bluuebunny/crossref_metadata_2025 huggingface dataset:
Some considerations on the use of adaptive methods for estimating interaural-delay thresholds
Authors: Saberi Kourosh
Published: October 2005

This letter makes recommendations on the use of adaptive psychophysics in binaural tasks involving the detection of interaural delays. Specifically, a logarithmic transformation of the stimulus dimension is recommended. The bases for this transform are explained. Monte Carlo simulations and data from human observers show efficient tracking of threshold.
Comments on ‘‘A generalized target strength model for euphausiids, with applications to other zooplankton’’ [J. Acoust. Soc. Am. 95, 2452–2466 (1994)]
Authors: Stanton Timothy K. , Foote Kenneth G.
Published: October 2005

A major premise upon which this paper by Macaulay was written is flawed as is the formulation (both conceptually and mathematically) of the scattering models. The formulation is actually an assemblage and adaptation of previously published approximate solutions rather than a generalization. The physics of the scattering is blurred through a multitude of empirical parameters. In order for this new ‘‘generalized’’ model to fit the data, the animals are required to change shape as the acoustic frequency changes (for any animal of fixed size, it is required to be a sphere at low frequencies, prolate spheroid at intermediate frequencies, and bent cylinder at high frequencies). The scattering models, as presented, are not only physically incorrect, but in two cases are not even dimensionally correct (terms that are dimensionless are added to terms of dimension area). These and other errors are listed in several categories and commented upon with references given.
A note on the use of an approximate formula to predict sound fields above an impedance plane due to a point source
Authors: Stinson Michael R.
Published: October 2005

The application of an approximate equation to describe the propagation of sound from a point source above an impedance plane [e.g., C. F. Chien and W. W. Soroka, J. Sound Vib. 43, 9–20 (1975)] is discussed. In particular, the correct choice of a complex root is specified. This choice is made by ensuring that the sound pressure be a bounded and continuous function of the geometrical parameters and the surface impedance.
Model-based environmental inversion: A shallow water ocean application
Authors: Candy J. V. , Sullivan E. J.
Published: October 2005

A model-based approach to invert or estimate the sound-speed profile (SSP) from noisy pressure-field measurements or more succinctly a solution to the model-based inversion problem is discussed. Recall that model-based signal processing is a well-defined methodology enabling the inclusion of environmental (propagation) models, measurement (sensor arrays) models and noise (shipping, measurement) models into sophisticated processing algorithms. Here the design of a model-based processor (MBP) is investigated based on a Taylor series expansion of the SSP about the most current set of parameter estimates augmented into the state-space representation of a normal-mode propagation model to solve the environmental inversion problem. The resulting processor is adaptive in terms of simultaneously estimating the sound speed, modes, and pressure field at each iteration. Using data obtained from from the well-known Hudson Canyon experiment, a noisy shallow water ocean environment, the processor is designed and the results compared to those predicted using various propagation models and data. It is shown that the on-line, MBP not only predicts the sound speed quite well, but also is able to simultaneously provide enhanced estimates of both modal and pressure-field measurements which are useful for localization and rapid ocean environmental characterization.
Passive localization in ocean acoustics: A model-based approach
Authors: Candy J. V. , Sullivan E. J.
Published: October 2005

A model-based approach is developed to solve the passive localization problem in ocean acoustics using the state-space formulation for the first time. It is shown that the inherent structure of the resulting processor consists of a parameter estimator coupled to a nonlinear optimization scheme. The parameter estimator is designed using the model-based approach in which an ocean acoustic propagation model is used in developing the model-based processor required for localization. Recall that model-based signal processing is a well-defined methodology enabling the inclusion of environmental (propagation) models, measurement (sensor arrays) models, and noise (shipping, measurement) models into a sophisticated processing algorithm. Here the parameter estimator is designed, or more appropriately the model-based identifier (MBID) for a propagation model developed from a shallow water ocean experiment. After simulation, it is then applied to a set of experimental data demonstrating the applicability of this approach.
A stochastic response evaluation using the digital filter supported by method for a double-wall-type sound insulation system excited by the actual music sound
Authors: Miyata Shigeharu , Ohta Mitsuo
Published: October 2005

Recently, for the purpose of reducing a residential environmental noise, many sound insulation systems are often improved acoustically by changing their geometrical scales and/or acoustical characteristics. In this paper, in a direct connection with the stochastic signal information processing along a real physical time, a practical method of identification and probabilistic prediction for insulation systems are theoretically and experimentally proposed in the object-oriented expression forms by functionally introducing a few system parameters. These functional system parameters are supported by many physical structural factors in close relation to the well-known statistical energy analysis (SEA) method. First, a new trial of identification of the above functional system parameters and the output probabilistic prediction for a geometrical change of double-wall-type sound insulation system altered by the above improvement work, especially, under the existence of a strong background noise, is theoretically proposed as one of the wide sense digital filters. Finally, by using the actual music sound as one of the arbitrary random inputs, the effectiveness of the proposed method is confirmed experimentally too by applying it to some problems of predicting a cumulative probability distribution of the transmitted sound level fluctuation.
Estimation of the stress distribution in metal using nonlinear acoustoelasticity
Authors: Kato Masaru , Sato Takuso , Kameyama Keisuke , Ninoyu Hideyuki
Published: October 2005

A new nondestructive method to estimate the stress distribution in a metal is proposed. It is based on the use of nonlinear dependence of ultrasound velocity on stress in metals. First, the dependence of ultrasound velocity on stress in an aluminum alloy is observed, and it is confirmed that the relation is nonlinear and that its derivative changes monotonically in the stress range of interest. Hence, if a certain stress perturbation is given to the observing region in a metal and the change of ultrasound velocity due to the perturbation is measured, the stress in that region can be estimated. This is the basic idea of the measurement of stress in a metal. Now, the field to give the stress perturbation is scanned along the probing ultrasonic beam and a set of data of phase change due to the velocity change is acquired. Then a matrix inversion technique is applied to them by taking account of the spatial distribution of the stress perturbation to obtain a precise stress distribution along the probing beam. The applicability of the proposed method is confirmed by computer simulations. Then a measuring system is constructed which consists of an electromagnetic transducer as the stress perturbation generator which gives approximately 0.4 MPa of perturbation, and a 50-MHz ultrasonic probing wave transmitter/receiver to detect the phase change. Finally, the stress distribution along a line in a specific aluminum alloy block is estimated by using the constructed system. The estimated stress distribution showed good agreement with the expected one.
Neural net-based monitoring of steel beams
Authors: Akerberg Peeter M. , Jansen Ben H. , Finch Robert D.
Published: October 2005

An artificial neural network (ANN) approach was used to characterize vibration data for nondestructive evaluation purposes. Acoustic signatures were obtained from clamped–clamped metal beams of rectangular cross section. The beams were either intact, or had one (small) slot in them. The digitized data were used to train the ANN to predict future samples of the measured time series given past and present samples. The trained ANNs were used in two ways. In the first method, an ANN was trained with vibration data from intact beams. Once the ANN could adequately predict the training signal, vibration signals obtained from beams with slots were presented. Significant differences between prediction errors for the intact beam and beams with slots as shallow as 0.1 in. were found. Furthermore, the resulting prediction errors gradually increased as the slots in the beams grow deeper, suggesting that this method is useful to estimate defect size. In the second method, the connection weights of the ANNs trained on vibration data from intact beams were compared to the corresponding weights of ANNs retrained on the test data. Again, this approach was very sensitive, but no useful relationship with the slot depth was found.
Environmental-correction factors for typical industrial workrooms
Authors: Hodgson Murray , Lewis David N.
Published: October 2005

ISO Standard 3746 presents methods by which the sound-power level of a noise source can be determined from measurements of sound-pressure level made over a survey surface (and vice versa) in arbitrary acoustical environments. Measured levels, which are higher than the required free-field levels because of reflections from the room surfaces, are corrected using an environmental-correction factor. The extent to which measured levels are higher than free field is also affected by source directivity. Here, the ‘‘effective’’ environmental-correction factor, including source directivity effects, is considered. According to the standard, the environmental-correction factor is calculated using the Sabine theory, which assumes a diffuse sound field and low surface absorption. However, the sound field in typical industrial workrooms may not be diffuse, for reasons related to the room shape and the distribution of surface absorption; thus the calculated correction factor may be inaccurate. This was demonstrated, in the case of vertically directional sources, by a series of sound-power- and pressure-level measurements made on a calibrated sound-power source and on two packing machines in workrooms of different dimensions, heights, and ceiling absorptions. The effective correction factor was found to be incorrect by as much as 4 dB. In order to investigate the accuracy of the correction method and, if necessary, obtain more accurate correction factors for typical workroom configurations, ray-tracing predictions, and 1:8-scale-model measurements of the effective environmental-correction factor were made for the cases of omnidirectional and directional sources. Correction factors calculated according to the standard were found to be accurate within 1 or 2 dB in the case of omnidirectional sources, but to be significantly underestimated in the case of vertically directional sources. Tables giving the average effective environmental-correction factors for machines (survey surfaces) of different sizes, in 5- to 10-m-high rooms with omnidirectional or vertically directional sources and without or with ceiling absorption, are presented.
Frequency response of elastic bodies of revolution by the boundary element method
Authors: Kondapalli Prasanna S. , El-Raheb Michael
Published: October 2005

A boundary element method (BEM) is developed to analyze the elastodynamic frequency response of bodies of revolution. The method computes both surface and internal variables. For bilaterally symmetric geometries and boundary conditions, discretization may be limited to one-half of the generator of the body. Quadratic elements are employed for accuracy. Prior to numerical integration by Gaussian quadrature, singularities in the integral equation are removed analytically. To validate the model, results from numerical examples of a solid sphere, a hollow sphere, and a thick disk are compared to analytical and other BEM solutions.
Perturbation theory for travel times
Authors: Snieder Roel , Aldridge David F.
Published: October 2005

In many applications such as travel time tomography, one needs to investigate the effect of perturbations of a slowness model on travel times. In order to carry this out efficiently, a perturbation theory is developed for the travel times along rays under perturbations of the slowness model. The derivation is based on a perturbation analysis of the eikonal equation and leads to simple expressions for the travel time perturbation to arbitrary order. An explicit proof is given that the second-order travel time perturbation, the travel time bias, is equivalent to the second-order travel perturbation that was derived previously. It is also shown that a bending scheme where the slowness model is fixed and where one iteratively updates the travel times can also be based on a perturbation treatment of the eikonal equation.
Refractive fiber optic microphones with ambient acoustic noise-canceling capabilities
Authors: Jost Bradley M. , Stec James P.
Published: October 2005

The design and analysis of refractive fiber optic microphones, including versions capable of ambient acoustic noise cancellation, are presented. Measurements of the transfer characteristics show that this new refractive optical geometry is 3.6 times more sensitive than a reflective geometry. In comparison with a standard (sealed, rigid cavity) refractive fiber optic microphone, a first-order vented cavity noise canceling optical microphone was measured to have an average signal-to-noise ratio improvement of 3.3 dB and a dual diaphragm noise canceling optical microphone was measured to have an average signal-to-noise ratio improvement of 1.7 dB. In addition, the near- and far-field frequency responses of the two noise-canceling optical microphones are provided for comparisons with both the standard refractive fiber optic microphone and a similar electroacoustic microphone.
Fiber-embedded mandrel-stiffened planar acoustic sensor
Authors: Lagakos Nicholas , Bucaro Joseph A. , Ehrenfeuchter Paul E. , Houston Brian H.
Published: October 2005

A plastic mandrel-based, embedded planar fiber-optic acoustic sensor has been developed by wrapping a single mode fiber around solid plastic mandrels which are embedded in an elastomeric encapsulant forming a planar device. The acoustic sensitivity of the sensor was found to be high and frequency independent. The high sensitivity is a result of the effective utilization of the low bulk modulus encapsulant (elastomer), while the frequency-independent sensitivity is a result of the low acceleration sensitivity of the sensor due to the out-of-plane fiber coil geometry and the presence of the mandrels. A comparison is made between this sensor and other large area fiber-optic sensor designs.
Experimental study of a thermoacoustic termination of a traveling-wave tube
Authors: Kordomenos John , Atchley Anthony A. , Raspet Richard , Bass Henry E.
Published: October 2005

Much of the recent experimental work in thermoacoustics has concentrated on standing wave systems. A recent analysis of traveling-wave thermoacoustics [J. Acoust. Soc. Am. 94, 2232–2239 (1993)] prompted an experimental study. The measurements presented herein confirm theoretical predictions that the work done on a traveling wave is thermoacoustic in nature and not due to a Stirling cycle. Experimental response curves are given for two different terminations and compared to theory. The thermoacoustic effect provides a modest change in the reflected and transmitted amplitudes.
Efficient navigation of parameter landscapes
Authors: Collins Michael D. , Fishman Louis
Published: October 2005

Optimization techniques can have difficulty navigating long valleys that are oriented obliquely to the parameter axes. Efficiency may be improved by rotating coordinates so that the most prominent valleys are optimally aligned with the parameter axes. The rotated axes are parallel to the eigenvectors of the covariance matrix of the gradient of the cost function, which may be estimated efficiently by Monte Carlo integration. The information in the covariance matrix may also be used to obtain an improved parametrization, which corresponds to the rotated coordinates associated with the largest eigenvalues. The approach is illustrated for the inverse problem of estimating ocean bottom parameters from acoustic data using simulated annealing. For multifrequency inverse problems, the resolution for each frequency is optimized.
A parabolic equation for poro-elastic media
Authors: Collins Michael D. , Kuperman W. A. , Siegmann William L.
Published: October 2005

The parabolic equation (PE) method is extended to handle problems involving poro-elastic layers [M. A. Biot, ‘‘Theory of propagation of elastic waves in a fluid-saturated porous solid,’’ J. Acoust. Soc. Am. 28, 168–191 (1956)]. The equations of motion are derived for heterogeneous poro-elastic media. Interface conditions that are appropriate for the PE method are derived for coupling to fluid and elastic layers. For the two-dimensional geometry (range and depth) that is representative of many problems in ocean acoustics, the equations of motion reduce to three coupled equations that factor into outgoing and incoming wave equations (the standard formulation has a redundant equation). The outgoing wave equation is solved with the PE method using rational approximations. The poro-elastic PE, which is a generalization of the elastic PE, is an efficient approach for solving range-dependent propagation problems involving an ocean overlying a poro-elastic sediment. The self-starter is generalized to handle compressional and shear sources in poro-elastic layers. The coefficients of the poro-elastic wave equation are derived from a set of natural parameters.
Network analysis of scattering from an internally loaded spherical shell: Resonance identification and frequency- by-frequency limits on the scattered field
Authors: Titterton Paul J.
Published: October 2005

A generalized network model is used to interpret the scattering from a submerged spherical shell with a two-spring/two-mass internal structure. In this model, generally available results are used for the fluid/shell transmission paths, and lumped element methods are used for the internal structure’s admittance. This formulation emphasizes the field quantities at the submerged-shell/internal- structure interface. Of course, the network model has several forms; admittance, impedance, and network scattering are possible (among others). Impedance comparison at the fluid/nominal-shell interface has been previously shown to determine high-Q resonances in the scattered response associated with the lower-branch spherical-harmonic modes. Admittance comparison at the submerged-shell/internal-structure interface shows that the internal structure shifts these lower-branch resonances and broadens their peaks. The sawtooth pattern generated by these shifted and broadened resonances distinguishes the form function with internal structure from the form function without. Finally, network scattering variables and a power-flow constraint are used to determine frequency-by-frequency bounds on the possible acoustic scattering for any passive internal point loading. The scattering with the given lossless, two-spring/two-mass internal structure oscillates between the maximum and minimum possible.
Wave scattering by a two-dimensional pressure-release surface based on a perturbation of the Green’s function
Authors: McSharry P. E. , Moroney D. T. , Cullen P. J.
Published: October 2005

An approximate method is described, to calculate the two-dimensional acoustic scattering, from a slightly rough, time-independent, pressure-release surface. The formulation is based on a perturbation of the Green’s function allowing an approximation of the propagator in the kernel of the Helmholtz integral equation which reduces the integral equation to a convolution equation. The complexity of the problem is reduced from 𝒪(MN2) (for an iterative method where M is the number of iterations) to 𝒪(N ln N), where N is the number of discretizations used to describe the surface numerically. The specific surface examined has a sinusoidal profile, although the method may be applied to any slightly rough surface (hK≪1). We use this solution to generate plane-wave reflection coefficients from the Fourier transform of the pressure gradient on the surface. These may be used to calculate the scattered pressure. The existence of the exact solution for periodic surfaces, due to Holford [J. Acoust. Soc. Am. 70, 1116–1128 (1981)], allows comparison of the convolution approximation with the exact solution. The reflection coefficients and scattered energy are then compared with the approximate solutions of Rayleigh and Kirchhoff.
Reconstruction of depth-dependent flow in a moving half-space using transient acoustic plane waves
Authors: He Sailing , Ström Staffan
Published: October 2005

An acoustic inverse problem for a stratified moving half-space is considered. Acoustic plane waves are obliquely incident on a stratified fluid in a half-space with horizontal flow, and the transient reflection for several incident angles is used as the input information. The time-domain Green’s-function approach based upon wave splitting is applied to reconstruct the depth-dependent velocity of the fluid flow, as well as the sound speed and the density. Partial differential equations for the Green’s functions are given together with the initial and boundary conditions. Numerical results for the reconstruction are presented.
Time and frequency domain characteristics of sperm whale clicks
Authors: Goold John C. , Jones Sarah E.
Published: October 2005

Regular clicks from diving sperm whales, both large bull males and smaller females, were recorded in deep oceanic water off the Azores and subsequently sampled to computer disks for digital analysis. A total of 8540 clicks were marked and analyzed. Simple temporal analysis of the interclick intervals during feeding dives revealed mean click rates for male sperm whales of 1.1713 s−1 and 1.9455 s−1 for females. Fourier analysis showed distinctive peaks in the spectra of bull male sperm whales at 400 Hz and 2 kHz which were stable over extended periods of up to 20 mins. The clicks contained higher frequency components with energy ranging up to at least 12 kHz but not concentrated at any sharply defined frequency. The clicks of smaller female sperm whales showed similar spectral peaks, shifted to 1.2 and 3 kHz, respectively, but these peaks were less pronounced than those in the male click spectra and less stable with time. Higher frequencies were also present up to at least 15 kHz. The previously reported multiple pulse structure of sperm whale clicks is confirmed, but digital filtering reveals this structure to be frequency dependent. Analysis using the short-time Fourier transform confirms the complex time–frequency structure of individual clicks. The frequencies at which the multiples emerge in male and female clicks supports the idea of air cavities in the sperm whale head acting as sound reflectors, although the magnitude of the second pulse at high frequencies suggests some form of off axis distortion. It is also possible that air cavity resonance in the head of the sperm whale may act to reinforce the high-frequency components of the click, and that such components may have superior range and resolution performance in terms of echolocation.
Effect of relative and overall amplitude on perception of voiceless stop consonants by listeners with normal and impaired hearing
Authors: Hedrick Mark S. , Schulte Laura , Jesteadt Walt
Published: October 2005

Previous studies of the /p/–/t/ contrast for normal-hearing listeners have shown that both manipulation of the amplitude of the burst relative to the vowel in the F4–F5 frequency region and overall presentation level can influence the perception of place of articulation [R. N. Ohde and K. N. Stevens, J. Acoust. Soc. Am. 74, 706–714 (1983); Gravel and Ohde, Asha 25, 101 (1983)], such that greater burst amplitude in the high frequencies and higher presentation levels result in more alveolar responses. The influence of relative amplitude and presentation level was tested for both normal-hearing (NH) and hearing-impaired (HI) listeners in the present study. Synthetic CV stimuli were used, and the amplitude of the burst relative to vowel-onset amplitude in the F4–F5 frequency region was manipulated across a 20-dB range. In addition, overall presentation level was varied across a 45-dB range. The findings revealed that the hearing-impaired listeners selected more alveolar responses than listeners with normal hearing when tested at equivalent SPLs. A group of five normal-hearing listeners were then presented the synthetic stimuli in a background of broadband noise at a level that produced thresholds at 4 kHz equivalent to the thresholds of five hearing-impaired listeners. Results from the noise-masked normal-hearing listeners did not consistently show more alveolar responses as presentation level of the stimuli was increased, thus failing to mimic the responses from the hearing-impaired listeners in quiet.
Discrimination of single and complex consonant–vowel- and vowel–consonant-like formant transitions
Authors: van Wieringen Astrid , Pols Louis C. W.
Published: October 2005

Two discrimination experiments were performed to determine auditory sensitivity for single and complex consonant–vowel (CV)-and vowel–consonant (VC)-like formant transitions. In experiment 1 difference limens in end-point frequency were determined by means of same/different paired comparison tasks for 20-, 30-, and 50-ms second formant (F2) speechlike transitions, followed or preceded by an 80-ms vowel-like steady state in initial or final position, respectively. The F2 transition was either single or part of a multiformant (complex) stimulus, also containing a fixed F1 transition with a steady state, a stationary third formant, and a 20-ms voice bar. Just-noticeable differences in end-point frequency decrease with increasing transition duration in all conditions and are smaller for single transitions than for transitions in a multiformant complex. Although difference limens in end-point frequency increase with increase in frequency extent, they are smaller in final than in initial position. As for relative rate-of-frequency change, the smaller the frequency extent, the larger the difference limens of decrementing transitions, possibly because the end points of the transitions approach the frequency of the steady state. Transitions varying in frequency and duration, in such a way that a constant rate of frequency change is maintained (experiment 2), yield considerably smaller difference limens than those varying in frequency extent at a constant transition duration (experiment 1). Discrimination is affected more by changes in duration than in frequency, even when the total duration of the stimulus remains constant. Both experiments show that transition rate is perceptually less important than end-point frequency or transition duration in discriminating short speechlike transitions.
Durations of young children’s word and nonword vocalizations
Authors: Robb Michael P. , Tyler Ann A.
Published: October 2005

The duration of word (W) and nonword (NW) forms was examined in the monthly vocalizations of seven children between 8 and 26 months of age. The W (i.e., glossable ‘‘adultlike’’) forms and NW (i.e., nonglossable but phonetically transcribable) forms occurring in each child’s consonant+vowel (CV) vocalizations were measured. The results were that W durations significantly decreased as a function of increasing chronological age. On the other hand, NW durations were not correlated with increasing age. The meaningfulness corresponding to W forms is hypothesized to account for the results.
Pure tone masking patterns in adult chickens before and after recovery from acoustic trauma
Authors: Saunders Samuel S. , Salvi Richard J.
Published: October 2005

Three female adult chickens were trained to detect pure tones using a positive reinforcement technique and an adaptive threshold tracking procedure. Masked thresholds for tone bursts were measured in the presence of pure-tone maskers at 500, 1000, and 2000 Hz. Masker levels of 40-, 60-, and 70-dB sound-pressure level (SPL) were employed. Masking was maximal at the frequency of the masker. Tone-on-tone masking patterns were symmetrical at low masker levels, but at higher masker levels, more masking was evident at frequencies above the masker than below the masker. Distinct low threshold notches were frequently observed above and below the peak in the masking pattern. After the normal masking patterns had been measured, the chickens were exposed to a 525-Hz traumatizing tone at 120 dB SPL for 48 h and then allowed to recover. After thresholds had returned to their pre-exposure levels, tone-on-tone masking patterns were remeasured. The post-exposure masking patterns were virtually identical to the pre-exposure masking patterns.
The perception of amplified speech by listeners with hearing loss: Acoustic correlates
Authors: Stelmachowicz Patricia G. , Kopun Judy , Mace Anne , Lewis Dawna E. , Nittrouer Susan
Published: October 2005

Audibility-based approaches to hearing-aid selection generally have focused on the long-term average speech spectrum (LTASS). Advances in amplification technology (e.g., multiband signal processing, level-dependent frequency shaping, full dynamic range compression, adaptive compression) make it difficult to predict the audibility of short-term components of speech from the amplified LTASS. This study was designed to quantify the audibility of specific phonemes as processed by two different hearing-aid circuits (linear and full dynamic range compression), and to investigate the relation between audibility and performance on a nonsense syllable recognition task. Data were obtained from three subjects with moderate sensorineural hearing loss. Nine unvoiced consonants were presented in two vowel contexts (/i/ and /open aye/) in both the pre- and post-vocalic position at three intensities. While the performance on selected conditions appeared to vary by hearing-aid type, only one subject showed a statistically significant difference between the two hearing-aid systems. Acoustic analyses revealed a variety of spectral and temporal changes to the speech signal following processing. Estimates of audibility were based upon each subject’s thresholds and an acoustic analysis of the amplified signal that varied across phonemes and consonant position. A signal detection approach was used to predict performance from a simple measure of audibility.
The role of frequency modulation in the perceptual segregation of concurrent vowels
Authors: Culling John F. , Summerfield Quentin
Published: October 2005

Two experiments investigated the effect of frequency modulation on the identification of vowel sounds presented concurrently with interfering vowels. In experiment 1, identification thresholds were measured for each of five target vowels, masked, in each trial, by one of ten masking vowels. Both target and masking vowels were synthesized using inharmonically spaced frequency components. Inharmonic spacing was used in order to prevent powerful grouping processes which exploit fundamental frequency from dominating the results. The target vowels were synthesized with sinusoidal frequency modulation on each frequency component which was either coherent (same phase) or incoherent (random phases). The masking vowels were synthesized with components which were either modulated in the same way as the target vowel or were unmodulated. Identification thresholds were lower when the masking vowel had no modulation. The effect occurred for both coherent and incoherent frequency modulation, indicating that it is mediated by the movement of each component independently, rather than by grouping of coherently modulated components. This result is consistent in some respects with judgments of the prominence of competing vowels [S. E. McAdams, J. Acoust. Soc. Am. 85, 2148–2159 (1989)], which show that modulated vowels are more prominent than unmodulated vowels regardless of the type of modulation applied to the competing vowels.

Experiment 2 used a paradigm similar to that developed by McAdams, in order to compare more directly the effect of FM on vowel identification and vowel prominence. On each trial, three vowels were presented concurrently. Either none, one, or two of the vowels were modulated throughout, while modulation was applied to another vowel (the target) halfway through the stimulus. The vowels were either harmonic (with different fundamental frequencies) and coherently modulated or inharmonic and incoherently modulated. Accuracy of identification of the target vowel was not significantly different in the harmonic/coherent and inharmonic/incoherent conditions and declined, in each case, as the number of modulated background vowels increased. Overall, the results of experiments 1 and 2, and of McAdams’ prominence judgment data, suggest that there is an auditory mechanism for detecting frequency modulation which can alert the listener to the presence of frequency modulated sounds, but which is insensitive to across-frequency differences in the pattern of that modulation.
Amplitude discrimination in masking release paradigms
Authors: Hall Joseph W. , Grose John H.
Published: October 2005

In the first phase of the study, detection thresholds were obtained in conditions that were associated with comodulation masking release (CMR), conditions that were associated with the masking-level difference (MLD), or conditions that were not associated with masking release (baseline conditions). In the second phase of the study, amplitude discrimination was determined for standard tones presented at sensation levels (SLs) 0–30 dB above the masked detection thresholds. Results indicated that, at matched SLs, amplitude discrimination was poorer under conditions of masking release than for baseline conditions, both for the MLD and for CMR. Implications of these results for models of masking release are discussed.
Melody recognition and musical interval perception by deaf subjects stimulated with electrical pulse trains through single cochlear implant electrodes
Authors: Pijl Sipke , Schwarz Dietrich W. F.
Published: October 2005

The perception of musical pitch was investigated in postlinguistically deaf subjects with cochlear implants. Stimuli consisted of sequences of biphasic electrical pulse trains at rates which represented the tones of the equal-tempered musical scale, delivered at equalized comfortable loudness levels to selected single bipolar electrodes along the array of the Nucleus cochlear implant. Seventeen subjects correctly identified a mean of 44% of rhythmically intact familiar tunes, presented in an open-set paradigm. Three subjects were tested with a closed set of melodies without rhythmic cues. The results showed relatively higher recognition scores at lower pulse rates, although melody recognition remained possible up to rates of approximately 600–800 pulses per second. Stimulation of apical electrodes yielded higher recognition scores than of basal electrodes. The perception of musical intervals, defined as frequency ratios between two trains of stimulus pulse rates, was investigated in an interval intonation labeling experiment, for intervals ranging from a minor 3rd to a major 6th. Within a range of low pulse rates, subjects defined the intervals mediated by electrical pulse rate by the same ratios which govern musical intervals of tonal frequencies in normal-hearing listeners. It may be concluded that temporal cues are sufficient for the mediation of musical pitch, at least for the lower half of the range of fundamental frequencies commonly used in music.
The ‘‘inverse problem’’ solved for a three-dimensional model of the cochlea. II. Application to experimental data sets
Authors: de Boer Egbert
Published: October 2005

With ‘‘classical’’ nonactive models of the cochlea it is impossible to simulate the degree of frequency selectivity that is revealed by modern mechanical measurements of the motion of the basilar membrane (BM). Locally active models have been proposed to alleviate this problem, but it remains uncertain whether the actual cochlea is active in this sense. In the first paper of this series [E. de Boer, J. Acoust. Soc. Am. 98, 896–903 (1995)], the ‘‘inverse’’ problem is solved for a (classical) three-dimensional model and a procedure is developed for recovering the BM impedance needed to simulate a given BM response function. It was found that the results of this procedure will be more accurate in the region of the response peak than in the more basal region of the model. In the present paper the same procedure is applied to data of recent mechanical experiments. For the peak region the outcome is unequivocal: Recent measurement results can only be simulated by the classical model when it is made locally active. Resynthesis of the model response, on the basis of the recovered BM impedance, confirms the validity of the method in the peak region and strengthens this conclusion.
Representation of a voice onset time continuum in primary auditory cortex of the cat
Authors: Eggermont Jos J.
Published: October 2005

The representation of voice onset time (VOT) for 197 single units in cat primary auditory cortex was studied for a /ba/–/pa/ continuum in which VOT was varied in 5-ms steps from 0 to 70 ms. The effect of stimulus intensity, characteristic frequency of the neurons, and age of the animals was investigated. The minimum VOT represented in onset responses to both the voiceless and voiced parts of the sound (a ‘‘double-on’’ response) was dependent on overall stimulus level. An interaction was found between the efficacy of the burst in evoking neural activity and the size of the subsequent response to the onset of voicing. There was only a minor difference in the mean values for the minimal neural VOT for young (42 ms), juvenile (36 ms), and adult animals (46 ms), albeit that for individual young and juvenile animals more frequently values close to 10–15 ms were found. The cumulative distribution for the adult group showed a relative lack of neural VOTs around 30–40 ms. No other cues in the single unit and local neuronal group firing rate representation of VOT were found that were related to the categorical perception boundary.
Constant false-alarm rate array detection of random signals in spherically invariant noise
Authors: Izzo Luciano , Tanda Mario
Published: October 2005

A constant false-alarm rate array detector for a random signal embedded in spherically invariant noise is proposed. In the considered observation model, the signal of interest is common to each array element and the noise samples are assumed to be spatially independent across the array, but temporally correlated. The implementation of the detection structure requires the knowledge, but for a scale factor, of the covariance function of the noise at each array element, but is otherwise independent of the noise distribution functions. The performance (detection probability for a fixed false-alarm rate) is assessed by computer simulations and is compared with those of two detectors. The first one is the locally optimum detector for the problem at hand; the second one is the locally optimum detector synthesized on the assumption of correlated Gaussian noise.
Near-field acoustic intensity mapping using a closed surface
Authors: Klein Ken , Guigné Jacques Y.
Published: October 2005

A procedure for constructing near-field acoustic intensity maps is presented in this paper. The normally coincident measurement locations for the X, Y, and Z components about each grid point are repositioned in order to construct an imaginary box centered on the grid point. The position and orientation of these measurements define the surfaces of the box, and are used to estimate the intensity vector at box center. This process is repeated at each grid point in the plane of interest. The technique allows a statistical assessment to be made of the intensity map’s reliability without making assumptions about the nature of the acoustic field.
Correlation distance measurements of the female breast
Authors: Zhu Qing , Steinberg Bernard D. , Arenson Ronald
Published: October 2005

Ultrasonic waves propagating through soft tissue experience wavefront distortion. Adaptive wavefront compensation algorithms attempt to correct such distortion. A valuable design parameter is the isoplanatic patch size of the imaging medium. Its lateral extent is the FWHM of the correlation function. The range extent is defined similarly. The significance of the isoplanatic patch is that a new wavefront correction vector must be obtained whenever the ultrasound beam is moved a patch length. This paper reports measurements of wavefront correlation functions as well as statistics of the lateral correlation distance ρd (half the FWHM) within the female breast obtained from a population of 22 women (44 breasts) and measured at 3 and 4 MHz with a large acoustic aperture (9.6 cm). A set of complex wavefronts radiated from single pointlike sources was measured from the opposing side of the breast. The propagation distance was 12 cm. ρd is the distance between two sources at which the correlation between their wavefronts drops to 0.5. The mean value at 3 MHz was found to be less than 1.5 mm for the premenopausal dense breast, 2.5 mm for the premenopausal fatty breast, and 2.0 mm for the postmenopausal breast. The mean value dropped by a factor of 2 at 4 MHz for a group consisting of premenopausal fatty and postmenopausal breasts.
A test of the hypothesis that cavitation at the focal area of an extracorporeal shock wave lithotripter produces far ultraviolet and soft x-ray emissions
Authors: Vona Daniel F. , Miller Morton W. , Maillie H. D. , Raeman Carol H.
Published: October 2005

Church hypothesized that the violent collapse of microbubbles in water in the focal area of an extra-corporeal shock wave lithotripter (ESWL) can generate biologically damaging far uv and soft x-ray photons. Two techniques were used to test this hypothesis. Gassy water (10 ml) was exposed to ten piezoelectric lithotripter shocks (P+=43 MPa, P−=9 MPa). The resultant sonoluminescence was filtered by optical band-pass filters and measured using a photomultiplier tube (PMT). Next, a commercially available scintillation cocktail (Ecoscint A©), which is formulated to convert high energy photons to visible light, was exposed to varying numbers of lithotripter shocks and the relative luminescence intensity measured and compared to background and distilled water luminescence readings. Results showed support for the hypothesized presence of near uv emissions (∼250 nm) and marginal support for the production of higher energy photons, possibly including far uv and soft x-ray emissions.
Blue and fin whales observed on a seafloor array in the Northeast Pacific
Authors: McDonald Mark A. , Hildebrand John A. , Webb Spahr C.
Published: October 2005

Calling blue and fin whales have been tracked using relative travel times and amplitudes from both direct and multipath arrivals to a seafloor array of seismometers. Calls of three fin whales swimming in the same general direction, but several kilometers apart, are believed to represent communication between the whales because of signature differences in call character, an alternating call pattern, and coordination of call and respiration times. Whale call tracks, call patterns, call character, and swimming speeds were examined during periods with and without the presence of noise. Noise sources included airguns, when the whales were subject to sound levels of up to 143 dB P-P (peak-to-peak) re: 1 μPa over the 10 to 60-Hz band, and transits of merchant ships, when the whales received continuous levels up to 106 dB rms re: 1 μPa over the 10 to 60-Hz band (115 dB P-P). Whale responses associated with these noises remain arguable.
Sensory integration in the bottlenosed dolphin: Immediate recognition of complex shapes across the senses of echolocation and vision
Authors: Pack Adam A. , Herman Louis M.
Published: October 2005

In matching-to-sample tests, a bottlenosed dolphin (Tursiops truncatus) was found capable of immediately recognizing a variety of complexly shaped objects both within the senses of vision or echolocation and, also, across these two senses. The immediacy of recognition indicated that shape information registers directly in the dolphin’s perception of objects through either vision or echolocation, and that these percepts are readily shared or integrated across the senses. Accuracy of intersensory recognition was nearly errorless regardless of whether the sample objects were presented to the echolocation sense and the alternatives to the visual sense (E–V matching) or the reverse, with samples presented to the visual sense and alternatives to the echolocation sense (V–E matching). Furthermore, during V–E matching, the dolphin was equally facile at recognition whether the sample objects exposed to vision were ‘‘live,’’ presented in air in the real world, or were images displayed on a television screen placed behind an underwater window. Overall, the results suggested that what a dolphin ‘‘sees’’ through echolocation is functionally similar to what it sees through vision.
Stimulus intensity and fundamental frequency effects on duplex perception
Authors: Vorperian Houri K. , Ochs Marleen T. , Grantham D. Wesley
Published: October 2005

Duplex perception occurs when part of the acoustic signal is used for both a speech and a nonspeech percept. This phenomenon has been interpreted as evidence of a distinct system for speech perception that precedes other specialized systems of general auditory processing (such as auditory grouping, and perception of pitch, loudness, and timbre). This interpretation was investigated by using an intensity-dependent form of duplex perception with the acoustic pair /dɑ/ and /gɑ/. The ‘‘base’’ portion of the stimulus, common to both, consisted of the first and second formants and the steady-state portion of the third formant (F3). The F3 transition (either a sinusoid or a true formant), which cued the difference between /dɑ/ and /gɑ/, was varied in intensity and fundamental frequency (F0). For every subject, the level at which each type of F3 transition was barely audible in the context of the base, i.e., duplex perception threshold, was first established. Next, identification functions were obtained by varying the intensity of the F3 transition relative to each subject’s duplex perception threshold. Results revealed that duplex perception thresholds decreased as the F0 of the F3 transition increasingly differed from the base. Also, identification functions showed that, as has been previously demonstrated, the F3 transition contributed to the speech percept over a wide range of intensities and fundamental frequencies. However, as F3 transition intensity increased well above duplex perception threshold, /gɑ/ identification decreased. Also, both /dɑ/ and /gɑ/ identification progressively decreased as the F0 of the F3 transition increasingly differed from the base. Contrary to previous duplex perception reports, such findings indicate that both intensity and F0 information is available to the specialized speech perception system. Thus, the computations of the speech perception system and its relation to the general auditory processing systems need to be reexamined.