic equation modeling of bottom reverberation using seafloor roughness spectra
Authors: Nghiem-Phu Lan , Tappert Frederick
Published: October 2005

A numerical implementation of the multiple forward, single backscattering approximation [F. Tappert, The Parabolic Approximation Method (Springer-Verlag, New York, 1977)] has been used to solve the parabolic wave equation for the prediction of bottom reverberation levels. Bathymetry for selected sites in the Mediterranean Sea is extracted from standard ocean databases and various scales roughness is superimposed on the deterministic seafloor via spectral techniques and used as inputs to the model. Examples will be shown to illustrate the dominant effects of seafloor ensonification, which primarily depends on transceiver(s) position(s) and on the acoustic frequency of interest. [Work of second author partially supported by ONR.]
Computer simulations of effects of small-scale fluctuations on bottom reverberation
Authors: Fulloral James K. , Slater Richard R. , Showalter James
Published: October 2005

The effects of small-scale fluctuations in the oceanic thermodynamic field on bottom reverberation are simulated for monostatic source/receiver geometries. Fluctuations in the near-source environment may lead to large changes in the vertical arrival structure at the water-sediment interface. The changes in vertical arrival structure lead to changes in observed reverberation time series. Using models based on observations of oceanic fine temperature structure, a series of upper ocean sound-speed profiles is derived, the resulting sound-speed fields are then used to derive the reverberation statistics.