bluuebunny/crossref_metadata_2025 huggingface dataset:
Nondestructive detection of fatigue damage
Authors: Green Robert E.
Published: October 2005

This paper will discuss the various nondestructive techniques which have been used or which are potentially useful for detection of fatigue damage. The ideal nondestructive testing technique would permit very early detection of fatigue damage so that proper assessment of the severity and rate of severity increase of the structural damage leading to failure can be made. Thus, the most sensitive systems would be capable of detecting motion, pileup, and breakaway of dislocations; the next most sensitive systems would be capable of detecting microcracks; the least sensitive systems would only be capable of detecting macrocracks. It is practically expedient to have nondestructive techniques which can successfully detect fatigue damage in each of these regimes since some components can tolerate larger amounts of fatigue damage or larger crack sizes than others without serious concern for the structural integrity of the component. It will be shown that the best nondestructive technique for detecting and sizing of macrocracks is ultrasonics; the best nondestructive techniques for detection of microcrack formation and possible pre-microcrack fatigue damage are ultrasonic attenuation and acoustic emission; a large number of nondestructive techniques for residual stress (strain) measurements are candidates for extremely early fatigue damage detection, with current interest primarily directed at ultrasonic wave velocity measurements.
A perceptual study of the voiced-voiceless contrast in syllable-final stops
Authors: Ingrisano Dennis , Hillenbrand James , Smith Bruce L. , Flege James E.
Published: October 2005

Computer editing techniques were used to remove voicing in 10-ms steps from syllable-final closure intervals of natural tokens of [pɛb,pɛd,pɛg, pag, pig, pug]. Release bursts were retained and a pitch-synchronous editing routine was used to equalize vowel durations at about 110 ms. Identification results from 23 listeners showed that, in general, relatively large amounts of voicing had to be removed from the closure intervals before perception changed from voiced to voiceless. For some of the continua (e.g., /pɛb-pɛp/), voiceless responses began to predominate when more than 95% of the closure interval was rendered voiceless. For other continua (e.g., /pag-pak/), the change from voiced to voiceless responses did not occur until voicing during the closure interval and a portion of the VC transition were removed. We found no place effect for the syllable-final stop; a significant, but relatively small effect for the vowel context was observed. In a second experiment, 11 listeners identified stimuli from which the release bursts had been excised. No significant differences were found between identification boundaries with or without bursts. Findings are discussed in relation to the nature of voicing cues in syllable-final stops. [Work supported by NIH: T32 NS 07100-04 and NIH Biomedical Sciences Support Grant: 5 S05 RR07028.]
Limits on the power of silence as a stop manner cue
Authors: Repp Bruno H.
Published: October 2005

Previous research has shown that a bounded interval of silence is an important perceptual cue to stop manner, and that it may be both necessary and sufficient. However, most earlier stimuli did not include stop release bursts. When a release burst is present, a preceding interval of silence is frequently not necessary for perception of a stop, particularly when place of articulation is alveolar or velar. New experiments varying burst duration and amplitude reveal listeners' exquisite sensitivity to burst information, suggesting that the release burst, and not silence, is really the primary cue to stop manner. As to the sufficiency of the silence cue, it is well-known that a reasonable stop percept can be created by splicing silence into a previously stopless utterance (e.g., /slɪt/→/splɪt/). However, new experiments show that under certain conditions listeners readily perceive these modified utterances as two stopless pieces separated by silence (e.g., /s-lɪt/). Stop perception, in these cases, seems to depend on an integrative perceptual strategy that, however, can be disengaged without leaving the speech mode. Thus, silence alone is not always sufficient for stop perception. [Work supported by NICHD.]
Nonmonotonic discrimination functions for temporal gaps in noise-buzz sequences. II: Filtering affects the discrimination peaks
Authors: Divenyi Pierre L.
Published: October 2005

It has been reported [P. L. Divenyi and W. F. Danner, J. Acoust. Soc. Am. Suppl. 1 55, S31 (1975) and J. D. Miller et al., J. Acoust. Soc. Am. 60, 410–417 (1976)] that discrimination of the gap bounded by a brief, filtered noise burst and a longer,/a/-like buzz is a nonmonotonic function of the gap duration, in a way similar to the well-known nonmonotonicity of the VOT discrimination functions [A. S. Abramson and L. Lisker, Proc. 6th Int. Congr. Phonet. Sci., 569–573 (1970)]. Earlier, we reported that discriminability of the gap in such sequences is affected by passing the stimulus through 1/3-octave filters and that the discrimination thresholds are correlated with the difference between noise and buzz intensity within the particular bands—i.e., With one of the main attributes of temporal masking. The present experiments extend the previous ones, in that the continuum of “buzz-onset-times” was explored in finer steps (10, 15, 20, 25, 30, 35, 40, 55, and 70 ms from noise onset to buzz onset). In addition to determining gap discrimination thresholds for the original, unfiltered stimulus, thresholds were also obtained at three filter settings (780, 1230, and 2550 Hz center frequencies). Three trained listeners participated. While the smallest thresholds at the 780 and 1230-Hz filter settings occurred at the same gap duration as for the unfiltered stimulus (15 ms), the threshold minimum at the 2550-Hz filter setting occurred at longer gap durations (20 and 25 ms). Implications of these results for variations in voicing category boundaries as well as for the possible role of listening bands in the perception of speech will be discussed. [Work supported by NIH Grant NS 16019 and by the Veterans Administration.]
Listening systems—Application analysis
Authors: Nábělek Anna K. , Łȩtowski Tomasz , Nábȩlek Igor V.
Published: October 2005

For many years, listening systems have been used for the hearing-impaireds in schools for the deaf and the general public for simultaneous transmission in several languages. Recently, listening systems started to be installed for pleasure listening in theaters, churches and concert halls. For multi-lingual conferences, the main emphasis was on separation of the channels. This is usually achieved by hard-wired connections with limited frequency response to the range most important for speech intelligibility. The systems for the hard-of-hearing have also limited frequency response, a philosophy inherited from hearing aids. The priority here was adequate gain allowed by the system, provision to couple with personal hearing aids, and sometimes flexibility in seat selection. The modern listening systems installed in public places should serve both the general public and hard-of-hearing listeners. They are designed for both speech and music transmission and quality of sound becomes the primary objective. The authors have examined several listening systems installed in the New York City and Washington, D. C. areas and will share some observations which can be helpful for the designers of the systems. [Work supported by NIH.]
Global measures of resolution in acoustic tomography
Authors: New Ronald , Eisler Thomas J. , Calderone Denise , Porter David L.
Published: October 2005

We have previously reported on applications of the Backus-Gilbert theory to the problem of Ocean Acoustic Tomography [New et al., J. Acoust. Soc. Am. Suppl. 1 70, S38 (1981)]. This theory directly provides estimates of two important elements of system performance, viz., resolution and variance. In this paper, we concentrate on the former. Resolution of a pointwise estimate of the sound speed anomaly is fully described by a resolution kernel which, when integrated with the actual values of the unknown function, produces the estimate. This kernel is a system property; its suitably defined width is used to define resolution length at a point. We consider a horizontal ocean slice and define a global measure of resolution to be the area within which a given resolution criterion is satisfied. The shape and size of such regions have been calculated for various tomographic configurations. The behavior of this global measure as a function of number of moorings suggests that many more moorings are needed for resolution of mesoscale phenomena than had previously been thought!
Remote sensing of environmental parameters
Authors: Nadeau Paul G. , Nowak Richard
Published: October 2005

This paper describes a prototype environmental parameter sampling system and the problems it is intended to address. Increasingly sophisticated sonar performance models are being developed in support of advanced sonar system. A graphic portrayal of the simulated performance of a notional sonar system is presented as an example of the sensitivity of a simulation to the input environmental parameters including the volume scattering profile. The prototype environmental sampling system is configured to provide an estimate of the volume scattering profile. Operationally, this profile is not available to simulation users. The prototype environmental parameter sampler will be used to demonstrate how simulation errors due to this critical environmental input can be eliminated. The use of the microcomputer based prototype hardware as a basis for estimating other environmental parameters will also be presented. [Work supported by NAVSEA.]
Prediction of ambient noise in the deep ocean from surface wind data obtained via satellite
Authors: Shonting D.
Published: October 2005

Open ocean measurements at both shallow and deep depths show that ambient acoustic noise above 10–15 kHz increases nearly linearly as the log of the wind speed. This relationship is studied under a variety of wind/sea conditions at 15, 20, and 25 kHz bands using a newly developed self-contained ambient noise recorder (SCANR). Assessment is made of possible ambient noise prediction from wind speed data obtained from a satellite mounted microwave scatterometer.
Acoustic observations of large internal soliton waves at the Sulu Sea
Authors: Tsai John J. , Proni John R. , Craynock Julius , Apel John R. , Holbrook James R.
Published: October 2005

Landsat imagery taken over the Sulu Sea, Philippines, reveals the existence of very long, soliton-like internal wave pockets of semi-diurnal periods. An experiment was conducted in May 1980 to test the hypothesis of tidal generation, and to parameterize these soliton waves by obtaining additional measurements using acoustic remote sensing techniques as well as conventional methods. In this paper, we compare wave characteristics from various measurements, and present different wave forms due to ship speed change with respect to the wave velocity. Effects of deep scattering layer on the acoustic measurements are discussed, using computer-digitized backscattering intensity contour.
The generation and measurement of CO2 laser-induced acoustic pulses in sand
Authors: Hickman G. Daniel , Edmonds John A.
Published: October 2005

A pulsed CO2 laser (1-μs pulse width, 10-J pulse energy) when focused to a 1-cm2 spot on the surface of sand generated an acoustic field in the sand. The sound pressure level (SPL) measured with a hydrophone 1 cm below the surface was 173 dB re 1μPa with a frequency spectrum <3 kHz. Focusing this same laser on a water surface yields equivalent values in water of 200 dB re 1μPa and 25–30 kHz, respectively. Sound velocities in sand ranged between 145–184 m sec−1 depending on the dryness of the sand. When the acoustic pulses were generated in a few cm of water covering the sand, the sound velocities measured in the sand increased to 610–750 m s−1 while the spectrum was composed of both the high frequency water spectrum and the low-frequency sand spectrum. Acoustic pulse measurements were made to sand depths of 40 cm. In addition, pulses were detected after reflection off a metal plate at a depth of 45 cm. A discussion will be given of possible remote sensing applications. [Research supported by ONR, Code 422CS.]
Acoustics of EPCOT
Authors: Long M.
Published: October 2005

Walt Disney Productions and their engineering arm WED Enterprises have undertaken to build the Experimental Prototype Community of Tomorrow (EPCOT), the largest theme park complex in the world. This park recently opened in Orlando, Florida and features 17 pavilions in two groups, Future World and World Showcase. The project which will cost over a billion dollars to build has a number of interesting and unique acoustical features. The author who served as consultant on 6 of 7 Future World Pavilions and 8 of 10 World Showcase Pavilions will review several of the most interesting acoustical aspects of this monumental facility.
Temperature effects in piezoelectric ceramic-stack resonators containing cement joints
Authors: Goodhart C. L.
Published: October 2005

Underwater acoustic transducers often employ a stack of piezoelectric rings that are cemented together, face to face, as the driving element. Recently, the performance of such a transducer was observed to seriously deteriorate when subjected to somewhat elevated temperatures. The source of the problem was determined to be the effects of elevated temperature on the cement joints. This paper discusses the temperature-related effects in cement joints on the performance of ceramic stack resonators. Measurements of cement-joint compliance and loss factor as a function of temperature are presented. The behavior of ceramic-stack resonators resulting from temperature effects in the cement joints is examined using results from both theoretical models and experimental measurements. It is shown that as temperature increases, a corresponding increase in the cement-joint compliance and loss factor results in detuning, decreased electromechanical coupling and increased heating (which further provokes the problem) in the resonator. Methods of modifying the cement-joint characteristics in order to reduce the ill effects of temperature are discussed. [Work supported by NAVSEA 63X.]
Low temperature diffusion bonding of transducer ceramics
Authors: Thompson Stephen C. , Lux Andrew A. , Gray John J.
Published: October 2005

A metallic diffusion bonding process has been developed to bond piezoelectric ceramics into stacks for use in sonar transducers. The assembly process starts with stacking the ceramics to be bonded alternately with layers of a diffusing material. This material is a thin shim of a four component alloy of indium, silver, lead, and tin. This alloy bonds by diffusing uniformly into the surface of the fired-on silver electrode on the ceramic face. A solid silver electrical lead may also be included in the stack. Surface preparation of all pieces is critical as minute amounts of surface oxidation or other contaminants will yield unacceptable bond strength. The stack is placed under moderated compression in an elevated temperature environment for the diffusion process. The temperatures involved are well below the Curie point for the ceramic material and give minimal degradation of the electrochemical coupling constant of the material. Tensile testing shows the diffusion bond to have higher strength than the ceramic material—The ceramic will fracture before the bond breaks. Preliminary results show the metallic bond to have less compliance and a lower loss factor than conventional epoxy bonding methods. Further tests are underway to better determine the bond mechanical properties.
Expendable sonobuoy hydrophone
Authors: Henriquez T. A. , Tims A. C.
Published: October 2005

A shock-hardened, low-cost pressure sensitive hydrophone has been developed for applications in expendable air-deployed passive sonar systems. The hydrophone is completely functional after sustaining the shock from an underwater explosion equivalent to 5 kg of dynamite at a distance of 3 m. A novel preamplifier integral to the hydrophone has very low power consumption and self-noise. The preamplifier also has a very low output impedance which is necessary to drive long cables. The design is easily assembled and is oriented toward mass production to provide large quantities at very low cost.
A monitor hydrophone system for accurate measurement of sound pressure levels produced by deep towed sources
Authors: Bush Ira J. , Demetres Ronald N.
Published: October 2005

The only way to accurately determine the sound pressure level of a towed high power acoustic source is to place a calibrated standard at a fixed distance in close proximity to the source. From a practical standpoint, the source and monitor hydrophone must share the same electromechanical cable. Crosstalk results from cable sharing because the drive signal to the high power source and the return monitor signal are of the same frequency and contained within the same cable. For the case of most tow cables, shielding between the two signals is almost nonexistant. The resulting cross-talk obfuscates the output and usefulness of the monitor. If the return signal is frequency modulated, the cross-talk signal no longer occupies the bandwidth of the information signal. It is a simple matter to filter out the cross-talk signal before the information signal is locally demodulated. Synchronous detection is utilized to insure accuracy, linearity and immunity to carrier drift. A prototype system has been successfully tested at calibration facilities and out at sea demonstrating accurate measurements for both electrodynamic and piezoelectric sources. [Work supported by NRL.]
Acoustical matching of a traveling wave heat engine
Authors: Ceperley P. H.
Published: October 2005

A traveling wave heat engine is a thermally driven acoustical amplifier which uses a differentially heated regenerator as the gain element. However, because it amplifies only the volumetric flow component of an acoustical wave and not the pressure component, the impedance of the wave changes in going through the device, thereby complicating the impedance matching to the input and output wave guiding tubes. This paper derives expressions for this impedance change and discusses possible matching methods. It also presents some experimental verification of both. Other relevent matching considerations such as impedance variations due to temperature and tubing dimension changes are discussed. Earlier papers by this author introduced the traveling wave heat engine and calculated its gain and efficiency [P. H. Ceperley, J. Acoust. Soc. Am. 66, 1508–1513 (1979)].
Historical development of the use of A-weighting
Authors: Botsford James H.
Published: October 2005

The original writers of the specifications for American sound level meters wished to provide a standard instrument that would indicate the loudness of sounds as perceived by the normal human ear. As loudness varies with the sound pressure level and frequency of a sound, three frequency weighting networks were specified to mimic the response of the ear at the loudness levels of 40, 70, and approximately 100 phons. The first of these was designated the A-weighting and the last was called the C-weighting, which was a flat response. When the rules given for using the weighting networks were followed, the sound level meter indicated very accurately the loudness levels of pure tones but not broadband noises. For measuring noises more precisely, special rules were developed by sound level meter users for combining the readings obtained using two or more of the weighting networks, particularly the A- and C-weighting networks. Long experience with estimating human responses to noise from sound level meter readings has led to the general use of the A-weighting network alone for many purposes.
Frequency weighting for community response and annoyance
Authors: Pearsons Karl S.
Published: October 2005

Subjective evaluations of various noises have been studied over the years by many researchers. The results of these studies have provided various schemes for predicting annoyance, loudness, or noisiness of the noises using objective measures. A review of some of the pertinent studies will be presented including evaluations of some of the measures in current use. A discussion of community response studies will also be presented which indicates the variety of frequency weightings which have been used in past research. Information on situations which may require special weightings such as those which include impulse noise, tones, or high-frequency noise will also be presented. Mention will also be made of an approach which considers the ambient noise as well as the intruding noise.
Frequency weighting for the assessment of speech interference
Authors: Webster J. C.
Published: October 2005

All known frequency weighting systems for evaluating the effects of noise on speech discrimination will be reviewed and evaluated. These will include the articulation index (AI); the speech interference level (SIL); speech, noise, and alternate noise contours (SC, NC, NCA); the speech transmission index (STI) and modulation transfer function (MTF). Also there will be a discussion of the problems associated with measuring speech level (vocal effort) and with measuring those aspects of noise that affect talking and/or listening levels and the discriminability of spoken language. It has been found that the equivalent noise level (Leq) measured in dB(A) is a good measure of the effects of time varying noise on speech intelligibility (discrimination). Therefore it will be suggested that the A-weighted level of noise, which is not the best speech interference measure for steady-state wideband noises, may be the best simple comprehensive measure of the speech interfering aspects of noise.
Frequency weighting to assess sleep and classroom environments
Authors: Lukas Jerome S.
Published: October 2005

Only a limited amount of specific data are available regarding the relative effectiveness of A- versus C-weightings in predicting the effects of noise on sleep or of noise on children in classrooms. These data will be reviewed, but suggest that with respect to sleep the results are equivocal. In the classroom, the single study available [Lukas, DuPree, and Swing, in press] suggests the C-weighted noise levels are more highly correlated (negatively) wth academic achievement that are A-weighted levels.
Relaxation mechanism for low-frequency sound absorption by boric acid in sea water
Authors: Mellen Robert H.
Published: October 2005

Investigation of the low-frequency boric acid relaxation in sea water by the resonator method shows that the mechanisms is an exchange reaction with the carbonate system, i.e., B(OH)3 + CO3−2⇄B(OH)4− + HCO3−. However, the absorption is small unless calcium is present; then the system can be modeled as a complex exchange involving the ion pairs CaCO3° and CaB(OH)4+. Theory and supporting experimental results are presented. [Work supported by DARPA.]
The effect of pulse length on transient cavitation inception
Authors: Roy Ronald A. , Crum Lawrence A.
Published: October 2005

Most ultrasonic systems used for diagnostic purposes employ short acoustic pulse rather than cw operation in order to accomplish imaging or Doppler scanning. It is also commonly thought that these short-pulse systems, although driven at high acoustic pressure amplitudes, do not induce cavitation effects due to the low duty cycle of the ultrasound. Although stable cavitation typically requires cw operation for its effects to be present, transient cavitation can theoretically occur in pulse lengths as short as one cycle. We have developed a detection scheme utilizing sonoluminescence to observe the cavitation produced by pulses of varying length and repetition rate. Normal detection methods that examine noise emitted by the collapsing cavity can not be used in short-pulse excitation due to the high spectral content of the pulse itself. We have observed transient cavitation effects (sonoluminescence) in pulse lengths as low as a few tens of cycles and are modifying our apparatus to go to even shorter pulses. [Work supported in part by the ONR and the NSF.]
Acoustical measurement of energy and momentum accommodation coefficients on polycrystalline platinum
Authors: Shields F. Douglas
Published: October 2005

The acoustical method of measuring energy (EAC), and tangential momentum (TMAC) accommodation coefficients [F. Douglas Shields, J. Chem. Phys. 62, 1248–1252 (1975)] has been used to measure the accommodation coefficients of Ne and He on vacuum flashed Pt and on Pt after exposure to O2, N2, CO, and CO2. Vacuum flashing the Pt at 1400 K in the presence of misch metal getter resulted in a surface that was reproduced a number of times after exposure to various adsorbants. The combined value of TMAC + EAC for this surface was 0.80 ± 0.04 for Ne and 0.64 ± 0.04 for He. These values are significantly greater than values measured earlier in a similar way for W [F. Douglas Shields, J. Chem. Phys. 72, 3767 (1979); 76, 3809 (1982); 76, 3814 (1982)]. Exposing the flashed surface to O2 raised the combined values of TMAC + EAC to 1.28 for Ne and 1.00 for He. Flashing the surface at 1100 K removed some of the O2 and flashing at 1400 K restored the original AC values. In contrast to this, exposure to N2 did not change the surface. Exposure to CO raised the AC values but not as much as O2 but the CO was removed by the 1100 K flash. AC values have also been obtained for N2 on Pt, for O2 on O2 on Pt and for CO on CO on Pt.
Pulsed signal distortion in dispersive constant-group-velocity media
Authors: Edwards P. L.
Published: October 2005

Ultrasonic velocity measurements in solids are often made by measuring the transit times through known distances by short transient pulses. In dispersive and/or attenuative media the signal may be distorted, and the question often arises as to which part, or parts, of the initial and transmitted waves are to be used in the transit time measurement, and just what velocity—phase, group, signal, etc.—is actually being measured. A theoretical and computer study has been made of the distortion undergone by three types of pulses in transmission through nonattenuative dispersive constant-group-velocity media, and the effect that this may have on velocity measurements. Theory shows that a pulsed sine wave in such media does not spread out as is normally the case for dispersive media, but repeats itself at definite repetition distances. Computer plots showing the changes that the pulses undergo during transmission will be presented, and how these changes effect velocity measurements will be discussed.
An acoustic anomaly, sound in a fluid with periodically stratified density and uniform sound speed
Authors: Schoenberg Michael , Sen P. N.
Published: October 2005

The exact solution for harmonic waves in a periodically layered acoustic medium [M. Schoenberg and P. N. Sen, J. Acoust. Soc. Am. Suppl. 1, 71, S44 (1982)] leads, in the long wavelength limit, to some curious results. The most striking is that the equivalent homogeneous acoustic medium is an ideal transverse isotropic fluid. Anisotropy is manifested through the density which must itself be a transverse isotropic tensor! In turn, this leads to the anomalous result that in a stratified medium composed of the same speed c, but different densities, the long wavelength sound speed need not be equal to c. Let ρi be the layer density of the ith layer and let 〈-〉 denote a thickness weighted average. In the long wavelength limit, propagation perpendicular to the stratification is at a speed c/(〈ρi〉 〈1/ρi〉)1/2 which is less than c. At high frequencies, wave fronts must propagate with speed c in all directions. We examine, over the entire frequency range, the dependence of wave speed normal to the stratification and the alternation of pass and stop bands for such propagation.
Frequency dependence of the velocity and attenuation of sound wave propagating in a fluid containing a dense suspension of particles
Authors: Varadan V. K. , Ma Y. , Varadan V. V.
Published: October 2005

Acoustic waves propagating through a suspension of particles in a fluid are dispersed and attenuated due to two reasons—geometric dispersion or multiple scattering as well as inherent losses if any associated with the particles or the host fluid. A self-consistent multiple scattering theory is used incorporating the T matrix of the individual particles to characterize their response and a configurational averaging procedure to characterize the statistics of the distribution. Numerical results are obtained for the phase velocity and attenuation of the coherent wave in such a system as a function of the frequency, size, shape, and distribution of the particles of the suspension. These results will be compared with single scattering approximation as well as more approximate methods. The results are of interest in the study of ocean bottom sediments and biological applications.
Resonance dependence of the dynamic effective properties of particulate composites
Authors: Gaunaurd G. C. , Überall H.
Published: October 2005

We have developed a general deterministic resonance theory for the effective dynamic properties of elastic (or viscoelastic) solids containing ensembles of randomly distributed solid spherical inclusions not necessarily of the same sizes. The method follows our earlier procedure for fluid inclusions [J. Acoust. Soc. Am. 71, 282–295 (1982)] which is generalized here for solid inclusions of known size distributions. We present a summary of the theory and numerous displays of calculations for the effective shear speed, the effective (sound) speed of compressional waves, and the effective attenuation versus ka at known concentrations Φ, or versus Φ at various values of ka. The current theory contains nearly all deterministic results available in the earlier literature and it compares quite favorably, at least within the “acoustical branches,” to the only set of dynamic measurements in the recent literature [i.e., V. Kinra and A. Anand, Int. J. Solids Struct. 18, 367–380 (1982)] which extends sufficiently beyond the long-wavelength regime to permit comparisons. [H. Überall is also at Catholic University, Washington, DC and is additionally supported by ONR.]
Recent developments in active vibration suppression
Authors: Meirovitch Leonard
Published: October 2005

In many mechanical systems, vibration is an undesirable effect, to be suppressed as expeditiously as possible. Traditionally, vibration suppression has been achieved by passive means, such as vibration absorbers, dampers, etc. In seeking more efficient vibration reduction, particularly for larger-order systems, there has been a recent trend toward active feedback control. Vibrating systems can be divided into lumped-parameter and distributed-parameter systems. Feedback controls for lumped-parameter systems can be designed by methods of modern control, such as pole allocation, optimal control, etc. In applying techniques of modern control to distributed-parameter systems, difficulties arise because modern control theory is essentially a discrete (-in-space) theory, not well suited for distributed systems. To circumvent this problem, the tendency has been to discretize (in space) the distributed system. This approach has the effect of imparting to the distributed system all the characteristics of a lumped system. A recently developed method, called the independent modal-space control method, is more faithful to the distributed model and is more suitable for the control of distributed systems.
Historical review and recent development of active attenuators
Authors: Leventhall H. G.
Published: October 2005

Active attenuators are now about 50 years old. Their history has been one of disappointment, demise, and rediscovery, although modern electronics and computing has resulted in remarkable advances in recent years. Early practical applications were in the mid 1950's for the control of transformer noise and a wide range of applications was proposed by Olsen. Some of these are the ones which are being developed today. Following the developments in the 1950's, there was little progress for a further 10 years when theoretical advances were made by Jessel. Much of the modern interest in active attenuators can be traced to Jessel, who was the first to use a unidirectional source which gives absorption of sound rather than cancellation by interference. Subsequently, a different form of unidirectional source was proposed by Swinbanks and active attenuators have attracted considerable interest during the past 15 years. The current trend is to return to simple radiating systems and make electronic compensation for their deficiencies. At the present time, the basic Olsen type of monopole attenuator gives a good prospect for an inexpensive duct attenuator for low-frequency noise.
Satellite remote sensing and underwater acoustics: What you see is not necessarily what you get
Authors: Henrick Robert F. , Johnson Charles L.
Published: October 2005

Continuing efforts in satellite systems and analysis methodology may soon result in routine production of ocean weather maps for substantial regions of the world's oceans. Such maps might illustrate positions of such mesoscale features as ocean fronts and eddies and be available to the acoustician on a routine basis. (In fact such products are presently available for the Gulf Stream region.) However, the use of such maps without guidance as to the acoustic significance of these features may have limited utility to an acoustician deciding on the deployment of an acoustic system, and may indeed serve to mislead. A front or eddy that may be clearly visible to a satellite sensor may not be acoustically significant to the system under consideration. Even if the mesoscale feature has acoustic impact, the surface expression of the feature may not coincide with the acoustically important portion of the feature. Alternately, acoustically relevant mesoscale anomalies may not be visible to a satellite sensor. To illustrate these cautions, oceanographic data taken by the U.S. Naval Oceanographic Office Ocean Measurements Program in the Northeast Atlantic and Norwegian Sea are utilized in conjunction with numerical acoustic propagation simulations. This region is typical of areas where significant mesoscale activity occurs and oceanographic sampling is sparse, so that satellite remote sensing may be very useful in predicting acoustic system performance. However, examples presented illustrate that significant work must be done before a useful satellite oceanographic product can be turned into a useful acoustic product. [Work supported by the U.S. Naval Oceanographic Office.]
SAW's, the connection between space and hydrophones
Authors: Nysen Paul A.
Published: October 2005

SAW (Surface Acoustic Wave) devices are a new class of signal processing and signal generating elements now in use with radar and communication systems. These devices can be made as filters, delay lines, or resonators which are small, rugged, and inexpensive. This paper discusses the mechanical and thermal sensitivities of SAW's with a view towards using these properties to sense temperature, pressure, and sound in water. Various modulation schemes are examined. Finally a buoy system is explored, involving a gauge string of SAW hydrophones/temperature/pressure sensors which can be enabled and interrogated from a space platform.
Satellite detection of anomalies in ocean ambient noise?
Authors: Dunlap Calvin R. , Jung Glenn H.
Published: October 2005

ASTREX (Acoustic Storm Response Experiment) observations made in mid-November 1980, along a flight line from Cape Mendocino, California, toward the central Gulf of Alaska, included concurrent airborne expendable bathythermograph (AXBT) measurements of surface and sub-surface temperatures and ambient noise measurements (1–2000 Hz) using sonobuoys, with hydrophones deployed at 400 ft (122 m). NOAA-6 satellite infrared observations also were available. It is shown that anomalous (low) ambient noise values, measured at 122 meters and frequencies 500–2000 Hz, appear to be associated with a warm-core eddy extending from the surface to depths greater than 250 m. This thermal feature is shown clearly at the sea surface in a satellite infrared photograph. It appears that the warm-core eddy had refracted ambient noise away from the hydrophone, since ambient noise levels are several decibels higher outside the eddy region. It is suggested that careful studies are warranted of relations between ambient noise, thermal structure, and surface temperature features detectable by satellite, exploring the possibility of using satellites in some circumstances to indicate subsurface ambient noise anomalies.
A suprasegmental component in a speech recognition system
Authors: Vaissière J.
Published: October 2005

This paper presents a method for integrating a suprasegmental component into an automatic speech recognition system. It aims to narrow down the number of word candidates proposed by the lexical component by rejecting candidates whose fundamental frequency (F0), durational, and intensity patterns do not match with the observed prosodic pattern of an input word. A stylized F0 pattern for each word in the lexicon is described by a network specifying F0 rises, levels, and falls. Expected durational relationship as which of two arbitrary segments (i.e., phonemes and/or syllables) within the word is longer are specified by a matrix: these dichotomous relations are marked in the matrix only when they are invariant, regardless of the context, i.e., the position of the word in the utterance, and of speakers. For intensity, a similar matrix is constructed for each word. As expected, the degree of suprasegmental constrains increases as the length of the words become longer. These suprasegmental characteristics are determined, manually or automatically, by using word-independent general rules, or by specific rules which can apply to a subset of the lexical words related to the task-dependent syntax [which constrains the position(s) of a word in the utterance, and consequently increases prosodic expectations for that word].
A nonuniform sampling approach to isolated word recognition
Authors: Soong F.
Published: October 2005

Speech signals are known to consist of both stationary and transient parts. The transient part is invariably shorter than it stationary counterpart. The standard uniform sampling approach for creating word reference patterns for speech recognition will naturally emphasize the longer steady-state portions more than the short transitional portions. Since the transient portions of words contain valuable consonantal information for recognition, a nonuniform sampling approach seems to be a natural choice for potentially improving the recognizer performance as well as for reducing the overall sampling rate and computational complexity of the recognizer. However, there are some inherent difficulties in this approach. They are (1) artifacts and distortions such as breath noise, mouth clicks, etc. contribute non-negligible spectral variations and result in over-sampling in these regions; (2) the inherent variance of the spectral estimate of the transitional regions is much larger than than of a steady-state region. Nonuniform sampling therefore suffers high uncertainty in its spectral estimate since the high variance regions are sampled more often. Recognition results are presented on a 39-word, alpha-digit vocabulary and compared with those from a uniform sampling approach.
Speaker-independent recognition of English letters
Authors: Cole Ronald A. , Phillips Michael S. , Brill Scott M. , Specker Philippe , Pilant Andrew P.
Published: October 2005

An isolated word recognition system is described which classifies English letters using features extracted from speech. Speech was sampled at 16 000 Hz, and pitch, endpoint detection, and location of vowel onset and offset were computed directly from the digitized data. A 128-point DFT was performed every 3 ms using a 20-ms window, and frequency samples were linearly compressed to 54 coefficients spanning the frequency range from 62.5 to 7000 Hz. Most feature extraction algorithms used this 54-coefficient representation. A knowledge-engineering approach was used to create feature extraction algorithms from digital spectrograms of 2080 tokens of English letters produced by ten male and ten female talkers. Featural measurements included voice onset time, onset abruptness, formant frequencies, and formant trajectories, and all made use of the end-points of the sound and/or vowel onset as temporal anchors. Feature values from a subset of the 2080 utterances were used to train the classifier, and system performance was evaluated on the basis of errors in classifying the remaining utterances. Features will be described and recognition results will be reported for each decision point within the system. [Supported by NSF.]
Decisions about features
Authors: Brill Scott M. , Phillips Michael S. , Lasry Moshe J. , Stern Richard M.
Published: October 2005

This paper describes the methods of statistical analysis used to classify letters in a feature-based, speaker-independent isolated letter recognition system. A hierarchical decision structure was implemented so that decisions at each node of the decision tree could be made on the basis of a small number of relevant features. For example, the 26 letters were first classified into vowel categories on the basis of first and second formant frequencies. The specific decisions, and the features used to make them, were selected by a clustering analysis of training data. At each decision node of the recognition system the test utterance was first analyzed using Fisher linear discriminant functions, with threshold weights individually set for each pairwise decision in order to minimize misclassifications. When a decision could not be made with certainty, classification was performed using a maximum likelihood procedure assuming multivariate Gaussian statistics. The sequential use of nonparametric and parametric discriminant functions produced superior classification to that obtained with either of the separate analyses. The overall system structure will be discussed in terms of practical tradeoffs between the number of features used at each decision node and the system's overall probability of error. [Work supported by NSF.]
A powerful post-processing algorithm for time domain pitch trackers
Authors: Specker Philippe
Published: October 2005

Existing pitch tracking algorithms are not precise nor reliable enough to be useful in feature-based recognition systems. It is possible, however, to analyze the errors produced by a particular algorithm and then reduce these errors using post-processing techniques. Error patterns were analyzed for a time-domain pitch tracker [W. H. Tucker and R. H. T. Bates, IEEE Trans. ASSP-26, 597–604 (1978)] and a post-processing algorithm, based on artificial intelligence techniques, was written in order to eliminate errors. Performance was compared for the original and modified versions of the pitch tracker for a number of speakers using both isolated words and sentences. All types of errors were reduced by the post-processing algorithm. Voiced-voiceless decisions were performed with less than 1% error for 2080 letters produced by males and females. The fundamental frequency microstructure was tracked sufficiently well to be used in extracting phonetic features in a feature-based recognition system.
Very large vocabulary recognition (VLVR): using prosodic and spectral filters
Authors: Waibel Alex
Published: October 2005

The use of Very Large Vocabularies (∼20 000 words) imposes two major constraints on the design of isolated or connected word recognition systems: the efficient reduction of the large search space to a subvocabulary of manageable size [V. Zue and D. Shipman, J. Acoust. Soc. Am. Suppl. 1 71, C7 (1982)] and the robustness of the search space reduction heuristics involved. Psychological evidence suggests that prosodic and robust segmental features may be used as preliminary decision criteria in human speech perception. In this paper we present attempts to apply such heuristics to the design of VLVR systems. As a data base a 20 000 word vocabulary has been compiled providing phonemic, prosodic, and pragmatic information. Based on this corpus, the tradeoffs between the robustness of certain features and their power to reduce the search space have been studied. Our results indicate that combining prosodic information (syllable counts, stress patterns) with a set of robustly detectable features (frication, stops, vowel nuclei of stressed syllable) can reduce the vocabulary size to groups of less than 400 words. Additional potentially useful prosodic features, e.g., rhythmic patterns, are currently being investigated.
Unifying dynamic programming methods
Authors: Baker James K.
Published: October 2005

Dynamic programming has come to be widely used and accepted in automatic speech recognition. However, two different but similar applications have often been described more in terms of their differences than their similarities. On the one hand, dynamic programming is used to find the best nonlinear dynamic time warping to align two instances of a word. On the other hand, dynamic programming may be used to find the best state sequence for a hidden Markov process. Not only are these procedures essentially equivalent, but significant generalization comes from an explicit unification. Dynamic programming may be used not only to align two instances of a word, but also to align an instance of a word with an arbitrary finite state model for the word, or even to align two arbitray models. Multiple instances of a word may contribute to a single model, and multiple passes on a finite set of training data can be used to further refine word models.
Dichotic listening and stuttering severity: statistical significance or sampling error?
Authors: Blood Gordon W. , Blood Ingrid M.
Published: October 2005

The clinical importance of dichotic listening tasks (DLT) in stuttering should be established. The purpose of the present series of experiments was to determine the relationship between stuttering severity and dichotic ear preference scores. Experiment I, conducted in 1978, administered a DLT to 51 stutterers between the ages of 7 and 15 years. Results revealed that the right ear preference (REP) weakened as the severity of the stuttering decreased. Experiment II, conducted in 1980, involved 24 stutters between the ages of 8 and 13 years. Analyses revealed that the REP weakened as the severity of stuttering increased. Experiment III, conducted in 1982, with a sample of nine stutterers (ages 9–11 years) confirmed the results of experiment II. Results of different analyses (X2, phi coefficient, t test) for significance of ear preference and analysis of the data as one cumulative study or three independent studies will be presented. Discussed will involve the limitations of future research with DLT's stuttering and its usefulness as a clinical tool in light of these findings.