bluuebunny/crossref_metadata_2025 huggingface dataset:
Implementation of focalization for processing field data
Authors: Collins Michael D. , Kuperman W. A. , Perkins J. S. , Kristensen Åge
Published: October 2005

It is impossible to localize an acoustic source in the ocean with conventional matched-field processing techniques if there are significant uncertainties in the acoustic parameters of the environment. Focalization is an approach for overcoming this difficulty that involves searching for key environmental parameters in addition to searching for the source location [Collins and Kuperman, J. Acoust. Soc. Am. 90, 1410–1422 (1991)]. The feasibility of this approach was investigated using implementations that might be difficult to apply to data. An implementation of focalization involving the parabolic equation method for computing replica fields and eigenvector-based signal processing is applicable to problems involving data from complex ocean environments. The covariance matrix is formed for a frequency corresponding to a peak in the source spectrum. The input data to the processor is the eigenvector corresponding to the largest eigenvalue. A large signal-to-noise gain is achieved with this approach, and the eigenvector is often a good representation of the true source field. To determine the source location, the eigenvector is compared with replica fields using a cost function that is minimized over the environmental parameter space with simulated annealing.
Localization of the sources of short-duration signals in a coastal ocean environment
Authors: Miller James H. , Chiu Ching-Sang
Published: October 2005

A time-domain method to localize the sources of short-duration acoustic signals is presented. Pulse propagation in a coastal ocean environment has been successfully modeled on a workstation using a modified version of the finite element parabolic equation (FEPE) code developed by Collins and Westwood (1991). The agreement between the modeled results and experimental data prompted the consideration of the problem of localizing sources of short-duration acoustic signals using time-domain processing. A new algorithm called time domain localization (TALON) has been developed. TALON uses the modified FEPE along with a time domain correlation technique to determine the range, depth, and bearing of the source of the short-duration signal. TALON is currently configured to use a single sensor or beam but can be modified for a multisensor/multibeam application. TALON is in the class of algorithms that have been described as coherent matched-field processors. In this paper, the localization procedure and the results of two computer simulation experiments are presented. The computer experiments illustrate localization performance for Monterey Bay and Barents Sea scenarios. Algorithm sensitivity to signal and environment uncertainty is addressed. [Work supported by the Naval Postgraduate School.]
Estimation of acoustic source signatures in an ocean waveguide
Authors: Finette S. , Mignerey P.
Published: October 2005

Some experimental results are presented on source signature extraction using vertical array data obtained from the acoustic field emitted by a point source in a deep ocean waveguide. Passively sensed field data obtained with a large aperture vertical array is processed by wideband, coherent, and incoherent matched-field methods, and a multichannel deconvolution approach in order to localize the source and reconstruct its temporal signature [P. Mignerey and S. Finette, J. Acoust. Soc. Am. 92, 351–364 (1992)]. A wideband normal modes code was implemented on a fine-grained parallel computer and used to calculate both the replica field vectors and the deconvolved estimates of the source signature. Signals in the band 25–110 Hz are processed for sources located at either the first or second convergence zones, approximately 48 and 97 km from the array. Normalized cross correlations between the measured signatures and their deconvolved estimates indicates that faithful reconstructions of the source signature can be obtained by processing a subaperture of the array data. The method depends on reliable estimates of the ocean Green’s functions linking the source and receivers, which in turn depend on good environmental knowledge concerning the ocean waveguide properties. [Work supported by ONR and NRL.]
Mode decomposition for a three-dimensional sparse array
Authors: Bogart Christopher W. , Yang T. C.
Published: October 2005

Matched-mode processing determines the source location (range, depth, and bearing) by matching the mode replica function with the mode amplitude decomposed from data. Mode decomposition has been successfully demonstrated for a vertical array or multi-vertical arrays that constitute a (three-dimensional) filled volumetric array. For these cases, the intrinsic difference of the matched-field and matched-mode processing lies in how the normal modes are processed (decomposed) by each method. For a sparse three-dimensional array such as a slanted vertical array, a slack array, or a horizontal line array, mode decomposition is not a straightforward process. A brute force and an approximate mode decomposition method are presented here; the latter has the simplicity of requiring only one mode decomposition for matched mode processing for all bearings. It is also shown that the performance of a sparse array is related to the accuracy of mode decomposition as previously demonstrated for a vertical array.
Recent technical enhancements to the fundamental digital sampling synthesizer with emphasis on vector synthesis, advanced vector synthesis, and wave sequencing
Authors: Limberis Alex
Published: October 2005

Sampling technology is the basis for the vast majority of the current generation of electronic musical synthesizers. Almost all of the present day electronic music synthesizers are based on sampling technology in one form or another. Over the past several years, many innovative enhancements have been added to the basic sampling technology. Each manufacturer has added their own special ‘‘tricks’’ to the fundamental underlying sampling technology. This presentation will review basic sampling technology and briefly mention techniques used by several current manufacturers. The presentation will focus on vector synthesis introduced on the sequential prophet VS synthesizer and advanced vector synthesis introduced on the Korg Wavestation series of synthesizers. Vector synthesis is a synthesis method which uses four sampled waveforms blended together through a two-dimensional vector mixer. Advanced vector synthesis combines vector synthesis with wave sequencing. Wave sequencing is a technique for sequencing a series of sampled waveforms in order to create complex timbers which evolve over time. Audio examples will be presented.
Predicting sound-pressure levels in the near field and far field from radiating partitions and building elements
Authors: Sharp Ben H. , Hessler George F.
Published: October 2005

One of the most common tasks of the noise control engineer is predicting sound transmission and noise reduction through structures and the sound radiation from the structures to the near and far fields. Concerning transmission loss, a large database of test data (at least from 100 to 5000 Hz), verified transmission loss theory, and practical equations for relatively simple constructions are all readily available to the noise control engineer. On the other hand, the topics of noise reduction from space to space as a function of transmission loss, and radiation patterns from partitions and building elements are rarely found in the literature or acoustical texts. A large variety of noise reduction situations arise in practice. Sound may transmit from diffuse spaces, semireverberant, or highly absorbent spaces to similar or dissimilar spaces including the outdoors. The transmission path is likely to be through the primary surface as well as windows, ventilation openings, or a combination of different wall and roof constructions. At times, the transmission from outdoors to interior spaces is the primary consideration. Often, in industrial situations, the level close to the structure is of interest for hearing conservation considerations, as is the level in the far field for community impact assessments. This paper reviews noise reduction equations as a function of transmission loss drawn from many sources for predicting sound-pressure levels in the situations discussed above. The radiation pattern from a plane wall or opening to the near and far fields is discussed.

The material is presented in an orderly fashion to serve as a guide for the noise control engineer. Recommendations are given for situations in which current analytical prediction methods appear inadequate.
Predicting outdoor noise propagation
Authors: Bass Henry E. , Sutherland L. C.
Published: October 2005

The propagation of sound outdoors involves spreading, absorption, interaction with the surface, refraction, diffraction, and scattering. Although there is still no widely accepted prediction scheme that includes all of these effects, the understanding of individual phenomena and contribution to transmission loss along the propagation path has steadily advanced. Transmission loss due to atmospheric absorption is standardized. The effects of real ground surfaces are predictable for a wide range of surfaces. Modern numerical techniques described in a different special session allow one to predict refraction and diffraction in an inhomogeneous atmosphere. These predictions are far superior to simple geometrical optics ray tracing at moderate and low frequencies. Diffraction by barriers of regular geometry can be predicted analytically but irregular shapes require further developments. Scattering by atmospheric turbulence as well as foliage requires additional work prior to providing engineering estimates well based upon experiment.
Oral-laryngeal timing patterns in voiceless stops
Authors: Cooper André M.
Published: October 2005

Voice-onset time (VOT) in voiceless stops is typically said to be shortest for labial stops, intermediate for alveolars, and longest for velars. One proposed explanation states that oral release is actively coordinated relative to the onset of glottal adduction (OGA) to produce place-related VOT differences−specifically, the earlier the release, the longer the VOT. The present study uses transillumination and acoustic data to test this claim directly. The results confirm that oral release tends to be timed earliest for /p/ relative to the OGA, intermediately for /t/, and latest for /k/, thus reflecting the rank order of VOT differences. Interestingly, another pattern of oral-laryngeal timing emerged from the data. Despite place-related differences in the magnitude and duration of the devoicing gesture, the OGA occurred at a fixed interval after oral closure across all stop categories for one speaker and at a fixed interval after closure across /p,t/ for the other. Thus while oral release may be coordinated relative to the OGA, the OGA itself appears be fixed relative to oral closure. [Work supported by NIH Grant DC-00121.]
An x-ray microbeam study of the effects of stress and position-in-syllable on consonant articulation
Authors: Turk Alice
Published: October 2005

This paper reports on an investigation of the effects of lexical stress and position-in-syllable on the kinematics of the primary articulators involved in the production of bilabial and alveolar consonants in American English. Data for the study were collected at the x-ray microbeam facility in Madison, Wisconsin, and consist of movement trajectories of pellets attached to the lips and tongue tip during the production of /p, b, t, and d/ in real words for two speakers of American English. Effects of stress and position-in-syllable on the relationship between peak velocity and displacement will be reported. Previous work (Ostry etal., 1983) has shown that stress significantly affects the relationship between peak velocity and displacement. Preliminary results of the present study suggest that position-in-syllable also affects this relationship. The prosodic status of intervocalic stops preceding unstressed vowels (labeled as ambisyllabic by Kahn, 1976 and syllable-final by Selkirk, 1982) will also be investigated by comparing their kinematic characteristics to those of unambiguously syllable-initial and syllable-final stops.
Limitation of reflecting irregular bathymetry on matched-field processing
Authors: Diachok Orest , Smith James F.
Published: October 2005

Simulations of the potential of matched-field tomography for inverting ocean sound-speed structure (Tolstoy and Diachok, 1991) have to date neglected effects of bottom interacting modes. A study to determine the effect of experimentally obtained bathymetry in the thinly sedimented Pacific on matched-field processing has been conducted. The range- and depth-dependent sound speed in the water column was obtained from tomography data. The acoustic field used as synthetic data ‘‘measured’’ at the array was calculated by the parabolic equation program FEPE (Collins, 1991). The program was shown to be highly accurate (0.15 dB at 1000 km at 15 Hz) in matched-field studies against ‘‘exact’’ results produced by the normal modes program KRAKEN (Porter, 1991) in a range-independent environment. The effects of irregular bathymetry and shifts in bottom sound speeds were studied through simulations conducted at a frequency of 14 Hz at various ranges and bottom sound speeds. The results showed an increase in signal array degradation and reduced source localization. Effects of interaction with far and near sides of seamounts (relative to array position), and the overall trend of signal loss with range due to multiple interactions with the bottom will be discussed.
Nonlinear panel response and acoustic radiation from plane-wave excitation
Authors: Frendi A. , Maestrello L. , Bayliss A.
Published: October 2005

The generation and propagation of nonlinear sound waves radiated from a flexible panel excited by an acoustic source at a near resonant frequency were considered. The panel response is coupled to the radiation field of the surrounding fluid in a fully consistent manner. For each time step, the panel response is calculated by solving the nonlinear plate equation with the pressure difference across the panel as a source term. The pressure near the panel is computed by solving the nonlinear Euler equations with a boundary condition on the normal velocity obtained from the plate equation. The Euler equations also yield the near- and far-field radiation due to the panel vibration. The results show that nonlinearity tends to arise by formation of subharmonics as the amplitude of the excitation increases. There are small windows of excitation amplitude for which apparently chaotic behavior is obtained. The far-field directivity pattern is highly directional with a marked reduction in acoustic energy emitted normal to the pattern. The relative high-frequency content of the radiated pressure increases as the far field is approached. As the amplitude of excitation is increased, the effect of the coupling to the acoustic fluid is to delay the development of nonlinearities on the structure.
Acoustic monitoring of metallic beams by evaluation of resonant frequency decays
Authors: Robin Pierre Y.
Published: October 2005

The resonant frequencies of intact metallic beams, impacted with a hammer, were observed to decay at different rates than for slotted beams. A new technique, based on time-dependent spectral analysis of acoustic vibrations, has been developed to measure these decays. The method involves computing the Fourier transform from short-time windows of 25 ms, passed over 0.5 s of recorded vibrations. As a result, the amplitude of the beam’s resonant frequencies can be observed as a function of time. In particular, the amplitude, on a logarithmic scale, is shown to decay linearly with time, so that the resonant frequency decay rates can be quantified in terms of the slope of a linear regression. An experimental study was conducted with 36-in.-long steel beams with a rectangular cross section of 0.75×2.0 in. A slot, approximating a real crack, was gradually cut to 1.25 in., in the center of one beam, so as to evaluate the sensitivity of the decay rates to the crack size. The results show that these decay rates are discriminative features of the beam integrity and can be used in a pattern recognition scheme classifying intact and defective beams. [Work supported by NSF Grant No. MSS-9024224.]
Acoustic monitoring of beams by wavelet analysis
Authors: Akerberg Peeter M. , Jansen Ben H. , Robin Pierre Y. , Finch Robert D.
Published: October 2005

This talk proposes to use the discrete wavelet transform (DWT) of the acoustic signal produced by a hammer impact, to examine the integrity of metallic beams. The DWT is naturally adapted to transient signals and provides better frequency resolution for low frequencies than Fourier transform-based methods. Time-scale representations, such as the discrete wavelet transform, are related to time-frequency representations, which provide a framework to study variations of the signal frequency content over time. It has been found that the rate at which the amplitudes of the resonant frequencies decay is an important indicator of the integrity of metallic beams [Robin etal., J. Acoust. Soc. Am. 91, 2359(A) (1992)]. The frequency-dependent decays have been investigated more closely with the discrete wavelet transform. The decay appears to ondulate, and the frequency of this ondulation is related to the integrity of the beam. DWTs of actual vibration signals will be presented to show that the ondulation in the decay rates of the resonant frequencies can be used to differentiate between intact and slotted beams. [Work supported by NSF Grant No. MSS-9024224.]
The dynamic behavior of a reinforced concrete beam with or without cracks
Authors: Wang Zhijing , Finch Robert D.
Published: October 2005

The vibration behavior of a reinforced concrete beam with classical end conditions, with or without cracks, is being quantitatively investigated. The experimental results are expected to show that the natural frequencies of transverse vibration of the concrete beam will be changed when cracks are present. Other questions of interest are the effects of large numbers of cracks and their sizes. A comparison with the theory by Branson [DeformationofConcreteStructures (McGraw-Hill, New York, 1977)] will be made. The direct and indirect effects of loading on the beam are important. There are potential applications for this new vibration technique in nondestructive evaluation of reinforced concrete structures such as buildings, piles, and highway bridges. [Work supported by NSF Grant No. MSS-9024224.]
The dynamic behavior of a three-member plane truss with and without cracks
Authors: Pandya Shail R.
Published: October 2005

The response of a three-member plane truss with and without cracks to impact vibration is being experimentally investigated. The impact location is varied and the results are being analyzed. It is expected that the presence of a crack would affect the response pattern. The crack will be introduced in that member which has the maximum probability of being cracked in a truss under externally loaded conditions. The crack will then be modeled and the vibration response predicted using finite element analysis. The results obtained from this analysis will then be compared with the experimental results. The purpose of this work is to understand the response pattern for a triangular truss which could be considered as the building block for more complicated trusses found in bridges and other structures. [Work supported by NSF Grant No. MSS-9024224.]
The role of stress in word segmentation
Authors: Yerkey Paul N. , Sawusch James R.
Published: October 2005

Research reported at a previous meeting of the Acoustical Society examined the role of strong syllables in segmentation [P. N. Yerkey and J. R. Sawusch, J. Acoust. Soc. Am. 91, 2338(A) (1992)]. Subjects were asked to monitor bisyllabic nonwords for the presence of a word. The stimuli initially contained two stressed syllables, and were later edited to reduce the amplitude and duration of the second syllable. The results showed no effect of vowel quality on segmentation. This suggests that the vowel was not influencing segmentation, but that stress alone served as the primary cue for segmentation. In the present studies, the experimental items were recorded with naturally stressed and unstressed second syllables. In different experiments, the stimuli were presented to listeners with their natural stress patterns or after alteration of the amplitude and duration of the second syllable. The pattern of results across variations in second syllable stress and vowel quality (tense, lax, or neutral) will be discussed with respect to its influence on perception and implications for models of word segmentation. [Work supported by NIDCD Grant No. DC 00219 to SUNY at Buffalo.]
How do two speakers’ speech rhythms interact?
Authors: Kakita Kuniko
Published: October 2005

The present study investigated how one speaker’s speech rhythm affected another speaker’s speech rhythm. Specifically, it examined the changes in speech rate and intersentence interval duration to see how two speakers’ speech rhythms interacted. A number of series of five short sentences were read by two speakers in three different ways. In one instance, a series of five sentences were read through by a single speaker; in other instances, the second speaker took over from the first speaker (a) after the first sentence or (b) after the third sentence. The speech rate of the sentences and the duration of the intervals immediately after the takeover were compared with those at comparable locations in single-speaker readings. As concerns speech rate, one speaker adjusted it in accordance with the preceding speaker’s speech rate, while the other speaker did not. As concerns interval duration, both speakers showed a significant amount of adjustment, but in two contrasting ways of assimilating and dissimilating to the preceding speaker’s speech rhythm. The results for additional speakers will also be discussed.
Early pitch accent placement within words in American English
Authors: Shattuck-Hufnagel Stefanie
Published: October 2005

Some words of English can bear their major prominence on a strong syllable earlier than the main-stress syllable; this ‘‘stress shift’’ phenomenon has recently been discussed in terms of early placement of phrase-level prominences, or pitch accents. Factors that might cause speakers to place a pitch accent early in a target word include (1) structural position (accent the first accentable syllable in a new prosodic phrase), (2) rhythmic regularity (avoid placing accents too close together or in irregular patterns), and (3) one or more earlier appearances of the same word within the discourse. In a preliminary experiment designed to establish whether these factors influence pitch accent placement, two FM radio announcers read stories constructed with a ‘‘shiftable’’ target word or phrase (e.g., caffein, international, Cape Cod) in four different positions. Prominence locations were transcribed by listeners who were naive with respect to the predictions. Results showed more frequent placement of prominence on the main-stress syllable of target words and phrases when they occurred in phrase-final position (first and second mention), but on their earlier strong syllable when they occurred in initial position or in clash context (third and fourth mention). Preliminary acoustic analyses suggest that the labeled prominences correspond to substantial F0 markers, strengthening the hypothesis that at least some instances of perceived stress shift are cued by pitch accent patterns.
Assessing duration and variability in speech and nonspeech tasks performed by adults
Authors: Smith Bruce L.
Published: October 2005

Research with neuromotorically impaired adults and with normally developing children has suggested that duration and variability measures of speech may be indicators of neurological integrity. It is also important to determine whether these same measures have any implications regarding the speech production abilities of normal adults. The present study assessed the performance of normal adult subjects for several speech and nonspeech tasks to determine (a) whether normal adults can voluntarily reduce variability in their speech and (b) whether similar duration and variability relationships exist for speech and nonspeech (tapping and reaction time) tasks. The speech tasks included a control condition in which subjects repeated stimuli with no specific instructions, and an experimental task in which they were to be as consistent as possible in repeating stimuli in an attempt to minimize variability. Preliminary analyses indicate that the variability of speech segments did not tend to differ when comparing the control and experimental conditions. Relationships between duration and variability for the speech tasks and for the nonspeech tasks were quite comparable, with the correlations between these two measures not being very strong.
Haptic illusions in human perception of ‘‘virtual objects.’’
Authors: Hogan Neville , Fasse Ernest D.
Published: October 2005

Humans are subject to haptic illusions (distortions of their haptic perception of object properties such as length, force, or stiffness) just as they are subject to visual illusions. Understanding haptic illusions is particularly important for implementation and use of virtual environment technologies. Haptic illusions may also reveal fundamental properties of spatio-motor processing in the brain, in particular the geometric foundations of haptic perception and its relation to movement production. If a consistent Riemannian metric underlies biological representations of object geometry, there is a precise mathematical relation between the distortion of the different object features (e.g., length, force, stiffness). Experiments measuring human psychophysical functions for discriminating features of ‘‘virtual objects’’ (lengths of sides, angles of corners, stiffnesses, forces) will be reviewed. They show that no single consistent metric underlies this sensory-motor behavior. A distorted perception of the environment would be expected to influence motor behavior. Further experiments will be reviewed which show that certain errors in human movement production can be explained by perceptual distortion. Implications for virtual environment and teleoperation systems will be discussed. [Work supported by NIH and the Fairchild Foundation.]
Speech intelligibility, localization, and binaural detection with monaural and binaural amplification
Authors: Koehnke Janet , Besing Joan , Goulet Christine , Allard Marla , Zurek Patrick M.
Published: October 2005

As part of a continuing study of monaural and binaural amplification, performance on binaural detection, speech intelligibility, and localization was measured for four listeners with bilateral, sensorineural hearing losses, and one listener with a unilateral sensorineural loss. Masking-level differences (MLDs) were measured using speech-spectrum noise maskers and octave-band speech-spectrum noise targets. Localization was measured in anechoic and reverberant environments using three-word phrases at nine locations in the horizontal plane. Intelligibility of monosyllabic words was assessed in quiet and in three signal and noise configurations (0/0, 0/+90, 0/−90) in anechoic and reverberant environments. Performance on all three tests varies among hearing-impaired listeners, even those with similar hearing losses, and is generally poorer than normal-hearing listeners. In most cases, subjects with bilateral losses have better speech intelligibility and localization and larger MLDs with binaural amplification than monaural amplification. Listening environment does not appear to affect localization or speech intelligibility in quiet, but speech intelligibility in noise is usually better in the anechoic environment. Combined data for all the subjects shows a tendency for the S/N ratios for 50% speech intelligibility and the rms localization errors to decrease as MLDs increase. [Work supported by NIH Grant No. DC00428.]
Normal hearing levels for noise-reducing earphone types
Authors: Frank Tom , Magistro Donna M.
Published: October 2005

This study determined normal hearing levels (HLs) and intrasubject threshold reliability for supra-aural earphones housed in noise-reducing enclosures. Monaural thresholds were obtained for 25 normally hearing adults from 500 to 6000 Hz over four trials separated by 1 but no more than 7 days using a typical supra-aural (TDH-49P) and four supra-aural (TDH-39 or 49P) earphones housed in an Auraldome, Audiomate, Madsen, and Audiocup noise-reducing enclosure. The output of the supra-aural and each supra-aural earphone normally housed in a noise-reducing enclosure was measured on a NBS 9A coupler so that the thresholds could be referenced to normal threshold sound-pressure levels (SPLs) re: ANSI S3.6-1989. Within each trial the thresholds approximated 0 dB HL for each earphone type at each frequency except for the Audiomate where thresholds were elevated by as much as 28 dB at 500 and 10 dB at 1000 Hz. Further, the Audiocup and Auraldome thresholds were consistently lower than the typical supra-aural earphone. Between trials the thresholds were consistent (reliable) for each earphone type. Implications regarding reference threshold SPLs for supra-aural earphones housed in noise-reducing enclosures, threshold reliability, and use in industrial hearing testing programs will be discussed.
Auditory looming perception: Influences on anticipatory judgments
Authors: Wuestefeld A. Paige , Rosenblum Lawrence D. , Saldaña Helena M.
Published: October 2005

Research in visual perception suggests that observers can judge the time of arrival of a looming object based on earlier portions of the visible trajectory. A few studies in the auditory perception literature hint that listeners might also be able to anticipate the time of arrival of an approaching sound source [e.g., L. D. Rosenblum, C. Carello, and R. E. Pastore, Perception 16, 175–186 (1987)]. In a series of experiments, listeners judged the time of arrival of a looming car based on various portions of its acoustic signal. Subjects were asked to press a computer key when they felt the car would just pass them assuming a constant approach velocity. A number of variables were tested including (a) the time between the offset of the signal and the (virtual) time of passage, and (b) feedback concerning accuracy. Results suggest that increasing the time between signal offset and time of passage decreases judgment accuracy while providing subjects with feedback significantly improves performance. Preliminary analyses of the looming car signal were conducted to help determine the acoustic basis for anticipatory judgments.
The effect of vibrotactile noise on the growth of sensation magnitude and on the intensity DL
Authors: Gescheider George A. , Bolanowski Stanley J. , Mascia Christopher C. , Hall Karen L. , Verrillo Ronald T.
Published: October 2005

In this study, the hypothesis that the DL for vibrotactile amplitude discrimination is independent of the slope of the sensation magnitude function was tested. The slope of the sensation magnitude function was varied by requiring subjects to make amplitude discrimination judgments in the presence of or in the absence of vibrotactile noise. The slopes of the sensation magnitude functions were determined through a matching technique in which the subject adjusted stimulus amplitudes of a 250-Hz stimulus presented alone and a 250-Hz stimulus presented with a masking noise so that their sensation magnitudes were equated. The slope of the matching function was found to increase as a function of the intensity of the masking stimulus. In the second phase of the experiment, the intensity DL was measured by the gated-pedestal method for test stimuli presented under the stimulus conditions used in the matching procedure. For all but the weakest stimuli, the DL was found to be independent of the masking condition. This finding supports the hypothesis that the DL is the same when sensation magnitude is held constant and is independent of the slope of the sensation magnitude function. [Work supported by P0 1 DC-00380.]
Neural mechanisms of tactile sensation: Implications for haptic interfaces
Authors: Johnson Kenneth O. , Hsiao Steven S.
Published: October 2005

Data from a variety of sources suggest complementary roles for the SAI, RA, and PC afferent systems. The following is a working hypotheses: The SAI system is the primary spatial system and is responsible for tactual form and rough perception when the fingers contact a surface directly and for the perception of external events through the distribution of forces across the skin surface. The PC system is responsible for the perception of external events that are manifested through transmitted high-frequency vibrations. The RA system has a lower spatial acuity than the SAI system but a higher sensitivity to local vibration and is responsible for the detection and representation of localized movement between skin and a surface as well as for surface form and texture when surface variation is too small to engage the SAI system. The evidence for these hypotheses will be reviewed. The issue for haptic interfaces is how to drive these systems in order to simulate real contact between the hand and objects in the environment. That will be the subject of the remainder of the talk [Johnson and Hsiao, Ann. Rev. Neurosci. 15, 227–250 (1992)]. [Work supported by NIH.]
Kinesthetic acuity in humans with an emphasis on the fingers
Authors: Clark Francis J.
Published: October 2005

A signal detection model of static-position and movement mechanisms involved in detecting angular displacements of joints will be presented along with experimental data. The model predicts that separate neural mechanisms exist for position and movement and the data indicate that the static-position mechanism is absent at the interphalangeal joints of the fingers. The issue of what consequences result from a lack of a static-position sense will then be explored with tests of position matching accuracy comparing joints with and without this sense. A metric for kinesthetic acuity using information transfer analysis was used from which an estimate of channel capacity could be made. These data indicate surprisingly poor kinesthetic acuity for the fingers with channel capacities from ca. 1.4 bits or 2.5 discriminable levels over a 90-deg range for the proximal interphalangeal joint of the index finger (no static-position sense) to 1.8 bits of 3.5 levels for the metacarpophalangeal joint (static-position sense present). A possible anatomical basis for this lack of a static-position sense at interphalangeal joints will be discussed along with speculations on why the fingers, where one might expect a very high kinesthetic acuity, have such a low acuity.
High-frequency elastodynamic boundary integral inversion using asymptotic phase transformation
Authors: Roberts Ronald A.
Published: October 2005

Use of conventional finite element or boundary element computational methods for problems of time-harmonic elastic wave transmission, diffraction, and surface wave generation at a nonplanar fluid–solid interface become inefficient at high frequencies due to the fineness of discretization required for numerical convergence. A more efficient computational procedure has been developed that exploits the field phase as predicted by high-frequency asymptotic analysis (i.e., the geometrical theory of diffraction, or GTD). In this method, the boundary integral equation which governs elastic wave transmission is transformed by assuming a solution in the form of the GTD ansatz A(x)exp[iωp(x)], where p(x) is the field phase, ω is time harmonic frequency, and A(x) is the field amplitude. The phase p(x) is prescribed analytically through solution of the leading term eikonal equations of GTD. The transformed boundary integral equation is then solved numerically for the amplitudes A(x). The amplitudes A(x) will have a narrow spatial frequency spectral bandwidth, and hence can be efficiently calculated. This approach exploits the strengths of the GTD eikonal equation solutions while avoiding the analytical pitfalls encountered in the GTD transport equations.
The Westinghouse automated acoustic test facility
Authors: James Leslie M. , Geil Fred G.
Published: October 2005

The underwater acoustic test facility at Westinghouse contains a weather-independent 50-ft-diam, 30-ft-deep redwood tank decoupled from the ground and supported on vibration isolators. Two independent test systems with separate water accesses can accommodate transducers up to 3 tons. The primary test system consists of a Digital MINC/DECLAB-23 processor interfacing to a Hewlett Packard frequency synthesizer and network analyzer. A newer test system consists of a Real Time Systems (RTS)/HP Series 9000 computer-based data acquisition system that has improved low frequency and data storage capabilities; emulating the Naval Underwater Warfare Center system in Newport, RI, but with special application software. The facility is calibrated for pulsed measurements from 200 Hz to 2 MHz. There are several power amplifiers available including an Instruments, Inc. 16.5-kVA amplifier. There are custom test fixtures for torpedo noses, imaging scanners, and other applications as needed, including an 18-in.-diam pressure vessel with an acoustically transparent dome capable of static pressures of up to 120 psi used for high power testing. Scientific Atlanta rotators with digital synchroposition indicators enable accurate plotting of narrow beam patterns and patterns with deep nulls. The work currently in progress is the addition of low-frequency acoustic intensity that will be used for radiated noise measurements.
Acoustic acquisition, test, and analysis Naval Undersea Warfare Center Division, Keyport, Washington
Authors: Hardin Walt
Published: October 2005

Started in 1909 with the purchase of land on the Kitsap Peninsula in Puget Sound, the Naval Undersea Warfare Center Division Keyport [NUWCKPT] has become an important link in the development, maintenance, and improvement of underwater vehicles and devices for the U.S. Fleet. With the introduction of smart devices entering the U.S. Navy Fleet, acoustics testing became a paramount requirement. The first acoustic test facility at NUWCKPT was opened in 1952. As the field of underwater acoustics and devices has expanded so with it have expanded the scope and capabilities of NUWCKPT. This paper presents acoustic facilities and capabilities in the Acoustic Acquisition and Analysis Division, of NUWCKPT. Specific areas will be facility characteristics, data collection systems, types of test, problems, and needs.
NSWCDDWODET underwater acoustic measurements facility
Authors: Phelps W. B.
Published: October 2005

The Naval Surface Warfare Center, Dahlgren Division, White Oak Detachment has an underwater acoustic measurements facility consisting of two sites, the White Oak site and Brighton Dam site. The White Oak site consists of a redwood tank which is 30 ft in diameter and 20 ft deep inclosed in a building. The tank is below grade and mechanically isolated from all structures and machinery. The water is constantly filtered and circulated to provide mixing at constant temperature of 72 °F. The Brighton Dam site is located on Brighton Dam Road, 14 mi north of the NSWC White Oak laboratory. The test barge is anchored and moored 200 ft offshore in approximately 55 ft of water on Triadelphia Reservoir about 1200 ft from Brighton Dam. The reservoir, impounded by Brighton Dam, forms a lake of 800 surface acres containing approximately 7 billion gallons of water. A full range of acoustic measurements are possible at ambient temperature and pressure.
Fundamental frequency determination of speech and singing voice−A review
Authors: Hess Wolfgang J.
Published: October 2005

Fundamental frequency determination methods and algorithms (PDAs) can be grouped into two major classes: time-domain PDAs and short-term analysis PDAs. The short-term analysis PDAs leave the signal domain by a short-term transformation. They supply a sequence of average fundamental frequency estimates from consecutive frames. The individual algorithm is characterized by the short-term transform it applies: autocorrelation, distance functions (average magnitude difference function), least-squares method, harmonic analysis, and multiple spectral transform (cepstrum) are the most common algorithms. The time-domain methods, on the other hand, track the signal period by period. Extraction and isolation of the fundamental harmonic, and investigation of the temporal signal structure are the two extremes between which most of these PDAs are found. Special attention is given to the time-variant aspect of frequency determination and to postprocessing methods that allow fundamental frequency tracking over time. Among these, simple smoothing and list correction procedures are found that remove isolated errors as well as sophisticated schemes that apply dynamic optimization or stochastic modeling.
Fundamental frequency tracking in the log frequency domain based on a pattern recognition algorithm
Authors: Brown Judith C. , Puckette Miller S.
Published: October 2005

When the Fourier components of a sound with harmonic frequency components are plotted against log frequency, a constant pattern independent of fundamental frequency is obtained. A computationally efficient implementation [J. C. Brown and M. S. Puckette, ‘‘An efficient algorithm for the calculation of a constant Q transform,’’ J. Acoust. Soc. Am. (to be published 1992)] will be described. One can then calculate the cross-correlation function of these spectra with the ideal pattern, which consists of ones at the positions of the harmonic frequency components, to obtain the fundamental frequency of the input signal. Extremely accurate values of the fundamental frequency can be obtained using a method based on phase changes of the discrete Fourier transform [J. C. Brown and M. S. Puckette, ‘‘A high resolution fundamental frequency determination based on phase changes of the Fourier transform,’’ submitted to J. Acoust. Soc. Am. (1992)]. Results for a variety of musical instruments will be presented, including recent results on tracking very low frequencies produced by a cello.
An automatic musical frequency detection and MIDI control system for brass instruments
Authors: Cook Perry R. , Morrill Dexter , Smith Julius O.
Published: October 2005

Fundamental frequency detection algorithms optimized for use with instruments in the brass instrument family are presented. A new frequency tracking scheme based on adaptive periodic prediction is presented, and it is shown that this algorithm is equivalent to a high-precision adaptive comb filter. Frequency detection schemes that do not take into account the unique spectral and acoustical properties of a particular instrument family usually generate errors of three types: (1) octave and harmonic detection errors, (2) half-step errors that jitter rapidly about the true estimate, and (3) latency of detection. A frequency detection and live MIDI control system has been constructed for the trumpet, which significantly reduces detection errors and latency. By limiting the search range to the natural playing range of the trumpet, sampling rate and computation can be optimized, reducing latency in the estimates. By measuring and utilizing valve position in the frequency detection algorithm, the half-step jitter problem is completely eliminated, and latency can be further reduced. Schemes for reducing harmonic detection errors will be presented. The trumpet system as implemented will be quantitatively compared to two popularly available frequency to MIDI systems. A short demonstration of the trumpet performance system will be presented.
A digital phase/time method for fundamental frequency tracking
Authors: Mellen R. H.
Published: October 2005

Phase/time processing [S. O. Rice, SelectedPapersonNoiseandStochasticProcesses (Dover, New York, 1954)] has long been in general use; however, it seems to have found little or no application in musical acoustics. A recent digital analysis of violin tones indicates potential for frequency tracking. The method used here is: (1) FFT the real signal S0(t) to obtain the complex spectrum G0(F); (2) Gaussian filter (Q≊1.5) G0(F) to pass the fundamental in the desired frequency range and to suppress harmonics and other unwanted components; (3) FFTinv the bandpassed G(F) to obtain the analytic signal Z(t)=S(t)+iSH(t) (SH=Hilbert transform); (4) calculate Z(t)Z*(t+Δt)=U+iV; (5) calculate phase frequency F(t+Δt/2)=ATN[V/U]/2πΔt. Plots of phase frequency for synthetic signals with rapidly varying frequency and transients and for recorded violin tones are presented. Although details of the violin tones are partially masked by jitter in playback to the digitizer, this and other artifacts observed can serve to demonstrate some of the limitations of the method.
Spherical wave propagation in layered poro-elastic and fluid systems
Authors: Attenborough Keith , Taherzadeh Shahram
Published: October 2005

A numerical procedure for computing sound propagation from a point source in an environment consisting of a stratified fluid half-space above a layered poro-elastic ground is described. The wave amplitudes are computed from depth-separated wave equation by forming a system of linear equations arising from the boundary conditions at each and every interface between layers and solving the resulting global matrix. The inverse Hankel transform is then performed by fast Fourier method (FFP) to obtain the pressure and particle velocity at any elevation as a function of range. Wave propagation within the porous elastic layers is calculated according to the modified Biot–Stoll theory which predicts two compressional and one shear wave types in the medium. Six boundary conditions are therefore required at each ground intefaces and four at the fluid–ground interface, while only two are needed at the fluid–fluid interfaces. Using this code the role of the ground elasticity on above-ground propagation and in acoustically induced seismic excitation are explored. It is predicted that on certain ground types, characterized by a thin layer with small compressional and shear velocities, and at frequencies that coincide with a peak in the seimic-to-acoustic coupling ratio spectrum there is an extra sound attenuation.
Application of numerical grid generation to sound propagation over irregular terrain
Authors: Di Xiao , Gilbert Kenneth E. , You Chulsoo
Published: October 2005

A two-dimensional numerical grid generating technique is presented for constructing boundary-fitted coordinates for irregular terrain. In the resulting curvilinear coordinate system, the parabolic equation method is used to compute sound propagation over the terrain. With the proposed numerical grid generating scheme, two separate generating systems are used. One is an orthogonal coordinate generating system, and the other is an elliptical generating system. Used separately, the systems give, respectively, either an orthogonal mapping or a ‘‘smooth’’ (equal scaling factors) mapping. By using the two generating systems in conjunction with weighting factors from a variational method, one can control the orthogonality and smoothness of the grid. For certain simple cases, a conformal mapping can be generated by carefully choosing the weighting factors. Preliminary results using the method are compared to exact analytic solutions for a simple test case and, in more complicated cases, to numerical solutions computed by a different method. In principle, the extension of the two-dimensional method to three dimensions is straightforward. The extension to three dimensions in practice and the application to propagation problems with realistic three-dimensional terrain will be briefly discussed.