bluuebunny/crossref_metadata_2025 huggingface dataset:
First- and second-order smoothing method for rough surface scattering
Authors: Ishimaru Akira
Published: October 2005

There have been several recent investigations on rough surface scattering. They are directed at removing restrictions on the conventional perturbation and Kirchhoff approximations and also to devise a theoretical model to explain recent experimental data on backscattering enhancement. This paper presents a theory based on the diagram method of Bass et al. and the smoothing method of Watson and Keller. The rough surface is represented by an equivalent fiat surface as was done by Bass-Fuks and Watson-Keller. The first-order smoothing solution is obtained from the Dyson and Bethe-Salpeter equations. The coherent field is identical to the Watson-Keller solutions. The first-order solution is close to Ito's solution, but it is reciprocal. The second-order smoothing solution has two terms. The ladder term is close to Ito's but the cross term exhibits some backscattering enhancement. Numerical examples for the one-dimensional Dirichlet surface are shown and compared with related theoretical calculations. [Work supported by NSF and ARO.]
Rough surface scattering with the parabolic wave equation using a split-step method
Authors: Thorsos Eric I. , Ballard John W. , Ewart Terry E.
Published: October 2005

The split-step algorithm introduced by Tappert and Nghiem-Phu [J. Acoust. Soc. Am. Suppl. 1 77, S101 (1985)] for solving the parabolic wave equation (PE) with 1-D rough surfaces subject to the pressure release boundary condition has been implemented. For an incident beam approaching the rough surface from below, the computational region is doubled to include an “image ocean” above the real ocean. A complicated index of refraction is constructed in the image ocean (dependent on the rough surface) so that the image beam and the incident beam combine to give a zero total field on the rough surface. The computed field below the rough surface should then be the solution desired. The details of this method will be outlined. Examples will be presented showing that the split-step solution is identical to that obtained with a PE integral equation approach. Comparisons will also be made between the split-step and exact integral equation solutions, showing the effect of the PE approximation in rough surface scattering. [Work supported by ONR.]
Community response to ground vibration excited by railway trains
Authors: Hatanaka Hisashi , Taya Koichi
Published: October 2005

The relation between the vibration level excited by railway trains and community responses in the vicinity of railway tracks is discussed. The vibration levels excited by trains were measured at eight residential areas. In the same areas, questionnaires on residents' responses to their environment were obtained. The measured vibration levels were arranged into maximum value Lmax, equivalent power value Leq, and day and night average value Ldn. The relation between these arranged values and the community responses was analyzed by the method of multidimensional scaling, etc. The community response corresponded to the magnitude of all arranged values in the same order. However, the vibration level excited by freight trains had a higher correlation to the community response than that excited by electric trains.
Effects of low-level sound on the mental task of writing random numbers
Authors: Tamura Akihiro
Published: October 2005

This study was undertaken to find out the basic problems involved in the effects of low-level sound on the performance of mental tasks by social psychological experiments in which subjects were not instructed that the test was to study the effects of sound on performance. The task was writing numbers from 0 to 9 randomly and successively. Subjects carried out the task under two different conditions; that is, a common condition and their own condition. They answered a questionnaire designed to investigate their feelings about the test room, the test condition, the experimenter, the co-experimenter, and themselves. These experiments were carried out three times for 3 years. They seem to justify the following conclusions. (1) The level of motivation has a major effect on the performance of the mental task. (2) Sounds which are not recognized as inevitable for behavior and the environment have a tendency to reduce performance even at low levels. (3) When the motivation for a task is present, low-level sounds, which are not recognized as inevitable, are found to be more annoying than those which are recognized as inevitable.
Noisiness of impulsive sound in background noise
Authors: Miyazono Hiromitsu , Ohto Keishi , Usagawa Tsuyoshi , Ebata Masanao
Published: October 2005

There are many studies of rating noise, but the method has not been established completely yet. Especially in the rating of impulsive noise, some problems remain unsolved even though an evaluation based on energy level is effective to some extent. In this study, the noisiness of impulsive sound is measured when background noise is present at various levels. The results show that the noisiness decreases as background noise increases and can be almost completely evaluated by the difference in the energy level of the impulsive sound and the background noise. However, when the level of the background noise is comparable to that of the impulsive sound, there is a considerable difference between the estimated value and the observed noisiness. To derive an accurate rating, two alternative methods are discussed: one is based on NOY and the other based on a background correction which is similar to a loudness rating.
Effects of fundamental frequency contour on the identification of resynthesized vowels with static formant frequency patterns
Authors: Hillenbrand James , Gayvert Robert T.
Published: October 2005

At a previous meeting, a study that was aimed at determining the identifiability of vowels based exclusively on static spectral information was discussed [Hillenbrand and McMahon, J. Acoust. Soc. Am. Suppl. 1 82, S81 (1987)]. In that study, a formant synthesizer was used to generate steady-state versions of 1520 vowels (76 speakers × 10 vowels × 2 repetitions) using Peterson and Barney's measured values of F0 and F1−F3 [J. Acoust. Soc. Am. 24, 175–184 (1952)]. The values of all control parameters remained constant throughout the 300-ms duration of each stimulus. Listeners in that study showed an error rate of approximately 25%, several times greater than the 5.6% error rate reported in the original Peterson and Barney study. The present study represents a follow-up designed to determine what role fundamental frequency movement might play in vowel identification. The new stimuli were identical to those of the previous resynthesis study except that all stimuli were generated with a falling pitch contour. Preliminary results suggest that the introduction of pitch movement decreases the error rate from approximately 25% to approximately 21%. [Work supported by Rome Air Development Center under contract F3060285-C-0008.]
The effect of tissue attenuation on Doppler flow imaging with linear phased array
Authors: Lee Xuguang , Ji Shusen
Published: October 2005

In Doppler flow imaging with linear phased array, each scan line is implemented on the basis of the pulsed Doppler principle. Transmitted pulses are backscattered by blood flow. Owing to a relatively deeper depth of detection, the backscattered Doppler signal must propagate a longer distance in soft tissues, which causes the acoustic signal to be influenced by the tissue's frequency-dependent attenuation characteristics. Consequently, a distortion of the power spectra of the backscattered Doppler signal appears. Unfortunately, there are little or no definitive data that address this problem. With the model of nonlinear frequency-dependent attenuation, the erroneous values of the image are predicted and an error matrix is presented to consider the scan conversion process. It is concluded that the mean frequency error of a sample volume depends on not only the attenuation characteristics along the propagation path but also on the flow direction relative to the probe. In the forward flow case, the mean frequency is shifted toward lower frequencies, while reverse flow causes the mean frequency to be shifted toward higher frequencies.
A study of the timing of sonoluminescence flashes from stable cavitation
Authors: Gaitan D. F. , Crum L. A. , Church C. C.
Published: October 2005

In a previous paper [J. Acoust. Soc. Am. Suppl. 1 80, S24 (1986)], experimental measurements of the timing of sonoluminescence (SL) flashes from stable cavitation in gassy water were described. The theory has been extended to examine larger acoustic pressure amplitudes and excellent agreement has been found between the measured and predicted phase angles of the emitted flashes. Since this phase angle is sensitive to the explicit formulation of the equations for bubble dynamics, these data can be used to examine the applicability of various theoretical models. The results of this comparison will be presented and reasons for the inapplicability of certain formulations will be given. [Work supported in part by the NIH and the ONR.]
Noise reduction by an active vibration controller
Authors: Sero Kazuto
Published: October 2005

This paper proposes a method of noise reduction by way of controlling the vibration of a wall between two rooms by an active vibration controller. This is done by using an additional sensor to measure the wall vibration and feed it back to the controller. The boardspeaker is used primarily to generate sound by exciting the board. However, in this study, it is also used as the actuator. The performance of the controller is tested on a veneer board, which is 1200 × 900 × 5 mm in size, under sound vibration by another speaker. The evaluation of this method is performed experimentally by comparison of the dynamic deflection of the board with and without the controller. The experimental results show a significant reduction level to 1/10 of the board dynamic deflection at frequencies below 100 Hz. The effectiveness of the method is confirmed theoretically by analog simulation studies. Also, the stability of the controller is examined using the root locus method.
Signal pressure received by a hydrophone placed on a plate backed with a compliant baffle
Authors: Ko Sung H. , Schloemer Howard H.
Published: October 2005

A theoretical model was developed to evaluate the signal pressure received by a hydrophone placed in front of a plate backed with a compliant baffle layer. The compliant baffle layer between the plate and the semi-infinite fluid medium is designed for reducing pressure fluctuations from Papers nonsignal directions. Because of its acoustic softness, the signal received by the hydrophone without a plate would be degraded. Therefore, it is desirable to improve the signal reception by covering the baffle layer with a hard plate. The baffle layer considered here is the compliant-tube array, modeled by Junger [J. Acoust. Soc. Am. 78, 1010 (1985)], to represent a homogeneous (dispersive) fluid layer. Effects of various parameters such as the angle of incidence, the aspect ratio of the compliant tube, the distance between tubes, and the damping of the tube on the received signal pressure are presented. Calculations made for the nondispersive fluid layer are compared with those made for the dispersive fluid layer.
Time domain finite difference modeling of acoustic wave scattering from an elastic cylinder
Authors: Dougherty Martin E. , Stephen Ralph A.
Published: October 2005

A 2-D finite difference formulation of the elastic wave equation has been used to investigate seismo/acoustic scattering from laterally heterogeneous oceanic crust. In an effort to validate the use of this code to handle such problems, the same formulation has been used to solve the canonical problem of scattering of an acoustic wave from an infinite elastic circular cylinder. Analytical theory predicts that an acoustic wave will be scattered only into certain distinct azimuths after interaction with a cylinder of given elastic parameters. These scattering angles and amplitudes are very sensitive to both the frequency of the source and the diameter of the cylinder. The finite difference results of scattering of both a relatively broadband plane wave pulse and a series of monochromatic cw plane waves at different frequencies are presented. In general, the scattering pattern produced by the pulse source does not agree well with the analytical solution for the pulse center frequency. This is due to the relatively broad frequency band of the pulse. However, scattering patterns and amplitudes produced by the monochromatic cw plane wave sources do agree very well with the expected analytical solutions.
Apical hair cells and hearing
Authors: Moody David B. , Prosen Cynthia A. , Sommers Mitchell S. , Stebbins William C. , Smith David W.
Published: October 2005

Hair cells located in the cochlear apex were destroyed by low-frequency noise exposure or by application of a − 140 °C cryoprobe to the bony wall of the apical turn. Complementary lesions that spared apical hair cells were produced by treatment with ototoxic antibiotics. Behavioral thresholds were measured in monkeys, chinchillas, and guinea pigs before and after these treatments. The results showed (1) with loss of up to 80% of apical outer hair cells, thresholds were unchanged; (2) with complete loss of both outer and inner apical hair cells, thresholds shifted approximately 20 dB at low frequencies; and (3) with complete loss of hair cells except in the apex, some hearing remained, with better hearing at low frequencies. Addition of a high-pass masker always elevated high-frequency thresholds, but only elevated low-frequency thresholds when all apical hair cells had been destroyed, indicating that basal hair cells can respond to low-frequency stimuli. Compared to data from more basal portions of the cochlea, these data suggest that functional differences exist between the base and apex, with comparable damage in the apex resulting in less hearing impairment. [Work supported by NINCDS grants NS05785 and NS25564.]
Spectral characteristics of the responses of primary auditory-nerve fibers to frequency-modulated signals
Authors: Khanna S. M. , Teich M. C.
Published: October 2005

The spectral responses of cat single primary auditory nerve fibers to sinusoidal frequency-modulated (FM) acoustic signals applied to the ear are reported. Period histograms were constructed from the neural spike-train data, and the frequency spectrum was determined by Fourier transforming these histograms. Several clusters of spectral components were present. The lowest-frequency cluster consists of components at dc, at the modulation frequency, and at its harmonics. In the next cluster, components surround the carrier frequency and are separated from it by the modulation frequency and its harmonics. Higher-frequency clusters surround frequencies that are twice and three times the carrier frequency. The components in each cluster are separated from the multiples of the carrier frequency by the modulation frequency and its harmonics. The magnitudes of the spectral components were investigated for carrier frequencies located below, at, and above the unit characteristic frequency (CF), and for different signal levels, modulation frequencies, and modulation indices. The components at the modulation frequency and its harmonics were strong and present over a wide range of signal levels, carrier frequencies, modulation frequencies, and nerve-fiber characteristics. The presence of components at the modulation frequency indicates that a demodulation process is occurring. This process may be significant for speech recognition. [Work supported by NIH and NSF.]
Comparison of rapid and short-term adaptation in the auditory nerve
Authors: Smith R. L. , Westerman L. A.
Published: October 2005

Events during the first few milliseconds of auditory-nerve adaptation appear to differ qualitatively from those occurring later on. This has previously been revealed using PST analysis with various bin-width sizes, and also by dissecting the perstimulatory response into two components: rapid and short-term adaptation. In order to further evaluate the extent of these differences, a series of perstimulatory and post-stimulatory comparisons were made involving the recovery from prior stimulation and the effects of sound frequency and intensity. Post-stimulatory effects on both rapid and short-term firing rates appear to be proportional to the magnitude of the conditioning or adapting response, and otherwise to be independent of sound frequency. Following stimulation, the rapid component recovers more quickly than does the short-term component. Changes in sound frequency do not alter the amount of perstimulatory short-term adaptation, but effect the rapidly adapting component in a complex manner. This may be due in part to frequency splatter produced by non-CF tones, and also to phase locking to individual cycles of low-frequency tones. Implications for models of adaptation will be discussed. [Work supported by NSF and NIH.]
Noise reduction in speech signal processing by neural network and vector quantization
Authors: Nakata Kazuo , Sugiura Akihiko
Published: October 2005

The autocorrelation function provides basic data for speech signal processing by the linear prediction algorithm. The first step toward developing a practical speech signal processing technique is noise reduction in the autocorrelation function of speech corrupted by noise. Recent developments in neural network (NN) techniques have achieved a respectable performance of classifiers even in the worse condition where SNR is less than 0 dB. The vector quantization (VQ) technique, on the other hand, also provides reasonable bases for the discrete description of continuous speech signals. The combination of the two techniques gives the possibility of excellent noise reduction in speech signal processing. Some recent experimental results based on the above principle are given in detail.
Improved training procedures for hidden Markov models
Authors: Rabiner Lawrence R. , Lee Chin-Hui , Juang Biing-Hwang , Roe David B. , Wilpon Jay G.
Published: October 2005

Techniques for training hidden Markov model (HMM) parameters from a labeled training set of data are well established and include the forward-backward algorithm as well as the segmental K-means algorithm. These algorithms have been shown to be capable of estimating the parameters of an HMM based on mathematically well-founded techniques. In practice, however, difficulties are often encountered when estimating some of the HMM parameters. These difficulties are generally the result of having insufficient training data to give robust and reliable parameter estimates. Typically, the model parameters most affected by having insufficient training data are the spectral parameter variance estimates, and the estimates of parameters related to the modeling of state duration. Although techniques have been proposed for improving estimates of the variances due to the effects of insufficient training data, the results have not proven adequate in some cases. As such, improved training techniques (which give better recognition performance) have been devised for controlling the minimum variance estimate of any spectral parameter, and for thresholding and clipping state duration parameter estimates. These improved training methods have been tested on several databases with good success. In addition, advanced techniques for creating multiple HMMs from the training data (i.e., for speaker independent recognition) have been devised and have proven successful for modeling large databases of training material.
Speaker adaptation in HMM
Authors: Niimi Yasuhisa , Kobayashi Yutaka
Published: October 2005

A method for speaker adaptation of a code book in vector quantization and its application to word recognition based on the HMM are reported. Under the assumption that speech vectors can be represented by the two-factor model—that is, a sum of the two main effects of “phoneme” and “speaker” and the interaction between the two—the vector space is divided into narrow subspaces in which the speaker effect is considered constant. In each subspace, the displacement vector due to that effect is estimated by using training utterances. The speaker adaptation of a reference code book is completed by moving all the code vectors contained in the subspace parallel to the displacement. The recognition of 65 Japanese city names was performed. A speaker-independent reference code book and the HMMs of the words were designed by using utterances spoken by five male speakers. For the utterances produced by 20 other male speakers, the average word recognition rate was 97.4% in the speaker-adaptive mode and 94.1% in the speaker-independent mode.
Impulse response and reverberation time measurements using a periodic pseudorandom sequence
Authors: Chu W. T.
Published: October 2005

The cross-correlation method using a periodic pseudorandom sequence or M sequence for impulse response and reverberation time measurements proposed by Schroeder and his colleagues [H. Alrutz and M. R. Schroeder, Proc. 11th Congr. Acoust., Paris (1983)] represents a very significant contribution to the fields of Architectural Acoustics and Noise Control. With this technique, it is not only possible to make measurements in auditoria or offices without disturbing the audience or the occupants, but also in factories or construction sites without closing down their operations. Implementation of this method requires special signal processing techniques which will be reviewed in this presentation. Examples of measurements in simulated noisy environments will also be presented.
Application of piezoelectric composite for large-area hydrophone arrays
Authors: Geil Fred G. , Ting Robert Y.
Published: October 2005

Large-area hydrophone arrays using flexible materials are subject to high levels of interelement coupling arising from the material in-plane sensitivity coefficient g31, in the presence of flexural energy within the array module. A “0—3” type piezocomposite offers a solution to this interelement coupling problem without introduction of fabrication complexity such as stiffening of elements. A simplified array fabrication concept will be presented, as well as results from small array tests using commercially available “0—3” piezocomposites based on lead titanate powder dispersed in a rubber matrix. These test results show a near-zero level of acoustic contamination, shown by directivity patterns and relative phase responses. Besides low interelement coupling, cost, waterproofing, and electrical interfacing are also important in the design considerations of affordable, realistic arrays for the seawater environment. The advantages of using “0—3” piezocomposites in meeting these additional requirements will be briefly discussed. [Work supported by ONT.]
Comparison of ceramic and thick-film PVDF hydrophones for large surface arrays
Authors: Tocquet B. , Fromont B. , Josserand M.
Published: October 2005

A special process allows the manufacturing, in industrial conditions, of thick, nonvoided PVDF films (up to 2 mm) with excellent piezoelectric properties. The characteristics of PVDF are compared with those of some classical PZT ceramics. From such films, very simple and reliable large surface sensors have been designed and developed. Compared to classical ceramic hydrophones, these sensors show a great stability versus temperature and pressure and a very good sensitivity to self-noise ratio for low-frequency passive listening systems. Furthermore, these large surface sensors fit very well with hard baffles and are less sensitive to flow noise than small ceramic hydrophones. Some examples of results obtained with such new hydrophones are given. Clusters of such PVDF hydrophones can be encapsulated with viscoelastic “pρc” materials in flat panels and fixed, with decoupling layers or directly on the hull of the ship, in direct contact with the hydrodynamic flow, leading to a very compact and reliable technology for large flank arrays.
Absolute measurement of particle velocity of time-harmonic acoustic waves
Authors: Vignola Joseph , Jarzynski Jacek , Berthelot Yves H.
Published: October 2005

Experiments are being conducted to measure by optical means the amplitude and the phase of a steady-state, time-harmonic compressional wave being produced inside a standing wave tube filled either with air or with water. The technique used is an extension of laser Doppler anemometry in acoustics developed by Taylor [J. Acoust. Soc. Am. 59, 691–694 (1976)]. It consists of illuminating with laser beams a small probe volume in water in which slowly drifting suspended microparticles are moving in phase with the acoustic field. The light scattered from the particles is Doppler shifted and contains information about the amplitude and the frequency of the particle motion. Preliminary results indicate that signal analysis performed on a spectrum analyzer is probably not the optimum processing technique, mainly because signals are being analyzed and displayed even at times when there is no particle in the probe volume to scatter the laser light. Instead, more meaningful and repeatable results can be obtained by using a digital data acquisition system triggered to capture the signal only when light is being scattered from a particle. The data are then transferred to a computer for further processing. [Work supported by ONR.]
Time-domain scattering from tungsten-carbide targets
Authors: Üerall H. , Dickey J. W. , Werby M. F. , Collins Michael D.
Published: October 2005

Scattering calculations are almost exclusively performed in the frequency domain. Although form functions are computed regularly, they are rarely utilized to obtain time-domain results. This is surprising because the payoff for the small amount of extra effort required for Fourier synthesis includes results that are easy to interpret in terms of causality. This approach will be used to study resonances of tungsten-carbide spheroids for aspect ratios ranging from 1–6 and for kL/2 up to 26. Tone bursts, cw pings, and Gaussian sources will be utilized to isolate resonances and to determine their nature. An analysis of arrival time as a function of aspect ratio gives credibility to the interpretation that resonances induced along the axis of symmetry are due to Rayleigh waves as first proposed by Flax et al. [J. Acoust. Soc. Am. 71, 1077–1082 (1982)]. [Work supported by the Naval Ocean Research and Development Activity.]
Characterization of surface-breaking and near-surface cracks by the acoustic microscope
Authors: Achenbach J. D.
Published: October 2005

The acoustic microscope is now becoming a more frequently used instrument for the study of surfaces and near-surface inhomogeneities. One potentially important application concerns the detection and characterization of very small surface-breaking or subsurface cracks. For this application, it is important that the instrument can be used in the quantitative rather than in the imaging mode. It is expected that surface waves in conjunction with a line-focus configuration would provide the most effective probing modality. An effective use of this modality will, however, require insight in the interaction of surface waves with cracks. It is the objective of this presentation to provide such insight on the basis of analytical and numerical solutions to the interaction problems. In this work, the crack is inclined under an arbitrary angle with the normal to the free surface of a half-space. The representation integral for the scattered displacement field is employed to obtain expressions for the surface tractions. In conjunction with the conditions on the crack faces and on the surface of the half-space, these expressions yield a set of boundary integral equations, which are solved numerically by the boundary element method. Results for the reflection and transmission coefficients, as well as for the time-averaged energy-flux radiated onto the solid by body waves, are presented. Parametrical studies reveal the variation of these results with the angle of inclination of the crack, its length, and with the frequency of the incident wave.
Acoustic microscope with an acousto-optic array detector
Authors: Cook Bill D. , Miller J. Ted , Lewis D. Kent
Published: October 2005

This very low-frequency microscope is a pitch-catch device where the receiver is a large, synthetic, linear acousto-optic array. The focused ultrasound is transmitted through and reflected from a sample off-axis. The reflected or transmitted pressure is sampled along several lines with a laser and photodiode. This recorded information is then processed to find the angular distribution of the sound. Comparison with incoming sound yields reflection and transmission coefficients. Measurements have been made on thin plates with both pulse and continuous waves. With continuous waves, the reflection and transmission coefficients are sometimes greater than unity, probably due to mode conversion. Pulsed waves yield presentations of dispersion of the modal velocities. [Work supported by NATO and DOE.]
Experimental assessment of a method for calibration of ear canals at high frequencies
Authors: Berkovitz Robert , Stevens Kenneth N.
Published: October 2005

An earlier publication [Stevens et al., J. Acoust. Soc. Am. 81, 470–484 (1987)] described a procedure for determining the absolute sound pressure at the medial end of the ear canal at frequencies in the range 8–20 kHz when a sound source is coupled to the ear through a lossy tube. The present paper reports comparisons of estimated and measured apex sound pressures for various ear-canal models. Agreement was within 1–2 dB for simplified models with rigid perpendicular and oblique terminations. Changes in length over a range similar to that of human ear canals had little effect on results. Similar agreement up to about 18 kHz was obtained for nonuniform ear-canal models with constrictions to less than one-half the cross-sectional area of the average ear canal, and the error was slightly larger above 18 kHz. Constrictions in the models led to nonuniformities in the frequency response at the apex. Estimates of apex sound pressures for a number of human ears exhibited features similar to those observed for the models. Improvements in signal processing and fitting techniques made since the earlier publication are described. [Work supported by a grant from NINCDS. K. N. Stevens is also at the Massachusetts Institute of Technology.]
Amplitude compression for cochlear implant speech processors
Authors: Knapp R. Benjamin , Dent Laurel J. , Liang David H. , Shannon Robert V.
Published: October 2005

One requirement in speech processor design for cochlear implants is determining the instantaneous mapping of the broad acoustic dynamic range into the narrow dynamic range for electrical stimulation. Magnitude estimation was used to measure implanted subjects' growth of loudness for several stimulus waveforms including sinusoidal, pulsatile, and amplitude modulated pulsatile signals. Loudness growth was best modeled by a power law function of the form first proposed by Stevens [J. Acoust. Soc. Am. 27, 815–829 (1955)] to describe normal listeners' loudness growth for acoustic stimuli. The power function exponent varied with electrical stimulus type, but when loudness growth was expressed as a function of proportion of dynamic range, a single exponent described all the observations. The Stanford real-time speech processor was programed to permit the subject to adjust a fractional power compression designed to elicit normal loudness growth by correcting for the difference between the normals' and implant subjects' exponents. The “correction exponent” was confirmed to be the same for all electrodes and waveforms. The physiological implications of the findings and the effects of the compression on speech discrimination will be discussed.
Temporal and spatial interaction of pulsatile electrical stimuli of the auditory nerve in cats
Authors: Liang David H.
Published: October 2005

It is fairly clear that the separate electrodes of a cochlear prosthesis do not represent independent channels for transmitting information. The currents from the different electrodes interact both spatially and temporally. To examine these interactions, recordings were made from single units in the VIIIth nerve of cats that had an eight-channel stimulating array acutely implanted in the scalar tympani. Spatial interaction was studied by determining the threshold for simultaneous in-phase stimulation of two electrodes. It was found that the resulting thresholds were not always predicted well by simple linear summation of the effectiveness of the two electrodes used in isolation. Temporal interaction was examined by studying the effect of a subthreshold stimulus on the threshold of a subsequent stimulus. This experiment has particular importance for stimulation schemes that seek to reduce interaction between adjacent electrodes by interleaving them in time. The results suggest that, for biphasic stimulation, the residual effect of stimuli up to 75% of threshold is minimal. [Work supported by NIH.]
Acoustic and psychoacoustic evaluation of suppression and synchronization of spontaneous otoacoustic emissions
Authors: Long Glenis R. , Sivaramakrishnan Savithri , Tubis Arnold
Published: October 2005

The impact of external tones on spontaneous otoacoustic emissions (tonelike stimuli of cochlear origin) depends on both the level and relative frequency of such tones. The most commonly measured kinds of interaction are suppression, beating, and entrainment. Psychophysical correlates of these forms of interaction are masking, monaural diplacusis. and some aspects of threshold microstructure. The relation between the different acoustic and psychophysical observations is investigated in three individuals with large spontaneous emissions both under normal circumstances and when the emissions are reduced by aspirin consumption. The acoustic observations will be compared to the output of digital cochlear models which assume that an otoacoustic emission can be modeled as a limit-cycle oscillator.
Spontaneous otoacoustic emissions cause disruptive tinnitus in 4 of 88 subjects
Authors: Penner M. J.
Published: October 2005

Experimental tests linking SOAEs with problem tinnitus are detailed by presenting one case study of a female subject whose tinnitus consists of an annoying SOAE in the right ear and tinnitus unrelated to SOAEs in the left ear. The tests involve demonstrating that the tinnitus only disappeared when a pure tone suppressed the SOAE and another continuous tone masked the tinnitus, and that the isomasking contour for the tinnitus caused by the SOAE was frequency specific whereas the contour for the tinnitus unrelated to the SOAE was not. Using similar tests on 87 additional subjects, an estimate was made of the percent of tinnitus sufferers whose tinnitus was caused by SOAEs: the 95% confidence limits of this estimate were 0.20% and 9%. [Work supported by NIH.]
Effects of high-level impulse noise intensity, number, and rate on hearing
Authors: Hamernik Roger P. , Ahroon William A.
Published: October 2005

The effects of very low-frequency, energy-content impulse noise exposures on hearing were studied in a population of 109 chinchillas exposed to 1, 10, or 100 impulses presented at 150-, 155-, or 160-dB peak SPL at rates of one every 6, 60, or 600 s. The TTS and PTS were measured using the auditory evoked potential. Histological measures were obtained using standard surface preparation techniques. The following preliminary conclusions can be made. (1) There was no statistical difference in the amount of hearing loss or the amount of sensory cell loss for exposure to a single impulse at 150 or 155 or 160 dB peak SPL. Individual animals showed no permanent hearing loss and no significant sensory cell loss. (2) There was a considerable amount of variability or degree of susceptibility across animals as the severity of the exposure increased. This increase in susceptibility seemed to be tied more to peak levels of the stimulus than to the total energy. The variability produces, in extreme cases, a complete dichotomy in the results. (3) The permanent effects seemed to be dependent upon peak levels more than upon the total energy in the exposure stimulus. Also, for a constant peak and energy level, the more rapid presentation rate produced the greatest effect.
Effects of periodic rest on hearing loss and cochlear damage following interrupted exposure to high-frequency noise
Authors: Clark W. W. , Bohne B. A. , Reilly T. J.
Published: October 2005

Two groups of chinchillas were exposed to an octave band of noise (OBN) centered at 4.0 kHz, 86 dB SPL for 6 h per day. The first, made monaural prior to the start of the exposure, was exposed for 36 days. The second was made monaural after 36 days of exposure and exposed for an additional 36 days. Thresholds were assessed twice daily with a positive reinforcement technique, and the cochleas were prepared for microscopic examination 3 months after the exposure. All subjects sustained initial threshold shifts of 45–55 dB at test frequencies within the exposure band (3.35–8.0 kHz); after 5–10 days, threshold shifts began to decline and recovered about 20 dB over the next 10 days of exposure. Thresholds were stable after about 20 days of exposure for both the 36- and 72-day groups. Small, permanent threshold shifts were measured within the exposure band for both groups of animals. Cochleas exposed for 36 days had mild but significant damage in the basal turn; those exposed for 72 days sustained slightly more damage. [Work supported by Grant No. OH 02128 from NIOSH.]
Auditory temporal and reading disability
Authors: Watson Betty U.
Published: October 2005

Deficits in auditory processing have been proposed by many authors as an etiology of some cases of specific reading disability [e.g., P. Tallal, Brain Lang. 9, 182–198 (1980); E. B. Zurif and G. Carson, Neuropsychologia 8, 351–361 (1970)]. Such deficits arc hypothesized to impair precise perception of speech sounds, which in turn interferes with acquisition of sound-symbol correspondence when learning to read. This hypothesis was evaluated in six studies that used the Test of Basic Auditory Capabilities (TBAC) [Watson et al., J. Acoust. Soc. Am. Suppl. I 71, S73 (1982)] to assess temporal processing. The first study found significant differences in temporal processing between normal and learning-disabled high school students, but these differences were not replicated in a second study with a college sample. The next two studies found substantial relationships between measures of academic aptitude/intelligence and temporal processing. The last two studies report correlations between the TBAC and performance on a variety of phonic analysis tasks. Few significant relationships were found when the effects of intelligence or academic aptitude were controlled. Overall, this series of studies fails to support the causal role of temporal processing deficits in the etiology of specific reading disabilty [Work supported by NIH.]
Voco-auditory functions of the chimpanzee
Authors: Kojima Shozo , Tatsumi Itaru F. , Kiritani Shigeru , Hirose Hajime
Published: October 2005

Auditory sensitivity, frequency, and intensity difference thresholds were measured in chimpanzees. Chimpanzees had a W-shaped auditory sensitivity function. That is, they were less sensitive to the midfrequency range of 2–4 kHz than to 1–8 kHz (the best frequencies). The frequency and intensity difference thresholds at 1 kHz and 70 dB SPL were about 10 Hz and about 1.5 dB, respectively. In the perception of vowels, reaction time for vowel discrimination was longer for [i]-[u] and [e]-[o] pairs than for other vowel pairs. This vowel perception can be explained by the W-shaped auditory sensitivity function. In the perception of stop consonants, distinction of the place feature was more difficult than that of the voicing feature. Preliminary experiments indicated that chimpanzees show vocal tract normalization effects for vowels and phoneme boundary effects for stop consonants.
Macaques show context effects in speech perception
Authors: Stevens Erica B. , Kuhl Patricia K. , Padden Denise M.
Published: October 2005

The phonetic distinction between the stop consonant /b/ and semivowel /w/ is cued primarily by duration of the initial formant transition. Syllable duration, however, is a context that influences human adults' and infants' perception of this distinction. For example, when overall syllable duration is long (296 ms), infants show poor discrimination of syllables with transition durations of 16 and 40 ms, but good discrimination of syllables with transition durations of 40 and 64 ms. However, when overall syllable duration is short (80 ms), the opposite is true [P. D. Eimas and J. L. Miller, Science 209, 1140–1141 (1980)]. The current study addresses whether rhesus and Japanese macaques' perception of the /b-w/ distinction is also influenced by syllable duration: Six CV syllables used in the Eimas and Miller (1980) study were presented to five macaques using a same-different discrimination paradigm. Results showed that macaques' pattern of discrimination was similar to the pattern shown by infants, thus supporting an auditory-based mechanism for the effect of syllable duration in phonetic perception. [Work supported by NSF and NIH.]
Tone-on-tone masking in the goldfish: The effects of signal phase for signal and masker at 500 Hz
Authors: Fay R. R.
Published: October 2005

Thresholds for detecting 20-ms, 500-Hz tone bursts (7-ms rise/fall times) in the presence of a 500-Hz masker were measured in goldfish using classical respiratory conditioning with a tracking psychophysical procedure. Thresholds were obtained using a continuous masker, and with the signal temporally centered in a 200-ms gated masker. Thresholds were determined as a function of the phase angle at which the signal was added to the masker. For the continuous masker, the just-detectable sound-pressure increment (at 0-deg phase angle) was 0.15 dB, and the just-detectable decrement (at 180 deg) was −0.22 dB. For the gated masker, the thresholds were 1.75 and −2.4 dB, for the in-phase and phase-inverted signals, respectively. Varying signal phase between 0 and 90 deg resulted in a progressively declining increment threshold, reaching about 0.5 dB at 90 deg. Since the threshold increment is not constant as a function of signal phase, some other cue(s) must underlie detection. Spectral “splatter” is not likely the cue since the increment threshold varied little when signal rise time was reduced to zero, or when the signal was bandpass filtered (96 dB/oct) at 500 Hz. This suggests that goldfish directly detect a transient phase shift occurring in the masker waveform when the out-of-phase signal is added. [Work supported by NINCDS Program Project Grant.]
The Franssen effect and the localization plausibility hypothesis
Authors: Rakerd Brad , Hartmann William Morris
Published: October 2005

The Franssen effect is obtained with two loudspeakers in a room. If a sine tone is abruptly turned on at the left loudspeaker, then slowly faded off while the right loudspeaker is slowly faded on, a listener will judge that the tone continues to come from the left loudspeaker, even though the left loudspeaker is not sounding at all. Our own studies of localization of sound in rooms have led to a principle of localization called the “plausibility hypothesis.” One of the predictions of this hypothesis is that in an anechoic room the Franssen effect should fail [B. Rakerd and W. M. Hartmann, J. Acoust. Soc. Am. 78, 524–533 (1985)]. Experimental studies are reported using the Franssen stimulus both in an ordinary room and in an anechoic room. The results of the experiment support the prediction. [Work supported by the National Institutes of Health.]
A method of measuring the nonlinearity parameter B/A of a medium using the angle dependency of β
Authors: Tsuchiya Takao , Watanabe Yoshiaki , Urabe Yasumasa
Published: October 2005

A new method of measuring B/A using a parametric receiving array is presented. The coefficient of nonlinearity β is treated exactly and is divided into two physically different terms. The effective value of βe (= cos θ + B/2A) is defined as the function of the crossing angle θ of two waves. The angle-dependent pattern of βe is applied to the measurements of B/A. The following two practical methods are presented: (1) using the non interacting angle θn [= cos−1(− B/2A)] and (2) using the front-back ratio of βe. The measurements were carried out using the two above methods for a gaseous medium and using (2) for water. The array lengths of the probing wave used were 90 mm and 40 kHz for gas, and 40 mm and 5 MHz for water, respectively. The phase modulation of the probing wave was observed and the angle dependency of β was detected. It was found that the estimated values of B/A showed good agreement with theoretical values and with the observed results presented by the other researchers.
Dependence of interaural time relationships in the cat upon sound source direction and frequency
Authors: Mind J. E. , Musicant A. D. , Chan J. C. K. , Kochhar R. K.
Published: October 2005

Using a probe microphone surgically implanted near each eardrum of an anesthetized cat, the time waveform was recorded in both ears in response to a click as the azimuth and elevation of a loudspeaker were varied. Two forms of interaural time delay (ITD) were calculated. Arrival time (AT) for each ear was based upon the leading edge of the click waveform and the first measure of ITD was taken as the difference in AT for the two ears. The FFT analysis of the time waveforms also yielded the interaural phase difference (IPD) for the individual frequency components. The phase-derived ITD was calculated as IPD/ω and enabled study of the dependence of ITD upon frequency. Both measures of ITD varied systematically with azimuth and elevation and reached values greater than would be expected from the interaural distance. Kuhn [Directional Hearing, edited by W. Yost and G. Gourevitch, (Springer, Berlin, 1987), Chap. 1] has shown for the human that ITDs for low and high frequencies are in the ratio of 3:2, respectively. The present study yields similar findings in the cat and confirms and extends earlier work.
Detection and lateralization based on interaural temporal disparities: Time or phase?
Authors: Bernstein Leslie R. , Trahiotis Constantine
Published: October 2005

This presentation will address whether the binaural auditory system's sensitivity to interaural temporal disparities is more appropriately characterized as a sensitivity to physical differences, across the ears, of time or phase per se. In two recent studies [W. A. Yost and R. H. Dye, J. Acoust. Soc. Am. 83, 1846–1851 (1988); W. A. Yost, J. Acoust. Soc. Am. 70, 397–409 (1981)], Yost and Dye measured thresholds for detection of, and extent of, laterality produced by interaural temporal disparities of pure tones between 200 and 5000 Hz. Because performance in both tasks appeared to be constant across frequency for a given interaural phase difference rather than for a given interaural time difference, Yost and Dye argued that phase is the most appropriate independent variable. This notion is inconsistent with several other sets of data obtained with complex signals and with pure tones. Furthermore, the constraints that the processing of interaural phase per se imposes on modern cross-correlation-based models of binaural hearing stand in contrast to the more parsimonious explanations based on interaural time.