bluuebunny/crossref_metadata_2025 huggingface dataset:
Finite element analysis of a large vibrating space structure
Authors: Sridhara B. S. , Crocker Malcolm J.
Published: October 2005

The large flexible space structure to be used in the Strategic Defense Initiative Project is modeled as a long flexible beam with three point masses. Using the Galerkin method, finite element equations are formulated that take advantage of the fact that the natural boundary conditions come out as a result of integration by parts. Hermite polynomials have been chosen for the shape or interpolation function. A tubular beam made of graphite epoxy material, 100 m in length with 2.0-m o.d. and 1.95-m i.d. is considered for the purpose of this analysis. Masses of 500, 10 000, and 1000 kg are placed at the left extreme end, at a distance of 20 m from the left extreme end, and at the right extreme end of the beam, respectively. Natural frequencies of the long flexible beam are calculated using standard methods and the mode shapes are also plotted. Results are also obtained for a different location and increased values of the largest mass. Computer codes are being prepared to obtain the dynamic and steady-state response of the beam subjected to an impulse and a sinusoidal force.
Influence of various parameters on the transmission of vibrational power
Authors: Gilbert T. , Cuschieri J. M. , McCollum M. , Rassineux J. L.
Published: October 2005

The transmission of vibrational power between two thin plates in an L-shape configuration is investigated using an SEA model. Only bending waves are considered in the model. Expressions are developed for both the ratio of energy levels in the two plates and the ratio of the transmitted power to the input power. It is found that these ratios are only dependent upon three parameters: frequency, dampings of the plates, and coupling loss factors between the two plates. Therefore, a way to decrease both the transmitted power and the energy level in the receiving plate over a wide range of frequencies is to increase the damping of the source plate. The coupling loss factor is influenced by the physical characteristics of the plates, such as area, thickness, and material. The effect of these parameters on the coupling loss factor and on the power and the energy ratios is found to be small. Thus the most efficient way to control the transmission of power and the energy levels is to increase the damping of the plates. This result is in agreement with other results obtained using the energy influence coefficient method. [Work supported by NASA.]
Power flow method for an L-shaped plate
Authors: Rassineux J. L. , Cuschieri J. M. , McCollum M. , Gilbert T.
Published: October 2005

The power flow method is extended to the analysis of the vibrational power flow for two-dimensional L-shaped plate structures. The analysis is for a plate that is assumed to be simply supported along the edges and pinned at the junction. Structural damping is introduced by means of a complex elasticity modulus. The transmitted power from one plate to the other is computed using input and transfer generalized mobilities. The mobility functions are obtained from the solution for the response of a flat plate using either point or edge moment excitation. The analytical solution obtained enables computation of the response for a broadband excitation, regardless of the range of frequency considered. Different locations of the excitation point and different values of damping and their influence on the response are analyzed. As a verification, the obtained results are compared with finite element analysis and statistical element analysis. [Work supported by NASA.]
Sonar project laboratory for undergraduates
Authors: Korman Murray S.
Published: October 2005

A portable microcomputer workstation for supporting basic acoustic experiments has been developed over the past three years in the Physics Department at the U.S.N.A. Students enrolled in underwater acoustics and sonar (a 3-h lecture course without a laboratory) perform fundamental acoustics experiments in the classroom without having to utilize large numbers of analog electrical boxes. The heart of the workstation (based on the 6502 microprocessor) includes a low-cost, dual-channel, 8-bit analog-to-digital board. This workstation features menued software that can capture a waveform and display its trace. Simple mathematical operations include squaring the waveform, integrating, differentiating, finding the rms value, and determining the probability density function. A 1024-point FFT program is also included in the menu. Students are not required to know computer languages to operate the workstation and no experiment is automated or simulated. During nonclass hours, students design an experimental project that might include “model” measurements of target strength, beam pattern functions, reflection loss, sound velocity, vibration, and studies of signals in noise. The workstations's versatility will be demonstrated.
Vector quantization of LPC parameters based on dynamical features of hearing
Authors: Ono Shigeru , Araseki Takashi
Published: October 2005

New code-word search criteria for vector quantization of LPC parameters are proposed. Various vector quantization schemes for LPC parameters have been proposed previously. In these schemes, an input speech waveform is partitioned into successive short time frames and code-word search is carried out under the criteria based on instantaneous speech features. Characteristics of quantization noise vary widely from frame to frame, independent of the dynamic characteristics of the input speech. At very low bit rates, where it is difficult to keep the quantization distortion at a low level, variation in the quantization noise significantly degrades the quality of the reconstructred speech. “Temporal masking” experiments indicate that there is a postive correlation beteen the masking value and the speech spectral transition rate. These experiments also suggest that spectral transition rate distortion is emphatically perceived. In this paper, new criteria, based on these temporal masking properties, are proposed. They would minimize not only the instantaneous spectral distortion, but also the spectral transition rate distortion. Using these criteria, vector quantization of the LPC parameters is developed. Subjective experiments confirm that these criteria are superior to those based only on instantaneous speech features.
LPC voicing periodicity correction
Authors: Milenkovic Paul , Sekimoto Sotaro
Published: October 2005

The autocorrelation method of linear predictive coding (LPC) analysis matches the first p samples of the autocorrelation function of the impulse response of an all-pole filter to the autocorrelation function computed from a speech waveform. The autocorrelation function of the speech waveform is influenced by the periodicity of the voice source, as well as by the waveform window. This influence, when incorporated into the autocorrelation function of the all-pole filter, results in errors in formant frequency and bandwidth values that become important at the fundamental frequencies typical of female speech. The use of an analysis-by-synthesis technique is proposed, where the autocorrelation function of the output waveform of an LPC synthesizer is matched to the speech data autocorrelation function. The all-pole filter determined by LPC analysis provides a match when the synthesizer is operated at an extremely low fundamental frequency. The synthesizer fundamental frequency is increased in small steps and corrections are made to the all-pole filter at each step to preserve the autocorrelation function match. This process continues until the correct value of the fundamental frequency is reached. [Work supported by NIH Grant NS 21516-3.]
Evaluation of articulatory codebooks
Authors: Larar J. N. , Schroeter J. , Sondhi M. M.
Published: October 2005

The design of a codebook of articulatory shapes was recently reported [M. M. Sondhi, J. Schroeter, and J. N. Larar, J. Acoust. Soc. Am. Suppl. 1 81, S78 (1987)]. The codebook is used in an automatic articulatory analysis/synthesis scheme relying on good initial estimates of vocal tract cross-sectional areas to start up an optimization procedure [J. Schroeter, J. N. Larar, and M. M. Sondhi, Proc. IEEE ICASSP, 308–311 (1987)]. An application-oriented way to evaluate the codebook would be to determine how much the final optimized shape deviates from the shape initially selected. A more general evaluation is to compute the average distortion incurred in quantizing some input record of speech. Using this procedure with an appropriate distance measure, various aspects of codebook formation are evaluated. In particular, the focus is on the strategies utilized for generating training data. Also considered is the effect of different approaches to code-word formation once the training data have been obtained. Evaluation results for this work done to refine the codebook will be presented along with the performance as a function of codebook size.
Speech formant trajectory estimation using dynamic programming with modulated transition costs
Authors: Talkin David
Published: October 2005

A new algorithm to track automatically speech formant frequencies have been developed. Dynamic programming is used to optimize formant trajectory estimates by imposing appropriate frequency continuity constraints. The continuity constraints are modulated by a stationarity function. The formant frequencies are selected from candidates proposed by solving for the roots of the linear predictor polynomial computed periodically from the speech waveform. The local costs of all possible mappings of the complex roots to formant frequencies are computed at each frame based on the frequencies and bandwidths of the component formants for each mapping. The cost of connecting each of these mappings with each of the mappings in the previous frame is then minimized using a modified Viterbi algorithm. Two sentences spoken by 88 males and 43 females were analyzed. The first three formants were tracked correctly in all sonorant regions in over 80% of the sentences. These performance results are based on spectrographic analysis and informal listening to formant-synthesized speech.
Some properties of autoregressive model related cepstrum
Authors: Juang Biing-Hwang , Mansour David
Published: October 2005

Autoregressive model related cepstrum, or, briefly, LPC cepstrum, recently gained renewed attention because it was shown to be an effective representation in speech recognizer designs [B. H. Juang, J. G. Wilpon, and L. R. Rabiner, Proc. IEEE ICASSP-86, 765–768 (1986); B. Hanson and H. Wakita, Proc. IEEE ICASSP-86, 757–760 (1986); Y. Tohkura, Proc. IEEE, ICASSP-86, 761–764 (1986)]. In this paper, some properties of the LPC cepstrum are investigated that are of important consideration when the recognizer is to be deployed in situations where mismatch between the training conditions and the testing conditions may potentially occur. More specifically, the effects of LPC analysis order, additive noise, and pole movements upon some key characteristics of the LPC cepstrum are considered, such as the average and the norm of the cepstral vector. A simple relationship among the cepstral coefficients, the predictor coefficients, and the reflection coefficients is established, and unlike the Itakura-Saito distortions, it is shown that the L2 distortion between two cepstra obtained from different-order LPC models of the same speech data may contain an inherent nonzero bias. Also considered are the effects of additive noise, using an additive power spectrum model and a constrained ARMA model (sum of an all-pole model and a constant). It is shown that the norm of the LPC-cepstral vector shrinks as a results of noise contamination, compared to that of the clean speech. Further, moving the poles of an LPC-model spectrum is shown to be equivalent to multiplying a power series to the cepstrum and thus LPC-model bandwidth broadening and pole enhancement can be easily accomplished in the cepstral domain. Consequently, evaluation of some frequency-weighted distortion measures that are believed to be desirable under certain conditions becomes straightforward via cepstral processing.
Speech enhancement in crypto-bridge communication systems
Authors: Lin Daniel , Yacobi Yacov
Published: October 2005

In secure audio teleconferencing, a crypto-bridge system synchronously adds simultaneous encrypted signals, modulo a known number P, and broadcasts the result to the participants. Each terminal, using a secret key, then decrypts the modular sum of encrypted signals to obtain the desired ordinary sum of plain text signals. Such secrecy systems are provably secure, assuming the existence of one-way functions (e.g., DES) for the modular sum cipher [E. F. Brickell et al., CRYPTO-87]. In this paper, the implementation of additive crypro systems in a narrow-band (2B + D) ISDN environment are considered. The problem of noise reduction for 6-bit linear PCM speech coding (an essential constraint for bridging the encrypted signals in a single B-voice channel) is discussed. In particular, an enhancement procedure using a prefilter, “dither” noise and an adaptive postfilter is proposed. The system is based on an all-pole model for the degraded speech. The coefficients of the postfilter are easily derivable from the all-pole filter parameters. Informal listening tests have shown that the enhanced speech is near “toll” quality.
A new mathematical model for intonation: Historical motivation and application
Authors: Rekart Deborah , Drumright David
Published: October 2005

Early attempts to characterize English intonation were theoretical descriptions in terms of tones, tunes, and segmental pitch levels. Recent acoustic descriptions of intonation have involved the perceptual determination of F0 contours in terms of global declination lines or as breath groups with terminal and nonterminal slopes. Lieberman et al. (1985) showed that F0 contours could be described objectively by all-points linear regression lines, but this did not yield a compact characterization of the contours. Hirst (1983) developed a model based on nonlinear (parabolic) function, but it is not unified in that it resets parameters for syllables independently. Attempts to compare intonation contours in different languages and dialects have been based on the existing models. In the following abstract a mathematical model capable of compactly quantifying intonation contours using four parameters is proposed [Drumright and Rekart (1987)]. This paper demonstrates how the model can be applied to analyze foreign language, non-native, and dialectal differences in the intonation of statements and questions in spontaneous and read speech.
A new mathematical model for speech intonation: Theory and hardware/software realization
Authors: Drumright David G. , Rekart Deborah M.
Published: October 2005

A mathematical model based on combinations of sinusoids and exponential decay is shown to be superior in representing actual intonation contours. The model is generative rather than ad hoc, in that the configuration is controlled by four overall parameters. The model has been realized and tested in software form by a program that reads an actual intonation curve [produced by the program PIG and described in J. Acoust. Soc. Am. Suppl. 1 81, S78 (1987)] and sets the four parameters by an iterative matching process. The hardware realization, by a circuit involving coupled oscillators, invites comparison with analogous neural structures.
Acoustic sensor development at NRL
Authors: Dandridge A.
Published: October 2005

The development of fiber optic acoustic sensors at the Naval Research Laboratory will be reviewed. The talk will describe the fundamentals of the design and operation of fiber optic sensors. The major area of this talk will be the transduction mechanism of acoustic energy to optical phase shift. Methods to detect the phase shifts and to provide a linear voltage output will only be briefly discussed. The frequency regimes to be covered range from a few tens of Hz to MHz. In the low-frequency regime (up to several kHz), where the pressure is hydrostatic, the role of the material moduli of fiber coatings and bulk mandrels will be considered. A number of sensor designs for operation in the higher-frequency regimes, where inertial effects do not permit direct axial strain terms to play a major role, will also be described.
Multiplexing techniques for interferometric fiber-optic sensors
Authors: Kersey A. D. , Dandrige A.
Published: October 2005

Considerable research interest is presently being directed toward the development of all-fiber, multisensor networks for use in arrays and applications where a large number of different physical parameters are of interest, such as those encountered in the industrial process control area. A number of “all-optical” approaches to the multiplexing of such networks have been proposed in recent years. These techniques are most attractive for use in a wide range of application areas, as the sensors operate passively and can be remotely located from the source and detection electronics. Furthermore, the overall system comprising the sensors and fiber links retains a low susceptibility to electromagnetic interference. The principle of operation of multiplexing techniques based on time, frequency, and coherence-division addressing schemes will be described. The paper will compare experimentally reported performance features such as the multiplexing efficiency, sensitivity, crosstalk, etc., and will discuss practical limitations of the various methods.
Thermal noise limitations in u fiber optic seismic sensor
Authors: Garrett S. L. , Hofler T. , Gardner D. L.
Published: October 2005

Optical fibers are neither sensitive nor selective as a transduction medium. The high sensitivity of most fiber optic sensors has been a consequence of the fact that optical interferometric demodulation is capable of resolving phase changes of order 1 μrad/Hz. In a fiber sensor of 10-m length, this is equivalent to a resolution of one part in 1014 during a 1-s observation using light of 900-nm vacuum wavelength. For comparison, a good quality 1-in. condenser microphone can resolve a relative change in diaphram-to-backplate separation of one part in 108 during a 1-s observation. This detection advantage of 120 dB has obscured many “sins” in several fiber optic sensor designs. This presentation will concentrate on the sensitivity limitations imposed by the fluctuation-dissipation theorem [L. D. Landau and E. M. Lifshitz, Statistical Physics (Addison-Wesley, Reading, MA, 1969), 2nd ed., Chap XII] at nonzero temperatures and will demonstrate that the thermal limitations are entirely analogous to fluctuations observed in “mirrored galvanometers” that also use “optical leverage” to increase sensitivity. The results will be applied to the performance of a seismic sensor with a sensitivity that is limited by its intrinsic thermal noise. [Work supported by the Office of Naval Technology and the Office of Naval Research.]
A compact fiber optic laser probe to monitor surface vibration displacements
Authors: Jarzynski Jacek , Lee Dowon , Berthelot Yves H.
Published: October 2005

An heterodyne laser interferometer is used to monitor the phase modulation of the laser light scattered from a vibrating surface. A laser beam is split into two arms of an interferometer where light is guided through single mode optical fibers so as to preserve phase information. The reference and frequency-shifted signals are recombined in a multimode fiber tightly connected to a receiving photomultiplier. Miniature gradient refractive index (GRIN) lenses are used to provide maximum efficiency and to make the probe as compact as possible. Both normal and tangential components of the surface displacement can be determined with a probe illuminating the vibrating surface at normal incidence and at two mutually perpendicular off-normal directions. The tangential component of the surface displacement is found from speckle reflection of the nonperfectly flat vibrating surface. This compact probe can be positioned at some distance (10–50 cm) from the vibrating surface for truly nonintrusive measurements.
Discriminability, loudness, and masking in the rat: A confirmation and extension
Authors: Raslear Thomas G.
Published: October 2005

Rats discriminated between two sound-pressure levels (SPL) of a pure tone: standard (STD) SPLs of 84 and 74 dB, and comparison (CO) SPLs 4, 14, and 24 dB below STD were tested in quiet and 60-dB noise at 4 and 12.5 kHz (24 conditions). The decibel difference between STD and CO accounted for only 43.52% of the variance in the signal detection measure of sensitivity (d′) across conditions, whereas the loudness difference (LD = STD0.35 − CO0.35) accounted for 89.82% of the variance in d′. These results confirm and extend previous observations that: (1) equal decibel differences are not equally discriminable [R. Pierrel, J. G. Sherman, S. Blue, and F. Hegge, J. Exp. Anal. Behav. 13, 17–35 (1970)]; (2) loudness for the rat increases as a power function of SPL, with an exponent of 0.35 [R. Pierrel-Sorrentino and T. G. Raslear, J. Comp. Physiol. Psychol. 94, 757–766 (1980)]; (3) masked loudness is a linear function of loudness in quiet [T. G. Raslear, R. Pierrel-Sorrentino, and F. Rudnick, Behav. Neurosci. 97, 392–398 (1983)].
Evidence for an acoustical pathway to the inner ear through the lower jaw for an echolocating dolphin (Tursiops truncatus)
Authors: Brill Randall L. , Sevenich Martha L. , Sullivan Timothy J. , Sustman Janet D. , Witt Ronald E.
Published: October 2005

A dolphin (Tursiops truncatus) was conditioned to perform a discrimination task, by means of echolocation, in a “Go/No-go” paradigm while stationed in an underwater hoop and wearing suction cups over its eyes. To investigate the hypothesis that the fat-filled lower jaw of odontocete cetaceans provides an acoustical pathway to the inner ear, the dolphin was additionally required to perform the task while wearing either of two rubber hoods designed to cover its lower jaw. One hood, constructed from nonfoamed neoprene, allowed returning acoustic signals to pass, while the other hood, constructed from closed-cell neoprene, substantially attenuated such signals. The dolphin's performance was significantly hindered while wearing the attenuating hood (p < 0.001, X2) as opposed to its wearing the hood made of nonfoamed neoprene or no hood at all. Acoustical data tape recorded during the experiment indicate that, with few exceptions, the spectral peaks of the dolphin's outgoing echolocation signals averaged between 30 and 50 kHz. These results support the hypothesis and agree with previous electrophysiological data that suggest that the lower jaw is involved in the transmission of high-frequency signals to the inner ear.
Modulation and gap detection with filtered and broadband noise signals
Authors: Formby C. , Muir K. , Uranga J. , Lynn C. , Ahlers A.
Published: October 2005

The higher audiometric frequencies arguably are the most important for some tasks requiring fine temporal discriminations. Both modulation and gap detection thresholds improve with increasing center frequency of noise bands. For modulation detection, however, not all investigators are convinced that improved performance is necessarily dependent on sensitivity at the higher center frequencies. They point out that the high-frequency advantage for modulation detection may be a consequence of concomitant increases in stimulus bandwidth as a function of increasing center frequency of the noise band. To determine whether signal frequency or signal bandwidth is more important for modulation and gap detection, modulation and gap detection thresholds were measured for low-pass (LP) filtered noise (fc = 4000 Hz, bandwidth = 4000 Hz), high-pass (HP) filtered noise (fc = 4000 Hz, bandwidth = 2500 Hz), and broadband (BB) noise (bandwidth = 6500 Hz). Modulation and gap detection thresholds for the BB and HP signals were similar, and were better than thresholds for the LP signal. The ability to hear the higher frequencies of the noise signal apparently is more important for these tasks than is perception of signal bandwidth. [Research supported by Division of Sponsored Research, University of Florida, and American Cancer Society.]
Effect of click rate and delay on breakdown of the precedence effect
Authors: Clifton Rachel K. , Freyman Richard L.
Published: October 2005

When clicks are presented from two loudspeakers, with one output leading the other by a few ms, a listener localizes the sound solely at the leading loudspeaker, a phenomenon known as the precedence effect. However, the precedence effect can be disrupted for several seconds if the leading and lagging (echo) locations are quickly switched [Clifton, unpublished manuscript]. Click trains were presented from two loudspeakers, placed 90° right and left off midline in an anechoic chamber, with lagging click delays ranging from 2–9 ms. Click rates were 1, 2, 4, 6, and 8/s. Under some conditions, subjects reported hearing clicks from both loudspeakers immediately following the switch in leading and lagging outputs, indicating a breakdown of the precedence effect. The length of time both clicks were heard following the switch was affected by both echo delay and click rate. Inhibition of the precedence effect was strongest when rate of clicks was slower and delays were longer, that is, just under the subject's echo threshold. At shorter delays and faster rates the precedence effect was experienced throughout the trial.
Inhibitory mechanisms and cortical processing
Authors: Daly D. M. , Daly D. D. , Drane J. W. , Wada J. A.
Published: October 2005

When balance of inhibitions declines/fails, patients with seizures in auditory cortex report certain synthetic [be]-[de]-[ge] as nasals, “bleats,” and then undifferentiated buzzes. Effects of increased inhibition can appear with auditory cortex in volume surrounding disinhibited areas (even contralaterally): Patients report same range of stimuli as clearly identical [J. Neurophysiol. 44, 200–222 (1980)]. Here a case with seizures in visual cortex and post-ictally impaired auditory comprehension (but normal peripheral acuity), is reported. Testing included prerecorded sets of [be]-[de]-[ge], [ge]-[ye], and [be]-[we] presented through headphones monaurally and binaurally. During episodes with more frequent seizures, patient had intervals (some lasting nearly an hour) when [de] boundaries for [be]-[de] and [de]-[ge] increased approximately 200 Hz; [be]-[we] and [ge]-[ye] boundaries increased 40–50 ms (with stimuli less than 50 ms now reported as vocalic). The BDG stimuli cover tonotopically differing extents of cortex; BW or GY stimuli cover tonotopically limited extents at different rates. Results appear consistent with augmenting infield inhibition.
The effects of pedestal level on incremental auditory responses in humans: Loudness matching study
Authors: Riquimaroux Hiroshi , Dallos Peter
Published: October 2005

The effects of pedestal level (LB) on incremental auditory responses were investigated in normal-hearing human subjects with amplitude modulated (AM) white noise by means of a loudness matching procedure. The method of adjustment was used to find equally loud increments (ΔL) for various standards ΔLs and LBs (30–80 dB SPL). The LB for standard ΔL was kept constant at 30 dB SPL. The hypothesis to be tested was that loudness of ΔL is independent of LB. The results show that loudness of ΔL appears to be almost independent of LB for very small ΔL (⩽ 6 dB). The findings also indicate the possibility that the stimulus peak level (LP) might be involved in the determination of loudness of ΔL when ΔL is large. The data demonstrate that the slope of loudness growth function increases as LB increases. For low LBs, the slope has a tendency to asymptote to the reference line with the slope of one. The results agree with earlier JND data. It is concluded that the hypothesis appears to be supported by the present data for very small ΔLs.
Frequency specificity of the cochlear action potential varies with stimulus presentation rate
Authors: Salt Alex N. , Vora Arti R. , Thalmann Ruediger
Published: October 2005

It is well documented that the amplitude of an averaged cochlear action potential decreases as the rate of stimulus presentation is increased. In the present study, frequency regions contributing to the AP elicited by tone burst stimuli at rates from 4 to 80/s have been investigated in guinea pigs using a tone masking procedure. Derived responses corresponding to specific regions were obtained by subtracting AP responses in the presence of a low-level continuous masking tone from those in the absence of masker. This is a modification of the technique described by Hood et al. [J. Acoust. Soc. Am. Suppl. 1 81, S8 (1987)]. When stimulation rate is increased, the derived response amplitudes decreased near the probe frequency region, while amplitudes at outlying frequencies were less markedly changed. These data suggest the contribution to the AP from the region corresponding to the probe frequency becomes less at high stimulation rates. With an 80/s stimulation rate the contribution from this region is negligible. The degradation of frequency specificity of responses is presumed to arise from each stimulus acting as a “forward masker” for the following stimulus. To optimize the frequency specificity of AP responses to tone burst stimuli, the rate of presentation must therefore be considered. [Work supported by NIH.]
Ultrastructural correlates of hair cell motility
Authors: Evans B. N. , Yonovitz A.
Published: October 2005

Nonstimulated as well as outer hair cells showing mechanical displacements to electrical and chemical stimulation were examined ultrastructurally to assess the lability of a rhodamine-labeled actin structure extending between the cuticular plate and the peri-nuclear region. Hair cells of varying lengths (20–80 μm) were stimulated and immediately fixed following mechanical displacement. All cells revealed a distinct non-membrane bound complex in cross section, but were infrequently observed in longitudinal sections. The cross-sectional area of the structure increased in successive ascending sections. This structure spirals apically in a nondirect manner, but eventually consolidates into a primary mass to become contiguous with the inner leaflet of the cells plasmalemma. A localized microtubule network was found to pass most prominantly in the thin basal aspect of the complex. The cross-sectional orientation appears to optimize the visualization of this potentially important aspect of the outer hair cells anatomy.
Tests of a modified energy-detector model of frequency discrimination
Authors: Emmerich David S. , Ellermeier Wolfgang , Butensky Brenda
Published: October 2005

Henning [J. Acoust. Soc. Am. 42, 1325–1334 (1967)] proposed a modified energy-detector model of auditory discrimination and detection. This model, as well as several other models, predicts that random variations of the amplitudes of the signals to be discriminated in a frequency-discrimination experiment should make the task much harder. However, Henning [J. Acoust. Soc. Am. 39, 336–339 (1966)] reported that, for signal frequencies less than 4000 Hz, variations in amplitude did not lead to the expected impairment on the frequency-discrimination task. A reexamination of this question indicates that when the appropriate control comparisons are made, subjects do in fact show the predicted impairment. Nevertheless a second experiment suggests that this result may not provide a good test of Henning's (1967) energy detection model since variations in pitch with intensity, which are not considered in the model, are likely to be involved. For this reason, a third experiment was conducted in order to provide an independent test of the model. The results of this third experiment do not support the energy-detector model.
A simple procedure for measuring iso-loudness contours
Authors: Allen J. B. , Chien P. S. , Hall J. L.
Published: October 2005

A procedure for measuring iso-loudness contours in normal and hearing-impaired subjects similar to that used by Geller and Margolis has been devised [J. Speech Hear. Res. 27, 20–27 (1984)]. The stimuli are bursts of noise 12-oct wide, centered at 1-oct intervals from 250 Hz to to 4 kHz. The subject is asked to rate the stimuli according to a six-point scale, using the descriptive words “very soft,” “soft,” “OK,” “loud,” “very loud,” and “too loud.” The instructions include a one-sentence description of each loudness category. Failure to respond is recorded as “nothing.” In an initial pass, the lower and upper limits of the subject's hearing range at each frequency are estimated. These initial trials also serve to train the subject. Following the initial pass, 15 levels, logarithmically spaced between the measured limits at each frequency, are presented in random order, three times each. The entire procedure takes about 30 min to complete, and it provides stable estimates (based on test-retest measurements) of loudness growth functions. These loudness growth functions are then replotted as iso-loudness contours. Data from normal and hearing-impaired subjects will be presented along with their pure-tone thresholds.
Pitch and timbre of inharmonic tone complexes of Balinese “Gender Wayang” instruments
Authors: Deutsch Werner A. , Födermayr Franz
Published: October 2005

The tones of the scale of two Balinese “Gender Wayang” instruments have been stored on computer disk and are used as test tones in pitch detection experiments. The subjects were instructed to identify the pitch of the inharmonic instrument tones by tuning a sinusoidal test tone on the appropriate pitch height in a computer controlled up/down adaptive procedure. The experimental results support the general ambiguity of pitch perception in case of inharmonic tone complexes, though a certain preference of the subjects on the lowest partial could be observed. This perceptual ambiguity could be increased further when the first partial of the instrument tones was removed by means of narrow-band digital filtering. Based on frequency analysis data of the natural sounds, a synthesis of the figuration “melody” of the Gender Wayang piece “Lagudelem” has been performed in which the inharmonic complex tones have been replaced by harmonic ones. The comparison of all three, the original sound probes and the filtered and the synthesized tones shows the strong dependence of pitch on timbre. The results of the pitch detection experiments are discussed on the basis of critical band analysis and the current theory of pitch perception as published by E. Terhardt (1972).
Range-dependent cut-an frequencies in a three-dimensional ocean environment
Authors: Glegg Stewart A. L. , Yoon Jong R.
Published: October 2005

Sound propagation in a three-dimensional wedge-shaped ocean environment demonstrates significant ray curvature in a direction parallel to the wedge apex. The theoretical prediction of this shows that these effects are more significant at low frequencies than at high frequencies; consequently, the three-dimensional ocean acta as a range-dependent acoustic filter. This paper describes an experimental and theoretical investigation of pulse propagation in a model scale three-dimensional wedge-shaped ocean. In the experimental setup the complete acoustic field could be measured for a given source position and the wedge angle could be changed. The results show good agreement with the idealized modal solution of Buckingham, and demonstrate how the pulse is distorted as a function of range. They also demonstrate how the cut-on frequency for acoustic propagation increases as a function of range parallel to the shoreline.
Scattering in basins or horizontally refracting modes: Which is it?
Authors: Harrison C. H.
Published: October 2005

In a large basin there are at least two types of 3-D ray paths from a source to a distant receiver via the sloping basin edge (other than the direct path). One is the simple scattered ray at the seabed, and the other is the multiple reflected ray that steepens in shallow water but finally turns in the horizontal plane towards deep water (otherwise interpreted as a horizontally refracting vertical mode contribution [C. H. Harrison, J. Acoust. Soc. Am. 65, 56–61 (1979)]). If the total change in heading of the latter ray is 2θ, its elevation angle will never exceed θ [C. H. Harrison, J. Acoust. Soc. Am. 62, 1382–1388 (1977)], and the path may exist despite reflection losses in realistic environments. Therefore, if the top and bottom surfaces are perfectly smooth, there will be no scattering and there will certainly be multiple reflection. Conversely, if the surfaces are rough, the multiple reflection will be inhibited but there will definitely be a scattered return. Since there is scope for misinterpretation of experimental results, this paper proposes some experiments to distinguish the two paths.
Diffraction of sound by hard strips and truncated wedges
Authors: Tolstoy Ivan
Published: October 2005

Starting from rigorous diffraction solutions for single, hard, semi-infinite wedges, it has long been possible to construct approximate solutions for more complex angular surfaces [A.D. Pierce, J. Acoust. Soc. Am. 55, 941–955 (1974)]. Demonstrated here is an exact procedure that formulates the multiple scatter between vertices of an angular body via the standard self-consistent algorithm. This allows one to obtain the correct diffracted field of each vertex. Summing these together with the relevant incident and image fields leads to an exact solution for the full sound field. This is done here for the case of a plane harmonic incident sound field for (1) a truncated wedge and (2) a hard strip of width l. Numerical results for the latter model are shown to compare satisfactorily for all kl with experimental data [H. Medwin et al., J. Acoust. Soc. Am. 72, 1005–1013 (1982)]. This theoretical procedure yields, in the plane wave case, a formal estimate of the error incurred in the “double diffraction” approximation (i.e., the procedure that neglects all but the first two terms of the multiple scatter series). This error decreases like elkl/kl for increasing kl. It is then possible to confirm theoretically the observation of Medwin et al. who showed that, for the geometry of their strip experiment, this approximation is actually very good for kl≳ 3 and quite adequate in the region 1 < kl < 3, using the Biot-Tolstoy rigorous impulsive point source solution for the perfect wedge [M. A. Biot and I. Tolstoy, J. Acoust. Soc. Am. 29, 381–391 (1957)] for their double-diffraction solution. [Work supported by ONR.]
Sound interaction with a sloping, upward-refracting ocean bottom
Authors: Ali Hassan B. , Arvelo Juan I. , Nagl Anton , Überall Herbert
Published: October 2005

Results of a parobolic-equation (PE) code are shown for the problem of sound interaction with an upsloping, upward-refracting ocean bottom. They are compared with the results of a coupled-mode code [J. F. Miller et al., J. Acoust. Soc. Am. 79, 562 (1968)] for a range-dependent environment. This includes bottom penetration effects at mode cutoff and a study of how these are affected by the sound-speed gradient and by the absorption in the sediment. The bottom penetration features are further illuminated by employing Gaussian sound pulses and studying their propagation, illustrating the effects of bottom penetration and reradiation back up into the water (as caused by the upward-refracting bottom gradient), arriving ahead of the water-borne pulse.
Spherically symmetric acoustic propagation across a fluid/fluid boundary
Authors: Buckingham Michael J.
Published: October 2005

The acoustic field produced by a point source at the center of a fluid sphere, which itself is immersed in a fluid medium, is of interest in connection with ambient noise in the Arctic Ocean. A solution for the field in the internal and external fluid domains is presented based on novel finite Hankel transforms. The method involves internal and external Hankel transformations of the Helmholtz equation, which introduce explicitly the boundary values of the field and its spatial derivative at the spherical interface between the two fluids. By combining the resultant equations with the boundary conditions (continuity of pressure and normal component of velocity) all the unknown constants are determined, and on performing the inverse Hankel transforms, the final expression for the field is obtained. The mathematical formalism of the finite Hankel transforms introduced here has the advantage of being able to handle complicated boundary conditions, whereas previous finite Hankel transform techniques are limited to problems involving Dirichlet, Neumann, or mixed (i.e., impedance) boundaries. [This work was supported by the Office of Naval Research, Contract No. N00014-86-K-0325.]
Effect of longitude on vertical distribution of ambient noise due to wind and coastal shipping
Authors: Fisher F. H. , Hodgkiss W. S.
Published: October 2005

Using multielement vertical arrays deployed from FLIP, measurements of the vertical distribution of ambient noise have been made at a latitude of 32N at three stations, 124W, 136W, and 150W, the latter station being about 1700 miles from the coast. The arrays used were deployed at the sound channel axis and were uniformly spaced at half-wavelength either at 200 or 300 Hz. From 75 to 300 Hz, the change in the vertical distribution of noise shifts, especially at higher frequencies, with increasing range from coastal shipping was in a manner consistent with the effect of chemical absorption on low-angle noise due to coastal shipping. Whereas a shipping noise pedestal at low angles is observed at all frequencies at short range, at long range the absorption effect makes the vertical distribution of ambient noise more isotropic at higher frequencies than that at low frequencies. Effects of wind speed on the vertical distribution of noise were also measured. As wind speed increases, ambient noise at angles greater than 15° from the horizontal (that is, noise of local origin) increases and approaches the levels of the pedestal at 124° W and presumably exceeds them at high wind speeds. Our results are orthogonal to those obtained by Kibblewhite et al., ARL/UT, with their deployment at 150 W of three Vedabs bouys on a north-south line to measure low-frequency absorption using northern shipping as a sound source. This conflict suggests a simultaneous north-south and east-west experiment as well as the utility of using a multielement vertical DIFAR array. [Work supported by ONR.]
Downslope propagation and vertical directionality of wind noise
Authors: Carey W. M. , Evans R. E. , Davis J. A.
Published: October 2005

Measurements of the vertical noise intensity versus angle [W. Carey and R. Wagstaff, J. Acoust. Soc. Am. 80, 1523–1526 (1986)] show the low-frequency distribution (≪200 Hz) to be broadly peaked about the horizontal, whereas the higher-frequency (≈400 Hz) distribution is peaked at the SOFAR angles (≈ ± 15) near the axis with a minima at the horizontal. This effect has been attributed to the noise from ships over the basin margins [R. Wagstaff, J. Acoust. Soc. Am. 69, 1009–1014 (1981)]. However, since spectral variation along the horizontal is smooth and the effect is observed in sparsely shipped areas of the world, wind noise must be included to explain this effect. R. W. Bannister [J. Acoust. Soc. Am. 79, 41–48 (1986)] has attributed this effect to high-latitude winds and the shallowing sound channel found in Southern Hemisphere waters. W. Carey [J. Acoust. Soc. Am. 79, 49–59 (1986)] attributed the effect to downslope propagation. Calculations between 50 and 400 Hz of the midbasin vertical directionality made with ASTRAL and PAREQ with a geoacoustic model were both found to show that the bottom behaves as a low-pass filter as the low-frequency energy at higher angles interacts with the bottom and is converted to low-angle energy in the deep sound channel while, at the higher frequencies and higher angle, energy is absorbed. This low-pass effect is consistent with experimental results. Propagation of low-grazing angle energy from the shelf is prominent in the computational results but not apparent in measured data, perhaps due to bathy-metric roughness.
Underwater noise produced by rainfall
Authors: Pumphrey H. C. , Crum L. A.
Published: October 2005

When a liquid drop falling through the air strikes a flat horizontal surface, there are acoustic emissions associated with this impact in both the air and the liquid [G. J. Franz, J. Acoust. Soc. Am. 31, 1080 (1959)]. Under certain conditions, a small air bubble can be entrained in the liquid by the impacting drop and stimulated into volume pulsations that radiate quite strongly. When several drops impact the surface in a short time interval as in rainfall, the acoustic emissions associated with the bubble oscillations appear to be a major contribution to the total acoustic emission over a considerable bandwidth. With artificially created rainfall, the spectral peak observed near 20 kHz for natural rainfall can be duplicated [Scrimger et al., J. Acoust. Soc. Am. 81, 79 (1987)]. However, with the addition of surface tension reduction agents, this peak is significantly reduced, implying that a major portion of the noise associated with rainfall is related to oscillations of the entrained bubble rather than due to the impact of the drop with the liquid surface. [Work supported by the ONR.]
Correlation and spectra of ambient sound generated by wind, whitecaps, and rain
Authors: Shonting David , Middleton Foster , Taylor Nancy
Published: October 2005

Although high correlation exists between oceanic ambient sound and wind speed and rain rate, the sound generation mechanisms remain obscure. This is due partly to the difficulty of simultaneously measuring the sound-generating phenomena at the location of the acoustic measurements. Also, there is a lack of frequency spectra of sound generated by the specific sources that may distinguish between wind- and rain (or spray)-generated sound and may hold important clues to the generating mechanisms. A pier facility was established on Narragansett Bay to monitor broadband sound associated with observed sea surface phenomena. Simultaneous recording of wind velocity, wave height, and rain rate allows precise comparison of the environmental variables with their acoustic response. In addition, visual observation is made of whitecapping and of interfering ship traffic. The ambient sound spectra are routinely generated as high-resolution FFT estimates on a MASSCOMP computer. Preliminary measurements show sound pressure to vary linearly with wind speed, with correlations ranging from 0.88–0.97. The slope changes around wind speeds of 6–7 m/s, i.e., where whitecaps form, The spectra of wind-generated sound display a faster roll-off than rain-produced sound, the latter exhibiting a broad peak around 15–18 kHz. Plots are presented of the intense insonification of the water column by rapidly advecting rain squalls. [Work supported by NAVSEA/AIRSEA systems commands.]
Observations of deep-ocean ambient noise using a seismometer array
Authors: Schreiner Anthony E. , Dorman LeRoy M. , Hildebrand John A. , Lahav Dalia , Bibee Dale
Published: October 2005

An array of 12 ocean-bottom seismometers (OBS) was deployed this past March in the deep ocean near San Diego. The array had a maximum extent of 150 m and a minimum interelement spacing of 8 m. The accuracy of the relative sensor positions was as good as 2 m in some cases. The seismometers were deployed in 3.8 km of water with a partially maneuverable vehicle at the end of a wire guided by a transponder net. The OBS capsules contained a vertical and two horizontal seismometers and a hydrophone. Three of the instruments had a low-frequency hydrophone of the Cox design. Three-min windows were recorded at intervals of 3 or 6 h for up to 30 days. Spectral levels have been calculated between 0.01 and 30 Hz. A power level drop of up to 40 dB below the microseism peak at frequencies below 0.1 Hz was observed. Similar spectra have been reported by Webb and Cox [J. Geophys. Res. 91, 7343–7358 (1986)]. At the time of the abstract deadline, correlation was being calculated as a function of sensor separation to determine the correlation length of the ambient noise.