bluuebunny/crossref_metadata_2025 huggingface dataset:
Towed-model apparatus for investigating underwater cavity tones
Authors: Elder S. A. , Johnson B.
Published: October 2005

The problem of steady tones generated by flow over a rectangular cavity in a flat plate with a sharp leading edge has been studied previously in air, utilizing a low-noise wind tunnel [S. A. Elder, T. Farabeem, and F. C. DeMetz, J. Acoust. Soc. Am. 72, 532–549 (1982)]. Three tone mechanisms already identified are cavity resonance feedback, edge-force feedback, and cavity turbulence response, the resonance effect being predominant in air. To isolate edge-force effects, new studies are being done underwater where cavity resonance is less pronounced. The towed model system was chosen over the continuous water tunnel, on account of large background noise associated with the latter. Fiberglass model is fabricated into thin fairwater structure designed for good towing stability. Though possessing slight convex curvature, the vertical fairwater approximates a flat plate with 9° angle of attack. Cavity instrumentation includes fixed hydrophone for sound detection and motor-driven hot film probe for velocity profile surveys. Preliminary results are compared with similarity predictions from air data. [Work supported by Naval Sea Systems Command General Hydromechanics Research Program, administered by the David W. Taylor Naval Ship R&D Center, Bethesda, MD.]
A fast, simple, and robust nonlinear high resolution technique
Authors: Wagstaff Ronald A. , Berrou Jean-Louis
Published: October 2005

A simple and fast technique is presented that can produce high-resolution spectra from the output of a conventional processor (i.e., beamformer or spectrum analyzer). The speed and simplicity of the algorithm results from three factors: the input data of the system are not used, only the output data; the significant mathematical operations are simple additions and subtractions of log-transformed quantities, which makes the technique nonlinear in power; the nonlinearity (in power) of the technique eliminates the need for matrix inversions and permits the calculations to be done with four significant figures. The technique is robust, being relatively insensitive to system and measurement errors. The algorithm is explained and results are presented for both modeled frequency and spatial spectra and measured towed array sonar data.
An experimental study of signal-to-noise ratio losses due to OR-ing
Authors: Struzinski William A. , Bloom Jerrold C.
Published: October 2005

OR-ing experiments are, currently, being conducted on the NUSC Acoustic Display Research Facility. The objective of the experiments is to experimentally determine the signal-to-noise ratio losses when OR-ing N bins of acoustic data, and to compare these results with some of the recently published theoretically derived losses [W. A. Struzinski, IEEE Trans. Acoust. Speech Signal Process. ASSP-30 (4) (1982)]. This paper discusses the details of the OR-ing experiment software, and the results obtained to date. [Work Supported by the Naval Sea Systems Command.]
Tasks for a sensor-controlled robot
Authors: Scheide Alfred W.
Published: October 2005

Several advanced applications of industrial robots in the areas of material handling, welding, and assembly were described. Emphasis was placed on potential improvements that could be made in these application areas through the use of acoustic sensors to gain additional information about the processes and working environment of the industrial robot. The objective of this paper was to stimulate future innovative work in the application of acoustic sensors and sensing systems to industrial robots.
Acoustic transmission through atmospheres in industrial plants
Authors: Bass Henry E. , Bolen L. N.
Published: October 2005

The theory of sound propagation in gases in the frequency range 50 kHz to 50 MHz will be reviewed briefly with emphasis given to the range of attenuation and wavelengths one might expect. A typical industrial environment would be mostly air, but we will also consider an inert gas atmosphere near room temperature and a UF6 atmosphere at high temperature as extremes which might be encountered. Some recent results on background noise at frequencies between 100 kHz and 1 MHz will be presented, and implications of these results to signal-to-noise ratios in the ultrasonic region will be discussed.
Ultrasonic transducers for robotic applications
Authors: Massa Donald P.
Published: October 2005

A new family of ultrasonic transducers will be described that are especially useful as position sensing probes in many robotic applications. Low-frequency (20–40 kHz) transducers using small exponential horns achieve efficiencies in the order of 50%, and are used for long-range (50-ft) distance measuring. High-frequency (200- to 500-kHz) transducers utilizing planar mode vibrating ceramic discs in combination with quarter-wavelength acoustic waveguides eliminate secondary lobes and are particularly useful for making shorter range measurements and for achieving very high resolution (± 0.001 in.) at close ranges. The new transducers provide a wide variety of beam angles from virtually omnidirectional to 5° conical. The transducer designs to be described operate on either the fundamental or on a controlled overtone mode of the vibratile element, depending on the frequency region of operation required. Special sensing modules will be shown which include the interface circuitry permitting simple control of the ultrasonic sensing probe by logic signal commands produced by the robot's computer.
A narrow-beam probe using shock waves
Authors: Griebeler Elmer L.
Published: October 2005

A narrow-beam, noncontacting probe has been developed with about an 8-in. depth of field and a beam diameter of about 1/2 in. using a single shape-encoded pulse. After observing the performance and fate of a variety of walking cane and plunger type probes in industrial applications, the need and potential benefits of an acoustic noncontacting probe has become obvious. The results of about five years of off-and-on attempts at solving the inter-related acoustic, mechanical, and electronic problems have resulted in a computer optimized system. The goal was to form a well defined, easily recognized acoustic pulse focused into a long but narrow region of space with an overlapping sensing region for detecting echos from objects that come within this region. Subsequent research revealed that, to make a useful probe, the pulse needed to be confined with scatter kept below 20 dB and restricted in range by using a minimum transient frequency component well over 100 kHz. High-efficiency, mid-air acoustic processing was developed, including a lens design that greatly reduces the inverse square loss along the length of the beam shaped region, but which allows the pulse to be attenuated relatively quickly beyond the intended operating distance.
The bone-anchored bearing aid, a study of the mechanical point impedance of the buman head with and without skin penetration
Authors: Håkansson Bo , Tjellström Anders
Published: October 2005

The possibility of implanting a skin penetrating titanium screw into the mastoid portion of human skull and of establishing a permanent reaction-free skin penetration has made it possible to develop and attach a new bone conduction hearing aid directly to the skull. To understand and improve this new hearing system, the mechanical point impedance of the titanium screw-skull system ZMT has been measured. The conventional point impedance of the skin-covered mastoid portion of the temporal bone ZMS has also been measured and the difference in magnitude between the two impedances, 2010 log |ZMT/ZMS|, was calculated. Seven patients have been investigated. The magnitudes of ZMT were found to be between 10 and 35 dB higher than those of ZMS. One conclusion is that the conventional point impedance of the skin-covered mastoid portion of the skull is essentially due to the properties of the skin and subcutaneous soft tissues. Another conclusion is that much lower velocity levels are needed, with skin penetration, for a given hearing sensation, which makes it possible to develop an “all in one housing” bone-conduction hearing aid.
An electroencephalographic evoked response correlate of auditory adaptation
Authors: Cohen Ronald L. , Weiler Ernest M.
Published: October 2005

Normal-hearing subjects were evaluated to determine whether changes in behavioral auditory thresholds, in both a heterophonic and homophonic listening condition, would produce equal two dissimilar changes in the amplitude and latency of the auditory evoked potential. Each subject completed a preadaptation loudness balance prior to any stimulation. A 2000-Hz pure tone was then presented to the subject's right ear for a period of 7 min at a level of 60 dB SPL. During this 7-min adapting period four AER runs consisting of 32 tone pips superimposed on the adapting tone were presented. Finally, a perstimulatory loudness balance was completed while the adapting tone had been on for 7 min. Measurements of amplitude and latency of the evoked potentials were analyzed and results suggest that more adaptation occurred for the homophonic condition. However, a significant amount of decrement in the amplitude of the monaurally elicited AER was measured for both the homophonic and heterophonic condition. It was suggested that AER may be a sensitive measurement technique of auditory adaptation.
Sound pressure distribution in the human ear canal
Authors: Stinson M. R. , Shaw E. A. G.
Published: October 2005

The geometry of the human ear canal differs considerably from its conventional description as a uniform cylinder with a plane perpendicular end. The differences can be expected to produce significant effects especially at frequencies greater than 10 kHz. Measurements of sound pressure distribution within an accurately scaled (×2.5) replica of a real ear canal confirm the effects of the tapered end observed previously in simple models [J. Acoust. Soc. Am. Suppl. 1 71, S88 (1982)] and show that the standing wave patterns in the main body of the canal are appreciably disturbed. In agreement with the earlier work, the variations in SPL across the human eardrum must be expected to exceed 15 dB at 15 kHz. An approximate theory has been developed to describe the sound field within ear canals of varying cross section and direction. The theory is an extension of Webster's horn equation, quantifying the canal geometry using effective cross-sectional areas defined along an appropriate curved axis. Agreement between theoretical and experimental pressure distributions is good.
Effects of glottal termination impedance on formant frequencies and bandwidths
Authors: Boves L. , Cranen B.
Published: October 2005

The finding that synthetic speech, based on the customary linear model of acoustic speech production, often sounds somewhat machinelike, has motivated various attempts to refine the model. Our approach to the problem concentrates on variations in formant parameters due to the changing termination impedance at the glottis. Open and closed glottis intervals are determined from the electroglottogram that is digitized and recorded together with the speech signal. Linear prediction analysis of the signal segments taken from the closed glottis intervals leads to stable results for both formant frequencies and bandwidths. LP analysis of segments from the open glottis intervals leads to extremely erratic results, which can be proved to be due to pronounced spectral zeros. We will present numerical data on within-cycle formant variations in the speech of adult males obtained from cepstrally smoothed spectra. [Work supported in part by the Dutch Organization for the Advancement of Pure Research Z.W.O.]
Vocal tract analog synthesizer with lumped losses
Authors: Guerin Bernard , Degryse Daniel , Lherm Bruno
Published: October 2005

The vocal tract analog synthesizer generally requires a lot of calculation when a real signal is to be obtained. Thus, when you want to undertake studies, a real time synthesizer is more convenient. One way to carry out sound wave propagation modeling in the vocal tract is to consider the Kelly-Lochbaum model. In this case, we have to introduce a damping coefficient at the junction of the two elementary tubes. In order to decrease the computation size, we propose to concentrate the losses at three places along the vocal tract. These places are chosen so that the end characteristics (glottis, radiation) will not be disturbed. If there are about 20 elementary cells, the losses are places between cells number 2 and 3, 10 and 11, 14 and 15 (number 1 is next to the glottis). The simulation of this model, including the wall vibration and lip radiation according to the principle proposed by D. Degryse (The 4th F.A.S.E. symposium, Venezia, 193–196 (1981)], gives formant characteristics close to those of the distributed model: the maximum error is less than 1% on the first three formant frequencies and 20% on the respective bandwith. Then, the synthesizer requires about 25% fewer computations than without the improvement.
Speech synthesis-by-rule of Japanese generated in the syntactic base
Authors: Sato Shigeru , Kasuya Hideki
Published: October 2005

This report is the first attempt in Japanese speech synthesis generated in the syntactic base. While the authors have acoustic-phonetically analyzed an amount of speech data, they also have brough forth theoretical refinement to the rule system of Japanese phonology and syntax [J. D. McCawlay, The Phonological Compoment of a Grammar of Japanese (Mouton, The Hague, 1968); H. Yoshiba Ling. Analysis 7, 241–262 (1981)]. Both seek a generative model for the speech data and for verifying the theory that a speech synthesis-by-rule system is now in progress comprising phonological rules as well as other higher-level generations. It is shown that purely phonological interpretation of some of the forms yields ad hoc rules and complicates the system, and hence that by restructuring the syntactic/phonological organization remarkable simplification in the phonological rules can be attained. It is expected that this significant modification in the theory can bring constructive generalization and rearrangement to the synthesis-by-rule system of Japanese.
The “morphology” of English spelling: A look at the SRS text-modification rules for English
Authors: Hertz Susan R.
Published: October 2005

Any scheme that accurately converts English text to phonemes must be based on morphological analysis—i.e., an analysis of words into prefixes, suffixes, and roots. This paper describes the SRS (Speech Research System) strategy for marking the morphohogical structure of words. Instead of a morpheme dictionary, SRS uses text-modification rules, which divide words into morphological units on which subsequent phoneme-generating rules are based. Because these units are not always true morphemes (for example, fibe + er for fiber on analogy with bribe + er for briber), they are more accurately cared “spelling morphs.” An analysis of text into spelling morphs simplifies not only phoneme and stress prediction (for example, the different pronunciations of im + bed vs climb + ed), but also the handling of exceptions (for example, the irregular pronunciation of cover in re + cover, discover, un + cover, and cover). In addition to such obvious advantages as saving space, using rules instead of a dictionary makes it easy to mark both the syllable structure and the morphological structure when the two disagree (for example, dis + cover for discover).
Simulator investigation of the effects of occluded ear characteristics on remote measurement of eardrum SPL
Authors: Gilman Samuel , Dirks Donald , Hanson David G.
Published: October 2005

Because of developments in remote measurement of eardrum sound pressure level (SPL) in hearing aid occluded ears, an investigation was made of the effects of external ear characteristics on such measurements. Two systems were investigated: a probe microphone contained within the earmold, and a miniature microphone in the ear canal located as close as possible to the medial end of the earmold. Ear stimulators (ANSI S3.25-1979) were modified to have a range of eardrum impedances approximating 96% of the population with normal middle ears [Zwislocki and Feldman, ASHA Monogr. No. 15 (1970)] in three steps and to have a range of ear canal lengths of 9.7 to 17.7 mm in four steps. SPL measurements were made over a frequency range of 0.2 to 6 kHz for each combination of length and impedance. As predicted by transmission line calculations, both ear canal length and eardrum impedance affected the measured SPL. Although individual measurements made by each system differed somewhat because of differences in measurement location, the total range of differences between measured and actual eardrum SPL was almost identical for both systems. With optimum calibration in an ear simulator, the maximum error produced by either system was within ±5 dB of the true eardrum SPL.
Pathological sensitivity enhancement in the tails of neuronal tuning curves—A possible explanation
Authors: Zwislocki J. J.
Published: October 2005

In many pathological cochleas the tips of the tuning curves of cochlear nerve afferents are desensitized, but the tails are sensitized. This apparent paradox can be explained through likely micromechanical events in the organ of Corti. If the excitation of the inner hair cells is due to shear motion between the reticular lamina and the tectorial membrane, and the tectorial membrane can vibrate in two modes—parallel and nearly orthogonal to basilar-membrane motion—both the amplitude and phase of the hair-cell excitation at the tail frequencies can vary over a wide range. This is so because, as we have shown in the past, the amplitude of the radial vibration of the tectorial membrane at the location of the inner hair cells may be nearly equal to that of the reticular lamina, leading to a vanishing shear motion between them. When the mechanical coupling between the organ of Corti and the tectorial membrane is weakened pathologically, the tectorial-membrane amplitude should be decreased and the shear motion enhanced with the resulting enhancement of the hair-cell excitation. [Work supported by NIH.]
External ear effects during early development in the kitten
Authors: Walsh Edward J. , McGee JoAnn , Javel Eric
Published: October 2005

At birth, the auditory system of the kitten is grossly immature; however, during the first ten postnatal days the external ear becomes patent, the middle-ear cavity clears, ossicular calcification progresses, and the cochlea differentiates. The effect of the closed external ear as an attenuator during early maturation was measured using auditory brainstem evoked response (ABR) comparisons between operated and intact animals. Kittens were anesthetized and the external ear was resected to the level of the tympanic annulus exposing the tympanum. Using equal intensity stimuli, ABRs recorded from operated animals were observed as early as the second postnatal day, about six days earlier than from intact animals. The frequency range over which responses occurred was smaller than the range exhibited in older animals. Attenuation effects of 10 to 20 dB were observed in the mid-to-low frequency range; however, above 2.0 kHz high thresholds (120 dB SPL) precluded our ability to elicit responses in the intact animal. Thus, although external ear attenuation effects were observed, the immaturity of the system during the first postnatal week minimized its effect, particularly for frequencies above 2.0 kHz. [Work supported by NINCDS.]
Topographic organization of interaural-intensity-difference sensitivity in deep superior colliculus
Authors: Wise Lisa Z. , Irvine Dexter R. F.
Published: October 2005

The sensitivity to interaural intensity differences (IIDs) of acoustically responsive neurons was examined in multiple microelectrode penetrations along the rostro-caudal axis of deep superior colliculus (SC) in anesthetized cats. The most rostrally located neurons were unresponsive to monaural stimulation but were strongly excited binaurally. IID sensitivity functions for these “predominantly binaural” (PB) neurons were sharply peaked with maximum response at or near zero IID. Caudal to the PB cell area, most neurons were excited by contralateral stimulation and inhibited by ipsilateral stimulation (EI cells). All EI cells were maximally excited over a range of IIDs favoring the contralateral ear and were totally inhibited over a range favoring the ipsilateral ear. The position of the cutoff between excitatory and inhibitory regions was located near zero IID (corresponding to the median plane) for the most rostrally located EI neurons and was shifted progressively to IIDs favoring the contralateral ear (corresponding to increasing contralateral azimuths) for more caudally located EI neurons. The consequence of this topography is that IID, and hence azimuthal location, is represented by the extent of deep SC activated.
ABR multielectrode recording: Determination of source position movement through cat brainstem
Authors: Gaumond R. P.
Published: October 2005

The click-evoked acoustic brainstem response (ABR), average of 512 responses, was recorded at 18 sites on the scalp of anesthetized cats. Scalp recordings were used to determine source position by attributing the scalp-surface potential field at each instant to the presence of a single current-dipole equivalent source, immersed within a spherical, homogeneous volume conductor. At each sampling instant, equivalent source position was determined by the method of Schneider [IEEE Trans. BME-21, 52–54 (1974)]. At those instants corresponding to vertex-positive response peaks, the single-source model was generally consistent with observed field patterns. Plots of equivalent source position along the axis from head center through vertex between 2.5 and 6.0 ms following the click. Movement along other axes is less prominent. Lack of movement along the centro-lateral axis may reflect the activity of sources arrayed symmetrically along either side of the brainstem, while the lack of expected ventral movement is attributed to limitations of the spherical head model assumed. [Work supported by the Whitaker Foundation.]
Benchmark design process for machine tool structures
Authors: Kamman Joe
Published: October 2005

Traditionally the technical aspect of the machine tool design process has been handled by the more experienced veteran engineer. Although this is still largely true today, the power of the computer is changing many aspects of this design process. The veteran engineer's experience and knowledge of past and current products still puts him in a good position to coordinate the new design process. However, the computer has provided us with many added tools to incorporate into this process. These tools will supplement our experience, allow us to quantify it, and allow us to use knowledge of past products to predict performance of the new ones. Particularly in the areas of stress, statics, and machine tool dynamics such computerized methods as finite elements, system modeling, and modal analysis are proving themselves to be invaluable tools. With them the modern design team can achieve an understanding level of their new product never possible in the past. The best features of various designs can be selected, combined in the form of a prototype, and tested with the confidence that all the initial goals will be met.
Optimization and application of unequally spaced milling cutters
Authors: Gygax Paul E.
Published: October 2005

Milling, the most efficient material removal process, is an interrupted cutting operation whose high power is concentrated at a few discrete frequencies (tooth impact harmonics). The following presentation focuses on the dynamics of the cutting process as a primary source of vibration. The behavior of multitooth cutters is first reconstructed from the individual force pulse and continuous spectrum of one single tooth. This procedure is readily extended to unequally spaced cutters which break the regular rhythm of successive cuts and randomize the excitation by spreading its energy over more frequencies (spindle speed harmonics). The analysis leads to the frequency response function of the cutter. Properties of such cutter signatures are discussed (periodicity, symmetry, influence of tooth number, sensitivity to out-of-roundness errors, etc.). Optimization criteria, strategies, and procedures are then derived and evaluated. The presentation concludes with potential applications of unequally spaced cutters. The most promising one, besides reduction of forced and self-excited vibrations, is testing of operating milling machines with cepstral and/or spectral averaging methods in order to validate model data obtained by artificial excitation at standstill.
A new accelerometer-based vibration monitoring system
Authors: Usher Theron
Published: October 2005

A recent standard of the American Petroleum Institute [API Standard 678 (May 1981)] has generated new designs of dedicated machinery monitoring systems utilizing piezoelectric accelerometers. These systems are dedicated in the sense that all adjustments and switch selections must be made internally. Accelerometer-based systems are often chosen to provide wideband monitoring and alarm functions for such applications as rotating machinery with high shaft-to-bearing transmissibility, gear condition in speed changers, blade passage phenomena, rubbing, cavitation, and reciprocating compressor forces. This paper includes a brief review of the pertinent requirements in API Standard 678. It also summarizes the development and features of one system designed in response to the standard.
Nonlinear bandlimited oscillations
Authors: Davies Huw G.
Published: October 2005

The response of an oscillator with a nonlinear spring is considered when the excitation force is white noise filtered through a first-order, low-pass filter. When the damping is light, one term in the Fokker-Planck equation for the joint density function of displacement, velocity and acceleration is negligibly small. The resulting approximate Fokker-Planck equation can be solved exactly. An equivalent damping that keeps the power input to the oscillator constant is used to extend the range of validity of the approximate solution. A density function for the distribution of peaks is obtained that covers both narrow-band and broadband response. A numerical example of the nonlinear peak distribution is calculated for the case of an oscillator with a bilinear spring. The results are extended to the continuous case, using as an example a beam impacting springlike stops. [Work supported by NSERC.]
Response of self-excited two-degree-of-freedom systems to multifrequency excitations
Authors: Nayfeh Ali H. , Asfar Khalid R. , Mook Dean T.
Published: October 2005

The response to multifrequency excitation of a two-degree-of-freedom, self-excited system is analyzed by using the method of multiple scales. Three cases of resonance are considered superharmonic, primary, and subharmonic. Also considered is the case of simultaneous superharmonic and subharmonic resonance. Steady-state and stability analyses are carried out for each case and numerical results are presented showing the influences of the several parameters.
The many linguistic functions of duration
Authors: Lehiste Ilse
Published: October 2005

This paper surveys the many ways in which duration serves to distinguish between different words and between higher-level linguistic units. At word level, durational differences that result in a difference in meaning are commonly analyzed as due to distinctive segmental quantity. This paper considers some instances in which segmental quantity is subordinated to durational patterns with larger domains, and cases in which duration interacts with F0 in establishing distinctions between words in languages usually not assumed to possess distinctive tone. At higher levels, duration plays a part in signaling boundaries of units as well as in establishing the identity of the unit. Interaction with F0 at these higher levels is likewise considered.
Perceiving the natural structure of coarticulated speech
Authors: Fowler Carol A.
Published: October 2005

Two problems for theorists of speech perception are those of segmentation and invariance. Both arise from the talker's overlapping productions of phonetic segments. Because segments are coarticulated, there are no boundaries between segments, and acoustic intervals dominated by properties of individual segments are context sensitive. Based on other perceptual research, an approach to the segmentation and invariance problems is proposed. I propose that listeners detect segment boundaries wherever acoustic information for individual segments begins or ends, and that thereby they preserve information about articulatory overlap; they do not impose boundaries between segments. That listeners respect articulatory overlap and use information for a phonetic segment within the acoustic domain of a predecessor is shown by recent research. This approach has implications for the invariance problem. If listeners use anticipatory information for a segment A within the acoustic domain of B as A information, perhaps the information does not, then, contribute to the percept of B. If listeners indeed “factor” A information from B, B will sound invariant over different As although its acoustic domain is context sensitive. I will present research in support of this hypothesis. [Work supported by NSF Grant BNS-8111470.]
Studies of phrase-level speech timing
Authors: Weismer Gary
Published: October 2005

In a previous presentation [G. Weismer, Speech Motor Control Conference, Madison, WI (April-May 1982)] data were presented which suggested that temporal ratios of acoustically defined events in a five-syllable utterance were approximately constant across speaking rate adjustments. These findings were related to similar results from typing sequence research, and to general notions of temporal sequence organization by the central nervous system. For the present paper additional data on relative phrase-level timing are described, both from the perspective of normal and disordered speech production. In particular, phrase-level timing characteristics of speakers with neurogenically based disorders are described, because the specific disorders are often thought to involve aberrant motor programming. Discussion will not only be devoted to the implications of the results, but also to the selection of “events” in a phrase-level sequence for use in the construction of temporal ratios. Finally, some attempt will be made to relate this work to a model of speech timing proposed by Fujimura [In Honor of Ilse Lehiste, edited by R. Channon and L. Shockey (in press)]. [Supported by NINCDS Award NS 13274.]
Timing rules in Klattalk: Implications for models of speech production
Authors: Klatt Dennis H.
Published: October 2005

The Klattalk text-to-speech program contains a system of rules for generating synthetic speech from an abstract representation that includes some syntactic markers, lexical stress, and phonemes. The program is a valuable tool for the evaluation of timing rules in English. It will be demonstrated that correct timing specifications at the acoustic level are essential to sentence comprehension and acceptability. The current rule system will be described. It contains a set of interacting rules which, individually, work well, but in combination, can still cause an occasional segment to be perceived as too long or too short. This nonindependence of rule action is captured in part by an incompressibility constraint, but additional constraints, or rules, appear to be needed. Theoretical implications of this approach and its relationship to other timing models will be discussed. [Work supported by NIH.]
Some temporal rules in speech production
Authors: Harris K. S. , Tuller B. , Bell-Berti F. , Kelso J. A. Scott
Published: October 2005

A central preoccupation of theorists since the beginning of the modern explosion in speech production research has been formulating the underlying rules which allow speakers to concatenate message units and overall temporal properties of utterances (Ohman, 1966, 1967; Lindblom, 1963). We argue that, in general, when stress and speaking rate was varied, relational invariance is preserved over substantial changes in the absolute duration of articulatory events, and that rules which preserve temporal structure can be used to describe coarticulatory changes. [This work was supported by NIH Grant NS-13870 to Haskins Laboratories.]
The role of fundamental frequency and duration in the perception of clause boundaries: Evidence from a speeded verification task
Authors: Luce Paul A. , Charles-Luce Jan
Published: October 2005

The perception of clause boundaries is determined by at least three suprasegmental variables: fundamental frequency, duration, and amplitude [Streeter, J. Acoust. Soc. Am. 64 (1978)]. We investigated the relative contribution of F0 declination and duration to clause boundary perception. Pairs of sentences were constructed so that both sentences of a pair were lexically identical up to a point at which a clause boundary occurred in one member of the pair; in the other member, the clause boundary occurred later. All sentences were truncated immediately after the last word shared by both sentences of a pair. Thus the pairs of truncated sentences were identical except for the suprasegmental cues indicating a clause boundary. In a second manipulation, truncated sentences were produced in which F0 and duration conflicted. In a speeded verification task, subjects decided if an auditorily presented truncated sentence matched a visually presented, intact sentence. The results of this experiment indicate that whereas F0 may facilitate clause boundary perception, duration is the decisive cue. [Work supported by grants from NIMH, No. MH-24027, and NINCDS, No. NS-12179.]
Coherence and the beam of sound projected into an acoustically fast fluid bottom from an overlying sound-bearing fluid wedge
Authors: Coppens A. B. , Humphries M.
Published: October 2005

The problem of the dumping of acoustic energy from a fluid wedge into an acoustically fast fluid bottom in the form of a highly collimated beam of sound has been approached by use of the method of images and the Green's function integral. With the assumption of phase-coherence among the images, computer generated results were obtained [A. B. Coppens and J. V. Sanders in Bottom-Interacting Ocean Acoustics, IV (SACLANT ASW Research Center, La Spezia, Italy, 1980), Vol. 5, p. 439]. Subsequently, with the help of a simple approximation to the Green's function integral, we have been able to study the case of phase incoherence among the images. Analysis leads to a simple closed-form expression for the beam of sound in the bottom. Its properties are compared with our previous results and the results of a different approach to this problem [A. D. Pierce, J. Acoust. Soc. Am. 72, 523 (1982)]. It is suggested that phase coherence is not, surprisingly, a very important aspect of the problem. [Research supported in part by NPS Foundation Research Program.]
A methodology for evaluating loss of array coherence
Authors: Markowitz A. E.
Published: October 2005

A methodology is presented for distinguishing between propagation and sensor effects as the cause of loss of signal coherence over a line array of acoustic sensors. Using signal-to-noise ratio (S/N) as a measure of coherence auto- and cross-spectral properties of the data are combined into interesting relationships which form the basis of this methodology. It is shown that γlm2 = γlγm = ((S/N)(S/N) + 1)l ((S/N)(S/N) + 1)m, (1) and that ((S/N)(S/N) + 1)¯i2 = 1z(z − 3)/2  ∑ y > z ∑ z ≠ i γ2izγ2iyγ2zy, (2) where γlm2 is the magnitude squared coherence between points l and m along the array and γl is the signal coherence magnitude at point l along the array. The directly measured coherence matrix is compared with the matrix obtained from measured S/N via Eq. (1). The reverse process, Eq. (2), is not unambiguous and average values are therefore obtained.
Modeling shear and compressional head waves at velocity steps and inversions
Authors: McAulay Alastair D.
Published: October 2005

Accurate modeling of head waves (lateral waves or refractions) enables improved estimation of sound velocities at and below the ocean bottom, [A.D. McAulay, in Int. Conf. Acoustics Speech and Sig. Proc. 1980, 1981]. Head waves arise when a homogeneous wave strikes an interface with a medium of greater velocity (step) at greater than the critical angle or, less well known, when an inhomogeneous wave strikes an interface with a medium of lower velocity (inversion). In practice head waves are caused by an inhomogeneous wave either when a point source is located near a velocity inversion, or at the exit of a tunnel through a high velocity region. A computer program was developed using the Thompson-Haskel formulation and the Delta matrix method to obtain fast accurate modeling of shear and compressional waves for a plane layered model and a point source. Various velocity models were used to illustrate the accurate modeling of shear and compressional head waves. In particular the accurate modeling of head waves at a velocity inversion is illustrated for the first time.
The FRAM IV vertical array—Design and performance
Authors: Giellis G. R. , Yang T. C. , Votaw C. W.
Published: October 2005

A vertical array was developed at NRL fur use in the eastern Arctic as part of the FRAM IV experiment. A primary objective was the measurement of the vertical sound field for long range coherent signals. The vertical directionality of ambient noise in the Arctic environment was also investigated. The array was 960 m long and had 28 working hydrophones. The design featured a fully balanced, current-mode signal transmission system, which provided maximum rejection of electrically induced interference. Provisions were also made for the reduction of flow noise and of cable strumming noise. The array was installed in late March 1982, at the site of the FRAM IV ice camp. It was operated successfully throughout a four week schedule of experiments. The recorded data exhibited a high signal to noise ratio.
Preliminary results of the FRAM IV vertical array experiment
Authors: Yang T. C. , Giellis G. R. , Votaw C. W.
Published: October 2005

A vertical array of 960-m length was deployed in the 1982 FRAM IV experiment. The 28 hydrophones of the array measured the vertical sound field of signals and noise in the Arctic Ocean. The depth distribution of intensity, vertical directionality, and vertical spatial coherence were estimated from the data. Some preliminary results will be presented. The data clearly indicates the presence of normal mode structure in the vertical distribution of the signal. Lowest order mode amplitudes and phases were successfully derived from the shot data measurements.
Occupational noise exposure and the epidemiology of high blood pressure
Authors: Talbott Evelyn , Matthews Karen , Helmkamp James , Redmond Gerald , Margolis Helene
Published: October 2005

The study population consisted of 196 male hourly workers from a noise exposed (>88 dBA) and 169 from a less noise exposed plant. Participants were 40–63 years of age and had at least 10 years of exposure. Clinical exams consisted of height, weight, pulse, blood pressure, audiometric testing, and occupational history. Body mass index, alcohol intake, and family history of hypertension were comparable for the two groups. There was a significant difference in the proportion of men with severe noise induced hearing loss (SNIHL) (>65 dB loss at 3, 4, and 6 kHz) between the two plants, as well as an increase in the number with hearing impairment. Other definitions of noise induced loss were explored and yielded similar results. A two-way analysis of variance was conducted with mean systolic and diastolic blood pressure as the outcome variable with age as a covariate. The main effects of plant and SNIHL were found to be significant at p<0.05 for diastolic blood pressure only. There was a greater proportion of hypertensives in the 56 + age group with severe noise loss in both plants (p<0.003).
Three-dimensional acoustic finite element analysis
Authors: Chino J. J.
Published: October 2005

Three-dimensional acoustic modeling is a novel application of finite element analysis (FEA) techniques and the successful use of FEA modeling towards the simulation of an empirically tested prototype assembly is described. The finite element analysis method is well known for its applications in structural design. The technique has attained widespread usage in numerous industries and it has enabled engineers to make structural design decisions long before the availability of test hardware or prototypes. The finite element methods, however, are not restricted to structural applications. In fact, this method provides a general approximation scheme, based upon the representation of a continuum by a finite number of interconnected elements. The method is applicable to numerous nonstructural problems and for many of these, the element properties have the same basic features as in the structural model. This paper describes the successful execution of finite element analysis techniques for three-dimensional acoustical analysis.
Vibration qualification of a photodetector susceptible to microphonic noise
Authors: O'Hearne Clifford S.
Published: October 2005

A missile-borne photodetector is used to receive uplink guidance information carried on a laser beam. It is supported on a structure which vibrates intensely in response to aeroacoustical excitation. Detectors of this construction have been found to have widely varying susceptibility to vibration in the degree to which they exhibit microphonic noise. Each manufactured detector is subjected to 2–20 kHz random vibration on a piezoelectric shake table and is rejected for use if its noise output exceeds the allowed level. The development of the environmental requirement is reviewed and the production test equipment is described.
Rayleigh waves excited to acoustic emission
Authors: Pott John , Harris John G.
Published: October 2005

Analytical results are presented that describe the Rayleigh waves excited by the acoustic emission from a crack that suddenly starts to grow. At first a two-dimensional crack is considered, then the results are extended to three dimensions. The plane of crack growth is inclined to the surface. The fracture mode can be brittle or some plastic yielding at the crack edge can take place. The analysis consists of two parts. First, wavefront approximations to the acoustic emission are calculated. Second, representation integrals for the Rayleigh waves are constructed using the compressional and shear emissions in combination with an appropriate Green's tensor. The integrations are taken over the wavefronts of the emitted signals and the integrals are evaluated asymptotically. Graphs of the resultant surface motion are given. [Work supported by NSF(MEA-8104738).]
Natural resonance caused by fracture-induced venting: An analysis of multichannel acoustic logging in a deep borehole
Authors: O'Malley Honor , Anderson Roger N.
Published: October 2005

A four-channel sonic log of a 1-km section of oceanic crust at site 504B, Eastern Equatorial Pacific Ocean has been spectrally analyzed. The cross-spectrum of seismic waveform pairs was converted into integrated energy and peak-frequency at each shot point for both compressional and shear coda. These plus an ultrasonic borehole televiewer fracture count were combined into a “fracture spectrum log” over the bottom 500 m of the borehole. Cross correlation of P and S frequency and energy variations downhole revealed a natural resonance at wavelengths of 30–40 cm in the oceanic crust. This resonance appears as a low-pass filter of P energy and a high-pass filter of S energy. Because the energy attenuation is phase coherent with the fracture density, we interpret this natural resonance to be caused by venting of acoustic energy through intersecting fractures of an aspect ratio appropriate to the resonance wavelength. This is the first time such uniform fracturing has ever been observed in the oceanic crust. [Work supported by ONR.]
Fault diagnosis of diesel engine valves using inverse filtered vibration signals
Authors: Bowen David L. , Lyon Richard H.
Published: October 2005

The condition of a valve in a diesel engine was determined by inverse filtering the vibration response of the engine block to a valve/valve seat impact. The source power spectra for a “normal” and a “faulted” impact (simulated by a thin strip of tape between the valve and seat) were estimated by measurements of a longitudinal stress wave produced by the valve impacting a valve seat machined into the end of an 11-ft-long steel rod. A modal analysis of the valve under free-free boundary conditions revealed that the dominant modes were a bending mode at 5.5 kHz (more prominent for faulted impacts) and a longitudinal mode at 16 kHz (more prominent for normal impacts). The transfer function between the valve seat in the engine and a “convenient” point on the block was estimated by a smoothed average of four separate transfer functions measured from different points around the seat to the point on the block. Next, the value was impacted in the engine and the magnitude of the inverse average transfer function was used to filter the power spectrum of the resulting acceleration measured at the point. The procedure was able to reconstruct the two major modes, thus enabling a fault diagnosis to be made by comparing their relative magnitudes.
Experimental and theoretical study of power flow between randomly vibrating welded plates
Authors: Dimitriadis Emillos K. , Pierce Allan D.
Published: October 2005

The authors have previously described [J. Acoust. Soc. Am. Suppl. 1 72, S77 (1982)] a modal analysis technique for estimating power flow between welded plates in a form compatible with statistical energy analysis (SEA) formulations. The theory is here extended to point excitation of simply supported and freely suspended welded thin plates. This is used to assess various theoretical hypotheses concerning the dependence of the power flow between the plates on internal damping, excitation position, and structural dimensions. The general formulation allows a study of contributions from cross-correlation between modes of the two plates and other effects that are customarily neglected in SEA studies. The accompanying experimental study is with an L-shaped aluminum plate suspended by strings and driven by an electromagnetic vibration exciter, successively attached at a number of points on the plate. Measurements are reported for power input and vibrational energy in each plate and subsequently used to calculate apparent coupling loss factors. The latter are compared with the theoretical estimates. [Work supported by Whirlpool Corporation.]
A variational formulation of acousto-structural vibrations, with application to flutter problems
Authors: Sarigül Nesrin , Dökmeci M. Cengiz
Published: October 2005

Presented herein is a general variational formulation for acousto-structural vibrations and its application to elastic plates. First, a variational formulation for gas-structure media is obtained by means of Hamilton's principle under the usual constraint conditions [B. Tabarrok, Int. J. Num. Meth. Eng. 13, 197–201 (1978)]. Then, the constraints, frequently undesirable in computation, are relaxed through Friedrichs's transformation, as illustrated by the authors [Finite Elements in Water Resources, (Springer-Verlag, New York, 1982) pp. 2.101–110]. Thus a variational formulation with no constraints is established for acousto-structural vibrations. Next, the field equations governing the motion of elastic plate in a potential gas flow are derived by using the unconstrained variational formulation together with the kinematic hypotheses of R. D. Mindlin [J. Appl. Mech. 18, 31–38 (1951)]. The governing equations accommodate all the types of motions of the plate. Further, the sufficient conditions are enumerated to ensure the uniqueness in solutions of the governing equations by the logarithmic convexity arguments. [Sarigül's work was supported by the Amelia Earhart fellowship.]