bluuebunny/crossref_metadata_2025 huggingface dataset:
Nonlinear propagation through a fluid of waves originating from a biharmonic sound source
Authors: Hedberg Claes M.
Published: October 2005

A sufficiently strong sound source generates in a thermoviscous fluid, due to nonlinearity, a frequency spectrum consisting of all multiples of the original frequencies and the sums and differences of these multiples. After a certain distance, a shock front is formed because of the energy transfer from lower to higher frequencies. In the case of two original frequencies as a source (the biharmonic case), the damping of high frequencies leaves us at a large distance from the source with primarily the difference frequency. The propagation of plane waves is described by the Burgers’ equation whose solution in the regions before and after the shock formation exhibits significantly different approximate analytical expressions. In this work, an analytical description of the total amplitude in the region after formation of shock in the case of a biharmonic sound source is found. This is a generalization of the well-known Khokhlov solution for a monochromatic (single frequency) source. This description is turned into a Fourier series which can be specialized into the classical Fay solution for a monochromatic source. From this Fourier series the behavior of the individual frequencies is obtained, in particular the difference frequency which is also examined.
Measuring auditorium seat absorption
Authors: Davies W. J. , Orlowski R. J. , Lam Y. W.
Published: October 2005

There is a need for a more accurate laboratory measurement method to predict auditorium seat absorption. The traditional method tends to overpredict the absorption of the exposed front and sides of seating blocks. An alternative reverberation chamber method was studied that involves the use of barriers to obtain realistic measurements of front and side absorption. This method was validated by comparing measurements of seats made in a reverberation chamber with in situ absorption data for the same seats, calculated from reverberation time measurements in ten auditoria with and without the seats present. The accuracy of the alternative method was satisfactory in all cases, although a severe lack of diffusion in two of the halls hindered the validation process. It was found that using a frequency-constant edge correction strip to account for side and front absorption could lead to significant errors in auditorium absorption prediction.
A new formulation for the vibrations and sound radiation of fluid-loaded plates with elastic boundary conditions
Authors: Berry Alain
Published: October 2005

This paper considers the dynamic response and acoustic radiation of baffled rectangular plates elastically restrained against deflection and rotation along the four edges, and exposed to a half-space of dense fluid. Classical methods using in vacuo plate eigenfunctions to formulate the fluid/structure coupling are limited to the simply supported case and lead to considerable difficulty, both theoretical and numerical. The proposed method employs a variational formulation of the structural motion and a minimization process over a subspace of suitable displacement functions. The selection of simple polynomials to generate this subspace is shown to have many advantages: (i) it permits an analytic solution to be obtained for both the resistive and reactive components of the plate’s radiation impedance, (ii) it furnishes a natural decomposition of the radiator into monopoles, dipoles,..., 2n-upoles, and (iii) it decreases the computation cost and increases the generality of the problem addressed, as compared to available methods.
Plate mode velocities in graphite/epoxy plates
Authors: Prosser W. H. , Gorman M. R.
Published: October 2005

Measurements of the velocities of the extensional and flexural plate modes were made along three directions of propagation in four graphite/epoxy composite plates. The acoustic signals were generated by simulated acoustic emission events (pencil lead breaks or Hsu–Neilsen sources) and detected by broadband ultrasonic transducers. The first arrival of the extensional plate mode, which is nondispersive at low frequencies, was measured at a number of different distances from the source along the propagation direction of interest. The velocity was determined by plotting the distance versus arrival time and computing its slope. Because of the large dispersion of the flexural mode, a Fourier phase velocity technique was used to characterize this mode. The velocity was measured up to a frequency of 160 kHz. Theoretical predictions of the velocities of these modes were also made and compared with experimental observations. Classical plate theory yielded good agreement with the measured extensional velocities. For predictions of the dispersion of the flexural mode, Mindlin plate theory, which includes the effects of shear deformation and rotatory inertia was shown to give better agreement with the experimental measurements.
Harmonic distortion in moving coil transducers caused by generalized Coulomb damping
Authors: Sherman Charles H. , Butler John L.
Published: October 2005

A generalized Coulomb damping model for transducer friction is presented and applied to an underwater sound moving coil transducer. Measured harmonic amplitudes in the acoustic output of a certain moving coil transducer showed characteristics that indicated they were caused by friction. However, calculations based on the usual definition of Coulomb damping predicted only odd harmonics while the measurements contained both odd and even harmonics. A generalization of Coulomb damping was developed in which the friction force depends on position and direction of motion. This damping model predicts both even and odd harmonics and gives satisfactory agreement with the data. This friction-generated distortion is independent of drive level and can be important since it puts a lower bound on the distortion-free dynamic range.
Acoustic scattering from swim-bladder fish at low frequencies
Authors: Ye Zhen , Farmer David M.
Published: October 2005

Models of acoustic resonant scattering from swim-bladder fish are sensitive to the choice of damping coefficient and resonance frequency. Existing theory does not provide a consistent representation of the role played by tissue surrounding the swim bladder in determining this coefficient and empirical fits are therefore often used to adjust model parameters for specific examples. In an attempt to provide a firmer theoretical basis for the interpretation of experimental results, it is shown how the elastic properties of the tissue can be combined with the swim bladder in a consistent way. Although the analysis must eventually be extended to swim bladders of arbitrary shape, for simplicity the essential features are illustrated with a spherical geometry. The results emphasize the need for accurate measurement of the bulk elastic properties of tissue, specifically the complex shear modulus.
Application of the operator expansion method to scattering from one-dimensional moderately rough Dirichlet random surfaces
Authors: Kaczkowski Peter J. , Thorsos Eric I.
Published: October 2005

A new method for computing wave scattering from rough surfaces, called the operator expansion (OE) method, has been proposed by D. M. Milder [J. Acoust. Soc. Am. 89(2), 529–541 (1991)]. In this paper, the OE method is examined in its application to acoustic scattering from one-dimensional randomly rough surfaces with Gaussian and Pierson–Moskowitz roughness spectra satisfying the pressure release (Dirichlet) boundary condition. The operator expansion solution, which is expressed in a systematic series, is found to converge rapidly and monotonically for moderately rough surfaces, that is, for surfaces whose slope-height roughness parameter khs, given by the product of acoustic wave number k, rms surface height h, and rms surface slope s, is less than about 0.25. Through comparison with a numerically exact integral equation solution, the OE method is found to be accurate over a wide range of incident and scattering angles. The method is currently used in a Monte Carlo computation of the scattering cross section, in which scattering is computed from one surface realization at a time and then averaged over 50 realizations. Nevertheless, its efficiency and accuracy in one-dimensional tests suggest that the operator expansion would be a useful method for computing scattering from two-dimensional surfaces in roughness regimes encountered in scattering of low-frequency sound from the ocean surface.
Modeling seafloor geoacoustic interaction with a numerical scattering chamber
Authors: Stephen Ralph A. , Swift Stephen A.
Published: October 2005

A numerical scattering chamber (NSC) has been developed to compute backscatter functions for geologically realistic seafloor models. In the NSC, solutions are computed to the elastic (or anelastic) wave equation by the finite-difference method. This has the following advantages: (a) It includes all rigidity effects in the bottom including body and interface waves. (b) It can be applied to pulse beams at low grazing angles. (c) Both forward scatter and backscatter are included. (d) Multiple interactions between scatterers are included. (e) Arbitrary, range-dependent topography and volume heterogeneity can be treated simultaneously. (f) Problems are scaled to wavelengths and periods so that the results are applicable to a wide range of frequencies. (g) The method considers scattering from structures with length scales on the order of acoustic wavelengths. The process is discussed for two examples: a single facet on a flat, homogeneous seafloor and a canonically rough, homogeneous seafloor. Representing the backscattered field by a single, angle-dependent coefficient is an oversimplification. In a strong scattering environment, time spread of the field is a significant issue and an angle-dependent separation of the wave field may not be valid.
Perceptual compensation for speaker differences and for spectral-envelope distortion
Authors: Watkins Anthony J. , Makin Simon J.
Published: October 2005

This study asks whether perceptual mechanisms that compensate for the spectral-envelope distortion of transmission channels also contribute to compensation for speaker differences. Subjects identified test words that were played after a carrier sentence. In some conditions the carriers were synthesized with F1 in low- and high-frequency ranges and in others they were distorted by filters whose frequency response is the spectral envelope of one vowel minus the spectral envelope of another. The filter /■/ minus /ε/ and its inverse were used. Test words were drawn from an /■tch/ to /εtch/ continuum. Carriers filtered by /■/ minus /ε/ and its inverse give a phoneme boundary difference, indicating compensation for spectral envelope distortion. A phoneme boundary difference also occurs between carriers with F1 in low and high ranges, indicating compensation for speaker differences. Neither of these effects is reduced by playing the carrier backwards, even though a measurement of the perceived naturalness of carriers is sharply reduced by this manipulation. Analysis of carriers synthesized with low and high F1 showed that they have different long-term spectra, and subsequent experiments used time-stationary filters to alter this characteristic. The results showed that the long-term spectra of the carriers govern their influence on the identity of subsequent test sounds. However, measurements of perceptual confusions among the carriers and of perceived talker-differences between carriers revealed that other, time-varying factors are more important for voice identification.
The multidimensional nature of pathologic vocal quality
Authors: Kreiman Jody , Gerratt Bruce R. , Berke Gerald S.
Published: October 2005

Although the terms ‘‘breathy’’ and ‘‘rough’’ are frequently applied to pathological voices, widely accepted definitions are not available and the relationship between these qualities is not understood. To investigate these matters, expert listeners judged the dissimilarity of pathological voices with respect to breathiness and roughness. A second group of listeners rated the voices on unidimensional scales for the same qualities. Multidimensional scaling analyses suggested that breathiness and roughness are related, multidimensional constructs. Unidimensional ratings of both breathiness and roughness were necessary to describe patterns of similarity with respect to either quality. Listeners differed in the relative importance given to different aspects of voice quality, particularly when judging roughness. The presence of roughness in a voice did not appear to influence raters’ judgments of breathiness; however, judgments of roughness were heavily influenced by the degree of breathiness, the particular nature of the influence varying from listener to listener. Differences in how listeners focus their attention on the different aspects of multidimensional perceptual qualities apparently are a significant source of interrater unreliability (noise) in voice quality ratings.
Control of voice F by an artificial neural network
Authors: Farley Glenn R.
Published: October 2005

An artificial neural network model is presented which might correspond to one component of fundamental frequency (F0) control by the brain. Good F0 control could be achieved using only seven neurodes. These included three motor neurodes, a single inhibitory neurode influencing only the thyroarytenoid (TA) motor neurode, and three excitatory neurodes, one of which excited all motor neurodes and two of which excited the inhibitory neurode. The potential utility of this type of model in the study of mechanisms of vocal control is discussed.
Signal-frequency uncertainty in spectral-shape discrimination: Psychometric functions
Authors: Dai Huanping
Published: October 2005

Psychometric functions were measured for spectral-shape discrimination (profile analysis) with the signal frequency either fixed, what we call the signal-known condition, or randomly varied, what we call the signal-unknown condition. The functions were obtained using an adaptive, up–down procedure. In the signal-unknown condition, independent tracks for each signal frequency were interleaved within the same block of trials. The mean slope of the psychometric function [k in log d′=k log ΔL+C, where ΔL=20 log(1+Δp/p)] was 1.13 for the signal-known conditions and 1.19 for the signal-unknown condition. The mean signal-to-standard ratio at threshold (Pc=79.4%) obtained in the signal-unknown condition was 3 to 4 dB higher than that obtained in the signal-known conditions. The psychometric functions of the ideal observer were derived for both the signal-known and signal-unknown conditions. A comparison of the measured and derived psychometric functions suggests that the internal noise component that is statistically independent across frequency channels contributes minimally to the total (internal and external) noise sources that are involved in the decision process.
The sound of a sinusoid: Spectral models
Authors: Patterson Roy D.
Published: October 2005

It is commonly believed that the spectrum of a sound can be used to explain not only the pitch and loudness of that sound but also its timbre, or sound quality. This paper presents an experimental refutation of this spectral timbre claim. Listeners were presented with pairs of ‘‘damped’’ and ‘‘ramped’’ sinusoids and asked to choose the one whose sound quality was most like a sinusoid. The damped sinusoids were constructed by repeating a short segment (25 ms) of a sinusoid with an exponential decay; the ramped sinusoids were constructed by reversing the damped sinusoids in time. The spectra of damped and ramped sinusoids are very simple, consisting of one main peak whose width broadens as the half-life of the decay function decreases. Fourier energy spectra of damped and ramped sounds with the same half-life are identical. Auditory spectra from a more realistic auditory model with a gammatone auditory filterbank show that the width of the spectral peak of the damped sinusoid is narrower than that of the ramped sinusoid for half-lives in the range 2–16 ms and more like the auditory spectrum of an unmodulated sinusoid. Nevertheless, for half-lives in this range, listeners consistently choose the ramped sinusoid as having the stronger sinusoidal quality.
The sound of a sinusoid: Time-interval models
Authors: Patterson Roy D.
Published: October 2005

When an amplitude modulator that is asymmetric in time is applied to a sinusoid, the direction of the asymmetry affects the timbre of the sound. If the modulator is a decaying exponential with a 4-ms half-life and a repetition period of 25 ms, the resulting ‘‘damped’’ sinusoid does not have the sound quality of a sinusoid. But if the modulator is reversed in time to produce a ‘‘ramped’’ sinusoid, the sound of a sinusoid returns [R.D. Patterson, J. Acoust. Soc. Am. (1994)]. This paper extends the domain of the phenomenon with an experiment in which the modulator period varies with the carrier frequency to fix the number of carrier periods per envelope period. This fixes the shape of the wave within a cycle of the sound across carrier frequencies, and it fixes the shape of the main peak in the spectrum of the sound. When presented with pairs of damped and ramped sounds having the same half-life, listeners choose the ramped sound as having the stronger sinusoidal component provided the sinusoids are discriminable. Discrimination is limited to conditions where the half-life is 1 ms or more. At longer half-lives, performance is near ceiling for carrier frequencies up to 1600 Hz; above this, performance falls off as carrier frequency increases. The sound quality differences in damped and ramped sinusoids are explained in terms of a time-interval model of hearing that simulates our auditory images of sounds. The number of carrier periods in the auditory image is greater for the ramped sinusoid in most conditions where the ramped perception is judged to have a stronger sinusoidal quality. Many of these carrier-period intervals are in channels well away from the carrier channel indicating that the sound of a sinusoid is more closely associated with time intervals at the carrier period than with energy in the carrier channel.
The effects of signal frequency and absolute bandwidth on gap detection in noise
Authors: Snell Karen B. , Ison James R. , Frisina D. Robert
Published: October 2005

Whether temporal resolution in noisebursts is primarily determined by the highest frequency component in the signal or its absolute bandwidth remains unclear. In this study, the absolute bandwidths and upper cutoff frequencies of signal noisebursts were varied across broad frequency ranges, several times greater than previously jointly studied. The purpose was to determine how each independently affects detection, taking into consideration that bandwidth effects at one signal frequency might be very different from bandwidth effects at another. Gap detection thresholds were obtained for five subjects with normal hearing in a 2 IFC paradigm. Signals were noisebursts whose bandwidths and upper cutoff frequencies varied among 1, 2, 4, 8, and 12 kHz. Their duration was 150 ms and they were presented at an overall level of 75 dB SPL above a 45 dB SPL white noise floor. The largest mean gap detection threshold, 6.98 ms, was obtained for a noiseburst with a bandwidth of 1 kHz and upper cutoff frequency of 12 kHz. The smallest mean gap detection threshold, 2.22 ms, was found with a bandwidth and upper cutoff frequency of 12 kHz. Significant interactions were found to exist between absolute bandwidth and upper cutoff frequency. Although gap detection thresholds generally decreased with increasing signal frequency and bandwidth, the pattern was complex. When the absolute bandwidth was at least one-half the upper cutoff frequency then upper cutoff frequency and not bandwidth determined gap sensitivity; but when the absolute bandwidth was less than one-half of the upper frequency, then both determined gap thresholds. Aspects of these data support the conclusion that the high frequency components of signals and the corresponding high frequency auditory fibers are critical to temporal resolution but show, in addition, that acuity is enhanced only in relatively wideband signals.
The effects of masking on the growth of vibrotactile sensation magnitude and on the amplitude difference limen: A test of the equal sensation magnitude-equal difference limen hypothesis
Authors: Gescheider G. A. , Bolanowski S. J. , Zwislocki J. J. , Hall K. L. , Mascia C.
Published: October 2005

In this study, the hypothesis that the difference limen (DL) for the detection of differences in amplitude of vibrotactile stimuli is independent of the slope of the sensation magnitude function was tested. The slope of the sensation magnitude function was varied by presenting test stimuli in the presence of or in the absence of vibrotactile noise. The slopes of the sensation magnitude functions were determined through a matching technique in which the subject adjusted stimulus amplitudes of a 250-Hz stimulus presented alone and a 250-Hz stimulus presented simultaneously with a masking noise, so that their sensation magnitudes were equated. The slope of the matching function was found to increase as a function of the intensity of the masking noise. In the second phase of the experiment, the amplitude DL was measured by the gated-pedestal method for test stimuli presented under the same stimulus conditions as used in the matching procedure. At all levels of stimulus intensity, the DL was found to be independent of the masking condition provided the sensation magnitudes of the stimuli were the same. This finding supports the hypothesis that the size of the DL is independent of the slope of the sensation magnitude function, provided the sensation magnitudes of stimuli are the same. The generality of this principle, first discovered in hearing, is thus extended to another sense modality.
Correction of ultrasonic wavefront distortion using backpropagation and a reference waveform method for time-shift compensation
Authors: Liu Dong-Lai , Waag Robert C.
Published: October 2005

A model is introduced to describe ultrasonic pulse amplitude and shape distortion as well as arrival time fluctuation produced by propagation through specimens of human abdominal wall. In the model, amplitude and shape distortion develops as the wavefront propagates in a uniform medium after passing through a phase screen that only causes time shifts. This distortion is compensated by a backpropagation of the wavefront using the angular spectrum method. The compensation employed waveforms emitted by a pointlike source and measured after propagation through the tissue. The waveforms were first corrected for geometric path and then were backpropagated over a sequence of increasing distances. At each distance, a waveform similarity factor was calculated to find the backpropagation distance at which the waveforms were most similar. A new method was devised to estimate pulse arrival time for geometric correction as well as to perform time-shift compensation. The method adaptively derives a reference waveform that is then cross correlated with all the waveforms in the aperture to obtain a surface of arrival times. The surface was smoothed iteratively to remove outlying points due to waveform distortion. The mean (±s.d.) of the waveform similarity factor for 14 specimens was found to be 0.938 (±0.025) initially. After backpropagation of waveforms to the distance of maximum waveform similarity for each specimen, the waveform similarity factor improved to 0.967 (±0.015). The corresponding energy level fluctuation in the wavefront was 4.2 (±0.4) dB initially and became 3.3 (±0.3) dB after backpropagation. For wavefronts focused at 180 mm, the −30 dB mean (±s.d.) effective radius of the focus was 4.2 (±1.2) mm with time-shift compensation in the aperture and became 2.5 (±0.5) mm with backpropagation followed by time-shift compensation. These results indicate that a phase screen placed some distance away from the aperture is an improved model for the description of wavefront distortion produced by human abdominal wall and that wavefront backpropagation followed by time-shift estimation and compensation is an effective method to compensate for such distortion.
Recognition of plosive syllables in noise: Comparison of an auditory model with human performance
Authors: Ainsworth W. A. , Meyer G. F.
Published: October 2005

Some of the effects of noise on the perception of plosive-vowel syllables have been investigated. It was found that noise added to the syllables for the duration of the speech had a more deleterious effect on perception than noise of the same intensity played continuously. Physiological experiments have shown that the response thresholds of cochlear nerve fibers to tones are raised by continuous background noise but not by short bursts of noise. It is suggested that this may be responsible for the speech perception results. In order to investigate this, an auditory model was developed which incorporated response threshold shifts. This was interfaced to a hidden Markov model recognizer and tested with the same sounds that were employed in the human perception experiments. The recognition scores were greater with the threshold shifts than without them.
Detection of a mistuned component in a harmonic complex
Authors: Lee Jungmee , Green David M.
Published: October 2005

The ability to detect a mistuned component in a harmonic complex was measured as a function of harmonic number under three different stimulus procedures: simultaneous monotic, successive monotic, and simultaneous dichotic. In the first experiment, the fundamental frequency was 200 Hz, and the stimulus duration was 410 ms. The threshold obtained in the simultaneous-monotic procedure was lower than those obtained in the other two procedures, especially for harmonics above the third. It is argued that an envelope cue associated with lower-frequency components in the power spectrum of the envelope was the primary cue for the detection of a mistuned component in a harmonic complex in the simultaneous-monotic procedure. Thresholds did not differ for successive-monotic and simultaneous-dichotic procedures, suggesting that the detection of the mistuning was mediated above the level of the cochlea. In the second and third experiments, the stimulus duration and the fundamental frequency of the complex was altered. The envelope cue seemed to be weakened by decreasing the stimulus duration and by increasing the fundamental frequency.
Effects of carrier frequency and background noise on the detection of mixed modulation
Authors: Moore Brian C. J. , Sek Aleksander
Published: October 2005

This article is concerned with the mechanisms underlying the detection of amplitude modulation (AM), frequency modulation (FM), and mixed modulation (MM), i.e., simultaneously occurring AM and FM. In a previous study [B. C. J. Moore and A. Sek, J. Acoust. Soc. Am. 92, 3119–3131 (1992)], psychometric functions were measured for the detection of AM alone and FM alone, using a 10-Hz modulation rate and a 1-kHz carrier frequency. Detectability was then measured for combined AM and FM, with modulation depths selected so that each type of modulation would be equally detectable if presented alone. The detectability of the combined AM and FM was better than would be predicted if the two types of modulation were coded completely independently. Significant effects of relative modulator phase were found when detectability was relatively high, but these effects were not correctly predicted by either of two excitation-pattern models considered. The first experiment reported here was similar to the earlier experiment, but performance was compared for carrier frequencies of 1 and 6 kHz; at the latter frequency, neural synchrony to the stimulus fine structure (phase locking) does not occur. The results at both carrier frequencies were similar to those of our earlier experiment, suggesting that the presence or absence of phase-locking information plays little role in the detection of MM. The second experiment was again similar, but bands of noise were used to mask selectively either the upper or lower side of the excitation pattern of the modulated carrier. The phase effects in this case were in the direction predicted by excitation pattern models. The overall pattern of the results could be predicted reasonably well using a multichannel excitation pattern model based on the assumption that listeners use an unweighted sum of decision variables across all suprathreshold channels with a positive signal-to-noise ratio.
Robust adaptive microphone array processing for hearing aids: Realistic speech enhancement
Authors: Hoffman M. W. , Trine T. D. , Buckley K. M. , Van Tasell D. J.
Published: October 2005

The problem of combining the outputs of an array of microphones as a single input for a hearing aid is investigated. Emphasis is placed on the conservative prediction of realistically achievable performance gains provided by the array over a single microphone. Performance improvement is measured as a change in the speech reception threshold (SRT) between single microphone and multimicrophone conditions. Consistent with previous work, predictions of this change in SRT using intelligibility averaged gain, 〈𝒢〉I, are shown to be good. Consequently, this measure is used, along with changes in signal-to-noise ratios (SNRs), to evaluate array performance. The results presented include the effects of acoustic headshadow, small room reverberation, microphone placement uncertainty, and desired speaker location uncertainty. It is in this context that realistic predictions of speech enhancement provided by robust adaptive microphone array processors are discussed. Performance improvements are demonstrated relative to the ‘‘best’’ single microphone in the array for three types of spatial filters: Fixed, robust block processed, and robust adaptive. The performance of the robust block processed arrays is shown to be attainable with adaptive implementations. One fundamental criterion employed in robust beamformer design directly limits the amount of cancellation of the desired signal that can occur.
Vibrotactile adaptation enhances frequency discrimination
Authors: Goble Alan K. , Hollins Mark
Published: October 2005

Human vibrotactile frequency discrimination (with respect to a 25-Hz standard stimulus, 20 dB above unadapted detection threshold) was measured on the thenar eminence and index fingerpad, using two-interval forced-choice tracking. Measurements were made in the unadapted state and following exposure to 25-Hz adapting stimuli of various amplitudes. The standard and all comparison stimuli were equated for perceived intensity, on the basis of matching experiments that were carried out separately under each adapting condition. Frequency difference thresholds were lowest when the amplitude of the adapting stimulus was equal to the amplitude of the standard. This result complements the earlier finding [] that adaptation sharpens amplitude discrimination of supraliminal stimuli that are similar to the adapting stimulus. Taken together, these discoveries suggest that somatosensory mechanisms that are engaged by extended stimulation serve to enhance detection of changes in the properties, both quantitative and qualitative, of that stimulation.
Electrically evoked otoacoustic emissions from the apical turns of the gerbil cochlea
Authors: Nakajima Hideko H. , Olson Elizabeth S. , Mountain David C. , Hubbard Allyn E.
Published: October 2005

Electrically evoked otoacoustic emissions were measured with current delivered to the second and third turns of the gerbil cochlea. The emission magnitude and phase are dependent on the characteristic frequency (CF) of the stimulating microelectrode location. The death of the animal resulted in an initial increase in emission below the CF of the electrode location and a decrease in emission near the CF of the electrode location. The group delay of the electrically evoked emission phase data is twice as large as the acoustically evoked cochlear microphonic (CM) data obtained by Schmiedt and Zwislocki [J. Acoust. Soc. Am. 61, 133–149 (1977)]. This suggests the possibility of two separate propagation modes for the forward and reverse traveling waves.
The brain-stem auditory-evoked response in the big brown bat (Eptesicus fuscus) to clicks and frequency-modulated sweeps
Authors: Burkard Robert , Moss Cynthia F.
Published: October 2005

Three experiments were performed to evaluate the effects of stimulus level on the brain-stem auditory-evoked response (BAER) in the big brown bat (Eptesicus fuscus), a species that uses frequency-modulated (FM) sonar sounds for echolocation. In experiment 1, the effects of click level on the BAER were investigated. Clicks were presented at levels of 30 to 90 dB pSPL in 10-dB steps. Each animal responded reliably to clicks at levels of 50 dB pSPL and above, showing a BAER containing four peaks in the first 3–4 ms from click onset (waves i–iv). With increasing click level, BAER peak amplitude increased and peak latency decreased. A decrease in the i–iv interval also occurred with increasing click level. In experiment 2, stimuli were 1-ms linear FM sweeps, decreasing in frequency from 100 to 20 kHz. Stimulus levels ranged from 20 to 90 dB pSPL. BAERs to FM sweeps were observed in all animals for levels of 40 dB pSPL and above. These responses were similar to the click-evoked BAER in waveform morphology, with the notable exception of an additional peak observed at the higher levels of FM sweeps. This peak (wave ia) occurred prior to the first wave seen at lower levels (wave ib). As the level of the FM sweep increased, there was a decrease in peak latency and an increase in peak amplitude. Similarity in the magnitude and behavior of the i–iv and ib–iv intervals suggests that wave ib to FM sweeps is the homolog of the wave i response to click stimuli. Experiment 3 tested the hypothesis that wave ia represented activity emanating from more basal cochlear regions than wave ib. FM sweeps (100–20 kHz) were presented at 90 dB pSPL, and broadband noise was raised in level until the BAER was eliminated. This ‘‘masked threshold’’ occurred at 85 dB SPL of noise. At masked threshold, the broadband noise was steeply high-pass filtered at five cutoff frequencies ranging from 20 to 80 kHz. Generally, wave ia was eliminated for masker cutoff frequencies of 56.6 kHz and below, while wave ib was typically observed for masker cutoffs down to 28.3 kHz. The results of these three experiments are compared and contrasted with data from other mammalian BAER studies.
Broadband beamforming for a microphone array
Authors: Sydow Carsten
Published: October 2005

An array of sensors can be used in conjunction with a beamformer, which processes the sensor signals, to achieve a directional response. The beamformer has to be designed such that a beam pattern with certain desired characteristics like specific main beam direction, defined main beam shape, and desired sidelobe level is formed. Conventional methods for the design of beamformers assume sensors with ideal features and do not take the disturbance of the sound field, caused by the mounting of the array, into account. Therefore the predicted theoretical polar response and the measured response often differ significantly. This paper presents a method for optimizing a beamformer for a one-dimensional microphone array, taking into consideration nonideal features of the sensors and the mounting. Thus the actual polar response can be improved. By evaluating cross-correlation functions of the sensor signals during a calibration procedure in an anechoic chamber and minimizing the mean squared error between the beamformer output and a prescribed response, optimum parameters for the beamformer are assigned.
Comments on ‘‘The envelope following response (EFR) in the Mongolian gerbil to sinusoidally amplitude-modulated signals in the presence of simultaneously gated pure tones’’ [J. Acoust. Soc. Am. 94, 3215–3226 (1993)]
Authors: Bernstein Leslie R.
Published: October 2005

Dolphin and Mountain [J. Acoust. Soc. Am. 94, 3215–3226 (1993)] recently reported measurements of the scalp-recorded envelope following response (EFR) in Mongolian gerbils. In their baseline conditions, the EFR was measured when it was elicited by a sinusoidally amplitude-modulated (SAM) tone (the probe) in isolation. These measurements were compared with those obtained when a pure-tone (the ‘‘interfering’’ signal) was presented simultaneously with the SAM probe. Of primary interest to Dolphin and Mountain was the magnitude of the EFR corresponding to the frequency of modulation (i.e., the envelope) of the SAM probe. In general, they observed that the magnitude of the EFR corresponding to the modulating frequency of the probe decreased for ‘‘interfering’’ tones with frequencies close to that of the carrier. Dolphin and Mountain interpreted this effect in terms of ‘‘synchrony capture.’’ This Letter reports an analysis that demonstrates that many of their data can be accounted for by changes that occur within the envelope of the effective stimulating waveform. Therefore, the use of ‘‘synchrony capture’’ as an explanation of the data appears to be unwarranted.
Trade-off between rigidity and damping in plate response
Authors: Sotiropoulou Alexandra G. , Lyon Richard H.
Published: October 2005

The response of a series of plates having the same mass, but differing areas and damping, was measured. The excitation was with an impact hammer applied at a series of locations on the plate(s). The purpose of the experiment was first, to determine the accuracy of a simple energy balance (SEA) prediction of response, and second, to give guidance in a general problem of foundation design. That problem is whether to make a foundation flexible, and therefore allow vibrational energy to be damped, or to make the structure as rigid as possible.
The flush-mounted hydrophone array response to a modified Corcos turbulent wall pressure spectrum (analytical approach)
Authors: Ko Sung H.
Published: October 2005

An analytical development for determination of the frequency spectral density that gives the response of a hydrophone array mounted on a rigid surface to a modified Corcos turbulent wall pressure spectrum is presented. The double integral that represents the frequency spectral density is replaced by the product of two single integrals. The evaluation of the single integral is based on the contour integral (residue) method. Numerically integrated results compare favorably with the results calculated using the exact expression presented here.
A feature-based semivowel recognition system
Authors: Espy-Wilson Carol Y.
Published: October 2005

A recognition system based on linguistic features was developed for the semivowels /w j r l/ in American English. The features of interest are sonorant, syllabic, consonantal, high, back, front, and retroflex. Acoustic correlates and events related to these features were used to detect and classify the semivowels. The recognizer was tested across semivowels occurring in a wide range of phonetic environments. The corpora included polysyllabic words and sentences spoken by males and females of several dialects. The results show that a feature-based approach to recognition is a viable methodology. Fairly consistent overall recognition results were obtained. Across the test data, acoustic events were detected within 97% of the semivowels and classification rates were 62% for /w/, 74% for /l/ (/w/ and /l/ were often confused), 90% for /r/ and 84% for /j/.
Acoustic manifestations of mora timing in Japanese
Authors: Han Mieko S.
Published: October 2005

Segment durations in 20 minimal-paired words in Japanese were examined to identify regular patterns of acoustic manifestations of mora timing in Japanese. The results indicate that segments consistently stretch or compress, providing temporal compensation within a mora and between moras, to attain targeted length at the word level. A segment is 11% longer in the pre-geminate-stop position than in the pre-single-stop position; in the word final position, a segment is 9% shorter following a geminate stop than a segment following a single stop; and [t] in the same phonotactic environment shows varied durations according to different moraic conditions. Fricatives followed by a devoiced vowel are considerably longer than any other inherently long, nonmoraic segments, and are comparable to moraic segments and the average mora duration. Syllable durations are not isochronous, varying by as much as 50%. The word duration is affected more by the number of component sounds and their inherent durations in two-mora than in three-or four-mora words. When the word durations are converted into ratios, they are markedly similar to the mora values. The results suggest a timing control mechanism adjusting segments to obtain targeted word durations. These phenomena describe mora timing in Japanese.
Scattering from an object in a stratified medium. I. Frequency dispersion and active localization
Authors: Yang T. C. , Yates T. W.
Published: October 2005

A signal scattered from an extended object is often very different from a point source signal emitted from the object. Hence active localization based on matched-field/mode processing may not be feasible, depending on how the signal is modified by scattering. In a stratified medium in which propagation of a signal can be expressed in terms of normal modes, the effect of scattering is to induce mode coupling—an incident normal mode is converted to a series of outgoing normal modes by the scatterer. This mode coupling is examined by ensonifying the object with a broadband pulse at some distance from the object. The time-frequency dispersion of a scattered signal is compared with the dispersion of a passive signal traveling the same path (with the scatterer absent). One finds that the coupled normal modes (created by scattering) arrive at a different time than the uncoupled (original) normal modes and can be identified using time-frequency analysis. For active source localization broadband matched-mode processing is applied to the scattered signal. It should be noted that in many problems, the often unknown scattering function (of an unidentified object) prevents one from calculating the replica field for a scattered signal that is required for matched-field processing. For some problems, the relative phase distortion between the modes due to scattering is minimal even though mode coupling is significant. For these cases, matched-mode processing successfully localizes the target; matched-mode processing can tolerate the lack of specific knowledge about the scattering function. Both incoherent and coherent broadband active processing are investigated. In general, one can expect two peaks in the active range and depth ambiguity surface corresponding to the locations of the target and the active source.
Ray-acoustic caustic formation and timing effects from ocean sound-speed relative curvature
Authors: Duda Timothy F. , Bowlin James B.
Published: October 2005

Using deterministic ray-acoustic modeling of 1000-km propagation in the North Pacific, a depth-dependent parameter of ocean sound channels has been found to strongly influence geometrical ray propagation. This parameter is the sound speed times the second vertical derivative of sound speed divided by the square of the first derivative. Ray and wavefront timing and intensity can be influenced within realistic ocean sound channels by unpredictable wavefront triplications and caustics. These triplications are associated with large values of the parameter at ray turning points. The parameter, a relative curvature, behaves as a random variable because of ocean finestructure, causing the unpredictability. The relative curvature has a higher mean value near the sound-speed minimum for both an internal-wave model and actual data, so that this mechanism is a plausible explanation of poor multipath resolution and identifiability late in North Pacific pulse trains.
Localization accuracies for a moving source in an oceanic waveguide
Authors: Fawcett John A. , Maranda Brian H.
Published: October 2005

The passive localization of a narrow-band target in the ocean is a standard procedure in sonar processing. In this paper Cramér–Rao lower bounds are derived for the accuracy with which target track parameters can be estimated using a horizontal array. The target, which may be moving, is observed over a long time period, so that parameters such as bearing rate and radial speed cannot be assumed to be constant. The acoustic propagation in the oceanic waveguide is modeled realistically, and the temporal coherence of the received signal is accounted for. Several computational examples are presented in which the parameter bounds are examined as a function of target position in the waveguide. The dependencies of the bounds on propagation characteristics, observation time, and signal coherence time are also examined. The bounds differ substantially from those computed using simple models that ignore the effect of propagation.
Toward local effective parameter theories using multiresolution decomposition
Authors: Steinberg Ben Zion , McCoy John J.
Published: October 2005

Using the recently developed theory of multiresolution decomposition, a formulation that governs the response of a linear, dynamical system with nonstationary, microscale heterogeneity is reduced to two coupled formulations: one governing the response smoothed on an arbitrary chosen reference scale with the response fine details as forcing, and one governing the response detail with the smoothed response as a forcing. By substituting the solution of the latter in the former, a new framework specifically tuned to macroscale variations of the response, in which the effects of the nonstationary, microscale heterogeneity are described via a macroscale-effective material operator, is obtained. Localization of across-scale couplings and the dependence of the smoothed response and the effective material operator on the microscale and macroscale geometries are investigated asymptotically and numerically. The numerical results are the response of a fluid-loaded elastic plate with nonstationary, microscale mass heterogeneity.
Elastic wave propagation in general transversely isotropic media. I. Green’s functions and elastodynamic holography
Authors: Spies M.
Published: October 2005

The formalism of plane-wave spectral decomposition of elastic wave fields is used to derive a simple method for solving the inverse scattering problem, which can also be regarded as a basis for further imaging techniques. For transversely isotropic materials like fiber composites, but also, e.g., unidirectionally grain-structured austenitic steels, the elastodynamic dyadic and triadic Green’s functions are derived in form of their two-dimensional space-time spectral representations. Based on a theory of plane-wave propagation in these media [], the resulting expressions appear in a coordinate-free form and contain the orientation of the materials’ fiber axis as an additional parameter. Thus the results are particularly useful for extension to the case of layered material. The formulation of Huygens’ principle for a source-free half-space provides the socalled elastodynamic holography, which allows forward-backward propagation of elastic wave fields in form of an integral representation for the displacement vector. This representation is evaluated with respect to space and time via fast Fourier transforms, the effectiveness of the resulting imaging algorithm is demonstrated in comparison with the conventional isotropic algorithm used so far. The integral representation mentioned above is derived for given displacement in a reference plane (specimen surface), the derivation for the case of given surface traction will follow in the second part of this presentation, providing an integral representation for the transducer field.