bluuebunny/crossref_metadata_2025 huggingface dataset:
Multiple diverse ligands binding at a single protein site: A matter of pre‐existing populations
Authors: Ma Buyong , Shatsky Maxim , Wolfson Haim J. , Nussinov Ruth
Published: July 2002

Here, we comment on the steadily increasing body of data showing that proteins with specificity actually bind ligands of diverse shapes, sizes, and composition. Such a phenomenon is not surprising when one considers that binding is a dynamic process with populations in equilibrium and that the shape of the binding site is strongly influenced by the molecular partner. It derives implicitly from the concept of populations. All proteins, specific and nonspecific, exist in ensembles of substates. If the library of ligands in solution is large enough, favorably matching ligands with altered shapes and sizes can be expected to bind, with a redistribution of the protein populations. Point mutations at spatially distant sites may exert large conformational rearrangements and hinge effects, consistent with mutations away from the binding site leading to population shifts and (cross‐)drug resistance. A similar effect is observed in protein superfamilies, in which different sequences with similar topologies display similar large‐scale dynamic motions. The hinges are frequently at analogous sites, yet with different substrate specificity. Similar topologies yield similar conformational isomers, although with different distributions of population times, owing to the change in the conditions, that is, the change in the sequences. In turn, different distributions relate to binding of different sizes and shapes. Hence, the binding site shape and size are defined by the ligand. They are not independent entities of fixed proportions and cannot be analyzed independently of the binding partner. Such a proposition derives from viewing proteins as dynamic distributions, presenting to the incoming ligands a range of binding site shapes. It illustrates how presumably specific binding molecules can bind multiple ligands. In terms of drug design, the ability of a single receptor to recognize many dissimilar ligands shows the need to consider more diverse molecules. It provides a rationale for higher affinity inhibitors that are not derived from substrates at their transition states and indicates flexible docking schemes.
Relation between carbon ion ranges and x‐ray CT numbers
Authors: Jäkel O. , Jacob C. , Schardt D. , Karger C. P. , Hartmann G. H.
Published: July 2002

Measurements of carbon ion ranges in various phantom materials and real bones are presented. Together with measured Hounsfield values, an empirical relation between ranges and Hounsfield units is derived, which is an important prerequisite for treatment planning in carbon ion therapy.
Clinical use of electronic portal imaging: Report of AAPM Radiation Therapy Committee Task Group 58
Authors: Herman Michael G. , Balter James M. , Jaffray David A. , McGee Kiarin P. , Munro Peter , Shalev Shlomo , Van Herk Marcel , Wong John W.
Published: July 2002

AAPM Task Group 58 was created to provide materials to help the medical physicist and colleagues succeed in the clinical implementation of electronic portal imaging devices (EPIDs) in radiation oncology. This complex technology has matured over the past decade and is capable of being integrated into routine practice. However, the difficulties encountered during the specification, installation, and implementation process can be overwhelming. TG58 was charged with providing sufficient information to allow the users to overcome these difficulties and put EPIDs into routine clinical practice. In answering the charge, this report provides; comprehensive information about the physics and technology of currently available EPID systems; a detailed discussion of the steps required for successful clinical implementation, based on accumulated experience; a review of software tools available and clinical use protocols to enhance EPID utilization; and specific quality assurance requirements for initial and continuing clinical use of the systems. Specific recommendations are summarized to assist the reader with successful implementation and continuing use of an EPID.
Extraction of primary signal from EPIDs using only forward convolution
Authors: Hansen Vibeke N. , Swindell William , Evans Philip M.
Published: July 2002

A model is presented in which the scatter signal in images obtained by electronic portal imaging devices (EPIDs) is removed by a forward convolution method. The convolution kernel, is a cylindrically symmetric kernel, generated by Monte Carlo, representing the scattered signal of a pencil beam at the image plane after the photons have gone through an object of thickness, A set of the kernels is presented and used to extract the primary signal. The signal from primary photons in the image, is extracted by an iterative method in which the essential assumption is that the scatter signal can be described by a superposition of the signal that would be obtained with the object removed from the beam, and the kernel, The thickness, that is used to choose the kernel, is directly related to by a simple exponential relationship; hence the thickness, of the object and the primary signal, are both iterated to better estimates through this procedure. The model is tested on Monte Carlo simulated data, where the extracted primary signal is compared with the “true” primary signal. Results are presented for a set of phantoms of uniform thicknesses up to 35 cm, and for field areas up to and for an inhomogeneous phantom containing a sphere of a different density. The primary signal can be extracted to better than 1.5%, even when the original Scatter‐to‐Primary Ratio (SPR) is more than 25%. Finally, we have tested the model on EPID images, a nonuniform (breast) phantom is presented here. The breast phantom both have a curved external contour and contains a structure of a different density (lung). The radiological thickness of this breast phantom, as extracted using the above convolution model, was found to be within 2.8 mm (1 sd) of the true radiological thickness.
Density functional theory: An introduction
Authors: Argaman Nathan , Makov Guy
Published: July 2002

Density functional theory (DFT) is one of the most widely used methods for ab initio calculations of the structure of atoms, molecules, crystals, surfaces, and their interactions. Unfortunately, the customary introduction to DFT is often considered too lengthy to be included in various curricula. An alternative introduction to DFT is presented here, drawing on ideas which are well-known from thermodynamics, especially the idea of switching between different independent variables. The central theme of DFT, i.e., the notion that it is possible and beneficial to replace the dependence on the external potential v(r) by a dependence on the density distribution n(r), is presented as a straightforward generalization of the familiar Legendre transform from the chemical potential μ to the number of particles N. This approach is used here to introduce the Hohenberg–Kohn energy functional and to obtain the corresponding theorems, using classical nonuniform fluids as simple examples. The energy functional for electronic systems is considered next, and the Kohn–Sham equations are derived. The exchange-correlation part of this functional is discussed, including both the local density approximation to it, and its formally exact expression in terms of the exchange-correlation hole. A very brief survey of various applications and extensions is included.
Estimation of structural wave numbers from spatially sparse response measurements
Authors: McDaniel J. Gregory , Shepard W. Steve
Published: July 2002

A method is presented for estimating the complex wave numbers and amplitudes of waves that propagate in damped structures, such as beams, plates, and shells. The analytical basis of the method is a wave field that approximates response measurements in an aperture where no excitations are applied. At each frequency, the method iteratively adjusts wave numbers to best approximate response measurements, using wave numbers at neighboring frequencies as initial estimates in the search. In comparison to existing methods, the method generally requires far fewer measurement locations and does not require evenly spaced locations. The number of locations required by the method scales with the number of waves that propagate in the structure, whereas the number of locations required by existing methods scales with the minimum wavelength. In addition, the method allows convenient inclusion of the analytic relationships between wave numbers that exist for flexural vibrations of beams and plates. Advantages of the method are illustrated by an example in which a beam is excited by a transverse force at one end. Using analytic data and experimental measurements, the method produces a wave field that matches response measurements to within 1 percent. One interesting feature of the new method is that, when applied to analytic data, it supplies more robust wave number estimates using responses at unevenly spaced locations.
The intensity coherence function of time for partially saturated acoustic propagation through ocean internal waves
Authors: Gerber James S. , Flatté Stanley M.
Published: July 2002

The intensity coherence function of time for partially saturated acoustic propagation through internal waves is calculated with a method that is improved over previous treatments. Two specific improvements are introduced: the usual expansion in (1/ΛΦ2) is carried out to a higher order, and then the terms of that expansion are calculated with a new perturbative method. The method is applied to propagation without a sound channel, for both phase-screen and continuous-medium cases. The validity of the new perturbative method is estimated by calculating the next order error terms. Accuracies at the few-percent level are found. The new analytic formulas are also corroborated with numerical integration. Finally, the method is applied to a specific ocean-acoustic experiment [Azores Fixed Acoustic Range (AFAR)]. In order to achieve good agreement with experiment it will be necessary to add an accurate treatment of the sound channel to the present perturbation method.
Using statistical decision theory to predict speech intelligibility. I. Model structure
Authors: Müsch Hannes , Buus So/ren
Published: July 2002

This article introduces a new model that predicts speech intelligibility based on statistical decision theory. This model, which we call the speech recognition sensitivity (SRS) model, aims to predict speech-recognition performance from the long-term average speech spectrum, the masking excitation in the listener’s ear, the linguistic entropy of the speech material, and the number of response alternatives available to the listener. A major difference between the SRS model and other models with similar aims, such as the articulation index, is this model’s ability to account for synergetic and redundant interactions among spectral bands of speech. In the SRS model, linguistic entropy affects intelligibility by modifying the listener’s identification sensitivity to the speech. The effect of the number of response alternatives on the test score is a direct consequence of the model structure. The SRS model also appears to predict the differential effect of linguistic entropy on filter condition and the interaction between linguistic entropy, signal-to-noise ratio, and language proficiency.
Using statistical decision theory to predict speech intelligibility. II. Measurement and prediction of consonant-discrimination performance
Authors: Müsch Hannes , Buus So/ren
Published: July 2002

The speech recognition sensitivity (SRS) model [H. Müsch and S. Buus, J. Acoust. Soc. Am. 109, 2896–2909 (2001)] was tested by applying it to consonant-discrimination data collected in this study. Normally hearing listeners’ abilities to discriminate among 18 consonants were measured in 58 filter conditions using two test paradigms. In one paradigm, listeners chose among all 18 stimuli. In the other, response alternatives were restricted to the correct response and eight consonants that were randomly selected among the 17 incorrect response alternatives. The effect of the number of response alternatives on performance can be described by statistical decision theory. Most filter conditions included one or more sharply filtered narrow bands of speech. Depending on the selection of bands, listeners’ performance in multi-band conditions falls short of, equals, or exceeds the performance expected from multiplication of the error rates in the individual bands. The performance advantage in multi-band conditions increases with average band separation. The SRS model provides a good fit to the data and predicts the data more accurately than does the speech intelligibility index.
Exploring the temporal mechanism involved in the pitch of unresolved harmonics
Authors: Kaernbach Christian , Bering Christian
Published: July 2002

This paper continues a line of research initiated by Kaernbach and Demany [J. Acoust. Soc. Am. 104, 2298–2306 (1998)], who employed filtered click sequences to explore the temporal mechanism involved in the pitch of unresolved harmonics. In a first experiment, the just noticeable difference (jnd) for the fundamental frequency (F0) of high-pass filtered and low-pass masked click trains was measured, with F0 (100 to 250 Hz) and the cut frequency (0.5 to 6 kHz) being varied orthogonally. The data confirm the result of Houtsma and Smurzynski [J. Acoust. Soc. Am. 87, 304–310 (1990)] that a pitch mechanism working on the temporal structure of the signal is responsible for analyzing frequencies higher than ten times the fundamental. Using high-pass filtered click trains, however, the jnd for the temporal analysis is at 1.2% as compared to 2%–3% found in studies using band-pass filtered stimuli. Two further experiments provide evidence that the pitch of this stimulus can convey musical information. A fourth experiment replicates the finding of Kaernbach and Demany on first- and second-order regularities with a cut frequency of 2 kHz and extends the paradigm to binaural aperiodic click sequences. The result suggests that listeners can detect first-order temporal regularities in monaural click streams as well as in binaurally fused click streams.
Similarity, uncertainty, and masking in the identification of nonspeech auditory patterns
Authors: Kidd Gerald , Mason Christine R. , Arbogast Tanya L.
Published: July 2002

This study examined whether increasing the similarity between informational maskers and signals would increase the amount of masking obtained in a nonspeech pattern identification task. The signals were contiguous sequences of pure-tone bursts arranged in six narrow-band spectro-temporal patterns. The informational maskers were sequences of multitone bursts played synchronously with the signal tones. The listener’s task was to identify the patterns in a 1-interval 6-alternative forced-choice procedure. Three types of multitone maskers were generated according to different randomization rules. For the least signal-like informational masker, the components in each multitone burst were chosen at random within the frequency range of 200–6500 Hz, excluding a “protected region” around the signal frequencies. For the intermediate masker, the frequency components in the first burst were chosen quasirandomly, but the components in successive bursts were constrained to fall in narrow frequency bands around the frequencies of the components in the initial burst. Within the narrow bands the frequencies were randomized. This masker was considered to be more similar to the signal patterns because it consisted of a set of narrow-band sequences any one of which might be mistaken for a signal pattern. The most signal-like masker was similar to the intermediate masker in that it consisted of a set of synchronously played narrow-band sequences, but the variation in frequency within each sequence was sinusoidal, completing roughly one period in a sequence. This masker consisted of discernible patterns but not patterns that were part of the set of signals. In addition, masking produced by Gaussian noise bursts—thought to produce primarily peripherally based “energetic masking”—was measured and compared to the informational masking results. For the three informational maskers, more masking was produced by the maskers comprised of narrow-band sequences than for the masker in which the frequencies were not constrained to narrow bands. Also, the slopes of the performance-level functions for the three informational maskers were much shallower than for the Gaussian noise masker or for no masker. The findings provided qualified support for the hypothesis that increasing the similarity between signals and maskers, or parts of the maskers, causes greater informational masking. However, it is also possible that the greater masking was a consequence of increasing the number of perceptual “streams” that had to be evaluated by the listener.
Stop bands for cubic arrays of spherical balloons
Authors: Kushwaha M. S. , Halevi P.
Published: July 2002

The acoustic band structures have been computed for cubic arrays of spherical hydrogen balloons in air. Stop bands are obtained for the face-centered-cubic (fcc) and body-centered-cubic (bcc) lattices, however, there is no band gap for the simple cubic lattice. These gaps are largest for a volume fraction of ∼38% and, at low balloon pressure (1.10 atm) and small latex wall thickness (10−3 times the inner radius), the corresponding gap/midgap ratios are about 0.2 for the fcc and 0.1 for the bcc structure. As the pressure or wall thickness increases, the stop bands diminish and ultimately vanish.
A re-examination of risk estimates from the NIOSH Occupational Noise and Hearing Survey (ONHS)
Authors: Prince Mary M. , Stayner Leslie T. , Smith Randall J. , Gilbert Stephen J.
Published: July 2002

This paper describes a new analysis of data from the 1968–72 National Institute for Occupational Safety & Health (NIOSH) Occupational Noise and Hearing Survey (ONHS). The population consisted of 1172 (792 noise-exposed and 380 “controls”) predominately white male workers from a cross section of industries within the United States. The analysis focused on how risk estimates vary according to various model assumptions, including shape of the dose-response curve and the amount of noise exposure among low-noise exposed workers (or controls). Logistic regression models were used to describe the risk of hearing handicap in relation to age, occupational noise exposure, and duration exposed. Excess risk estimates were generated for several definitions of hearing handicap. Hearing handicap is usually denoted as an average hearing threshold level (HTL) of greater than 25 dB for both ears at selected frequencies. The frequencies included in the biaural averages were (1) the articulation-weighted average over 1–4 kHz, (2) the unweighted average over 0.5, 1, and 2 kHz, and (3) the unweighted average over 1, 2, and 3 kHz. The results show that excess risk estimates for time-weighted average sound levels below 85 dB were sensitive to statistical model form and assumptions regarding the sound level to which the “control” group was exposed. The choice of frequencies used in the hearing handicap definition affected the magnitude of excess risk estimates, which depended on age and duration of exposure. Although data were limited below 85 dB, an age-stratified analysis provided evidence of excess risks at levels ranging from 80 to 84 dB, 85–89 dB, and 90–102 dB. Due to uncertainty in quantifying risks below 85 dB, new data collection efforts should focus on better characterization of dose-response and longitudinal hearing surveys that include workers exposed to 8-hour time-weighted noise levels below 85 dB. Results are compared to excess risk estimates generated using methods given by ANSI S3.44-1996.
Wave-number space response of a near periodically ribbed shell
Authors: Photiadis Douglas M. , Williams Earl G. , Houston Brian H.
Published: July 2002

Employing near-field acoustic holography measurements, the wave-number space response of a ribbed shell to a point drive is investigated. The data show the dominance of flexural vibration in the mid-frequency range, and demonstrate the prominent role played by the eigensolutions of the periodic structure. The supersonic region of frequency–wave-number space is examined in detail and the fast membrane waves, although suppressed in level relative to the subsonic bending waves, provide a clear highlight in this domain at frequencies below the first radiating flexural passband. The results are also examined in kz−kφ space for several frequencies. This representation enables an interpretation of the results in terms of the slowness surface of the flexural waves and provides a simple explanation for the various complex beaming and focusing phenemena which occur in this system.
Sound absorption height and width limits for polymer relaxations
Authors: Hartmann Bruce , Lee Gilbert F. , Lee John D. , Fedderly Jeffry J.
Published: July 2002

An analysis was performed to relate the height and width of the peak in shear sound absorption per wavelength as a function of frequency for the glass transition of a polymer. For the single relaxation time model, the width is 1.14 decades only at small values of the ratio of relaxed to unrelaxed sound speed. For the larger values of sound-speed ratio observed in polymer glass transitions, the width increases as the height increases. For the Havriliak–Negami model, the width increases as the height decreases in a manner similar to that for the complex modulus, though in this case there is a cutoff below the absolute maximum. The curve can be described by the relation that height times width is 1.5 decades of frequency. These predictions are in good agreement with experimental data for 21 polyurethanes.
Musical pitch perception with electrical stimulation of the cochlea
Authors: McDermott Hugh J. , McKay Colette M.
Published: July 2002

Studies were undertaken to investigate the ability of a user of the Nucleus multi-electrode cochlear implant to judge pitch in the context of musical intervals. The subject had qualified as a musical instrument tuner before he received his implant, and was able to judge the intervals between electrical sensations with neither training nor the guidance of familiar melodies. The procedures used were interval estimation, and interval production by the method of adjustment. The pitch of the electrical stimulation was controlled by varying the pulse repetition rate, the active electrode position, or two combinations of these parameters. Further studies employed sinusoidally amplitude modulated pulse trains with varying modulation frequency. The results showed that rate or modulation frequency could convey musical pitch information over a limited range (approximately two octaves). The data were directly comparable with the relationship between musical intervals and frequency for normal hearing. The pitch related to electrode place varied in accordance with the tonotopic organization of the cochlea, and also appeared to be able to support musical intervals. When both place and rate varied together, the place-related pitch was generally dominant. In all cases, the judgement of intervals tended to diverge from their acoustic counterparts as the intervals became larger.
The effect of head rotations on vertical plane sound localization
Authors: Perrett Stephen , Noble William
Published: July 2002

Current understanding gives predominant weight to stationary cues for auditory localization. Two experiments were conducted to investigate the possible existence of a dynamic cue. The first experiment involved localization of concealed sources in the upper median vertical plane (MVP) and showed, as expected, that elevation was not detectable with motionless listening when high-frequency energy was absent or when normal pinna function was distorted. Elevation under such conditions did become detectable with horizontal head rotations, provided low-frequency energy was present in the signal. This indicates that the basis of the dynamic cue is variation in the rate of transformation of low-frequency interaural time/phase differences. The second experiment involved localization of sources arrayed throughout upper and lower regions of the MVP and in the left lateral vertical plane (LVP); it showed that upper hemisphere sources can be distinguished somewhat from those in the lower hemisphere, even in motionless listening conditions, but more so with rotation. The greatest benefit for localization from rotation of the head appears to be gained for sources positioned in the front MVP.
Source track localization via multipath correlation matching
Authors: Westwood Evan K. , Knobles David P.
Published: July 2002

A method for determining the track of a source emitting continuous broadband acoustic energy in an oceanic waveguide is presented. The method involves cross correlating the measured signals at horizontally separated receivers over a period of time and identifying the traces on the resulting correlogram in terms of the ray paths at each receiver that produce them. Range and bearing information is contained in the structure of the multipath correlation traces. Environmental parameters are used as inputs to a ray model to obtain the ray travel times t̄ as functions of range R. The t̄j(R) functions for the ray multipaths are used to obtain simulated correlogram time delays given a source track and receiver geometry. Constant-velocity, constant-depth source tracks are parametrized by four variables, and a nonlinear optimization algorithm is used to find the track that provides the best fit between measured and simulated correlation traces. The method is applied to measurements of a passing surface ship in the shallow waters of the English Channel. Successful localization is obtained using 4-min intervals and correlation traces produced by rays with 3–15 traversals of the water column. Receiver separation, unknown because of experimental uncertainties, is also obtained.
Insertion loss measurements of an acoustical enclosure by using Intensity and MLS methods
Authors: Cobo Pedro , Ranz Carlos , Santiago Salvador , Pons Jose , Siguero Manuel , Delgado Carmen
Published: July 2002

The Insertion Loss (IL) is one of the most appropriate descriptors for the acoustical performance of an enclosure. The IL is conventionally measured by comparing either sound power of space-averaged sound-pressure measurements before and after enclosing the sound source. In this paper, a comparison is made between a power-based IL measurement, using a sound intensity meter, and a pressure-based IL measurement, using the MLS method. Special attention is paid to practical and technical aspects of both methods, such as time consuming, low-frequency limit, signal-to-noise ratio, and background noise immunity. [Work supported by CICYT, Project AMB97-1175-C03-01.]
The level and growth behavior of the 2 f1−f2 distortion product otoacoustic emission and its relationship to auditory sensitivity in normal hearing and cochlear hearing loss
Authors: Kummer Peter , Janssen Thomas , Arnold Wolfgang
Published: July 2002

The 2 f1−f2 distortion product otoacoustic emission (DP) was measured in 20 normal hearing subjects and 15 patients with moderate cochlear hearing loss and compared to the pure-tone hearing threshold, measured with the same probe system at the f2 frequencies. DPs were elicited over a wide primary tone level range between L2=20 and 65 dB SPL. With decreasing L2, the L1−L2 primary tone level difference was continuously increased according to L1=0.4L2+39 dB, to account for differences of the primary tone responses at the f2 place. Above 1.5 kHz, DPs were measurable with that paradigm on average within 10 dB of the average hearing threshold in both subject groups. The growth of the DP was compressive in normal hearing subjects, with strong saturation at moderate primary tone levels. In cases of cochlear impairment, reductions of the DP level were greatest at lowest, but smallest at highest stimulus levels, such that the growth of the DP became linearized. The correlation of the DP level to the hearing threshold was found to depend on the stimulus level. Maximal correlations were found in impaired ears at moderate primary tone levels around L2=45 dB SPL, but at lowest stimulus levels in normal hearing (L2=25 dB SPL). At these levels, 17/20 impaired ears and 14/15 normally hearing ears showed statistically significant correlations. It is concluded that for a clinical application and prediction of the hearing threshold, DPs should be measured not only at high, but also at lower primary tone levels.
Sound production by fishes
Authors: Mann David A.
Published: July 2002

Many fishes produce sounds associated with aggression, courtship, and mating. These sounds are typically low frequency (<1000 Hz), and generally either tonal or broadband pulsed sounds. Sounds are typically produced by vibrating the swimbladder directly with muscles, or through stridulation of bones or pharyngeal jaws. The loudest sounds are produced by fishes that have drumming muscles located on their swimbladder, such as the toadfish (140 dB re: 1μPa). Some fishes like croakers and drums form large spawning aggregations, which can produce loud, sustained sound. Temporal patterns of sound production often follow patterns of courtship and mating, and in many species show an increase in activity at dusk and dawn. While a few fishes have been intensively studied, there is a need for more data on the diversity of fishes that produce sounds. For example, of 59 species of toadfishes, only 4 have been recorded, and of 209 species of cusk eels, only 1 has been recorded. Furthermore, more quantitative data on SPLs and sound propagation are needed to understand the role of the environment on fish sound production. [Work supported by NIH, ONR, and TDT.]
Potential of microperforated panel absorber
Authors: Maa Dah-You
Published: July 2002

Many applications have been found for the microperforated panel (MPP) absorber, on which the perforations are reduced to submillimeter size so that they themselves will provide enough acoustic resistance and also sufficiently low acoustic mass reactance necessary for a wide-band sound absorber. The most important parameter of the MPP is found to be the perforate constant k which is proportional to the ratio of the perforation radius to the viscous boundary layer thickness inside the holes. This, together with the relative (to the characteristic acoustic impedance in air) acoustic resistance r and the frequency f0 of maximum absorption of the MPP absorber, decides the entire structure of the MPP absorber and its frequency characteristics. In other words, the MPP absorber may be designed according to the required absorbing characteristics in terms of the parameters k, r, and f0. Formulas and curves are presented toward this end. It is shown that the MPP absorber has tremendous potential for wide-band absorption up to 3 or 4 octaves and for low-frequency absorption with a cavity of depth small compared to the wavelength. Techniques of making minute holes (of 0.1–0.3 mm, say) have to be developed, though.
Nonmonotonicity of informational masking
Authors: Oh Eunmi L. , Lutfi Robert A.
Published: July 2002

Neff and Green [Percept. Psychophys. 41, 409–415 (1987)] report that the masking of single tones by random-frequency multitone maskers varies nonmonotonically with number of masker components (peaking at 10–50 components). In this paper it is shown that such results are well predicted by a model (the component-relative-entropy model, CoRE) wherein thresholds increase linearly with the ensemble variance of masker spectra smoothed by peripheral auditory filters [R. A. Lutfi, J. Acoust. Soc. Am. 94, 748–758 (1993)]. Three experiments were conducted. In the first, the nonmonotonic relation was replicated for 9 of 11 listeners in conditions similar to those of Neff and Green. In the second, the frequencies of masker components were fixed and the levels of components were varied randomly across presentations to simulate Gaussian noise. In this case, the nonmonotonicity and the total amount of masking for these listeners were shown to be significantly reduced. In the third experiment, masked thresholds for the signal were found to vary monotonically with the frequency spacing of masker components for a fixed number of masker components. Large individual differences among listeners were obtained in some experimental conditions. Individual as well as mean thresholds were well predicted by the CoRE model with an appropriate selection of the values of the two free parameters of the model for each listener.
Nonlinear effect of the inertia of the fluid on acoustic streaming in cylindrical guides
Authors: Menguy Ludovic , Gilbert Jöoel
Published: July 2002

Acoustic streaming is the mean flow created by a high sound level wave. Classical treatment [J. W. S. Rayleigh, Philos. Trans. R. Soc. London 175, 1–21 (1884)] considers that in cylindrical guides a standing acoustic wave creates a symmetrical toroidal vortex (slow streaming) due to the action of Reynolds stress forcing resisted by viscosity. A perturbation method followed by a time averaging applied to the conservation equations leads to the equations describing the behavior of the mean flow velocity in a cylindrical air-filled guide. According to a dimensional analysis, the effect of the fluid inertia cannot be neglected if the streaming becomes perceptible. This effect, previously studied for free or semi-infinite space, is controlled in the case of a waveguide by the dimensionless parameter Renl=M2/Sh2 (M and Sh are the acoustic Mach number and the Shear number). Numerical resolution of this nonlinear system of equations (Newton–Raphson method) is performed using nonslip conditions on the tube wall and mass flux conservation accross the section of the tube. Results indicate that inertia distorts perceptibly the streamlines, and renders vortex pattern unsymmetrical. A comparison of slow and nonlinear acoustic streaming is finally achieved.
A boundary integral formulation for the prediction of acoustic scattering from periodic structures
Authors: Lam Y. W.
Published: July 2002

The scattering of sound by reflecting surfaces is an important feature in room acoustics. The boundary element method has been shown in the past to be an accurate method to calculate the scattering pattern of complex surfaces. However, this numerical method is expensive to use on large surfaces where a large number of elements is required. This paper proposes a method of reducing the number of boundary elements required to model large, but periodic, structures. The method is based on a consideration of the relationship between the field variables at two points exactly one period apart. The problem is reduced to the modeling of just one period of the structure. In cases of plane-wave incidence, the prediction using this method agrees very well with the prediction obtained by modeling the complete structure. With the point source in the near field of the surface, the incidence field is no longer periodic and the assumption of a constant periodic factor does not hold. In this case, it is shown that the method can still be used with a simple modification to the periodic formulation, and produces good agreements with the full structure predictions and with measurements.
Level discrimination of single tones in a multitone complex by normal-hearing and hearing-impaired listeners
Authors: Doherty Karen A. , Lutfi Robert A.
Published: July 2002

A conditional-on-a-single-stimulus (COSS) analysis procedure [, J. Acoust. Soc. Am. 86, 1743–1746 (1989)] was used to estimate how well normal-hearing and hearing-impaired listeners selectively attend to individual spectral components of a broadband signal in a level discrimination task. On each trial, two multitone complexes consisting of six octave frequencies from 250 to 8000 Hz were presented to listeners. The levels of the individual tones were chosen independently and at random on each presentation. The target tone was selected, within a block of trials, as the 250-, 1000-, or 4000-Hz component. On each trial, listeners were asked to indicate which of the two complex sounds contained the higher level target. As a group, normal-hearing listeners exhibited greater selectivity than hearing-impaired listeners to the 250-Hz target, while hearing-impaired listeners showed greater selectivity than normal-hearing listeners to the 4000-Hz target, which is in the region of their hearing loss. Both groups of listeners displayed large variability in their ability to selectively weight the 1000-Hz target. Trial-by-trial analysis showed a decrease in weighting efficiency with increasing frequency for normal-hearing listeners, but a relatively constant weighting efficiency across frequency for hearing-impaired listeners. Interestingly, hearing-impaired listeners selectively weighted the 4000-Hz target, which was in the region of their hearing loss, more efficiently than did the normal-hearing listeners.
A comparison of monotic and dichotic complex-tone pitch perception in listeners with hearing loss
Authors: Arehart Kathryn Hoberg , Burns Edward M.
Published: July 2002

The perception of fundamental pitch for two-harmonic complex tones was examined in musically experienced listeners with cochlear-based high-frequency hearing loss. Performance in a musical interval identification task was measured as a function of the average rank of the lowest harmonic for both monotic and dichotic presentation of the harmonics at 14 dB Sensation Level. Listeners with hearing loss demonstrated excellent musical interval identification at low fundamental frequencies and low harmonic numbers, but abnormally poor identification at higher fundamental frequencies and higher average ranks. The upper frequency limit of performance in the listeners with hearing loss was similar in both monotic and dichotic conditions. These results suggest that something other than frequency resolution per se limits complex-tone pitch perception in listeners with hearing loss.
On the relative influence of individual harmonics on pitch judgment
Authors: Dai Huanping
Published: July 2002

Spectral weighting functions were estimated in a pitch-comparison task to assess the relative influence of individual harmonics on listeners’ pitch judgment. The stimuli were quasi-harmonic complex tones composed of the first 12 components, with fundamental frequencies ranging from 100 to 800 Hz. On each stimulus presentation the frequency of each harmonic was randomly jittered by a small amount. The perceptual weight for each harmonic was calculated as the correlation coefficient between the binary responses of the listener and the frequency jitters for that harmonic. Although in general the present results conform to previous ones showing the predominant role of several low-ranked harmonics, discrepancies exist in details. Contrary to some previous reports that the dominant harmonics were of fixed harmonic ranks regardless of their frequencies, the current results showed that the dominant harmonics were best described as close to a fixed absolute frequency of 600 Hz.
Vibration characteristics of bone conducted sound in vitro
Authors: Stenfelt Stefan , Håkansson Bo , Tjellström Anders
Published: July 2002

A dry skull added with damping material was used to investigate the vibratory pattern of bone conducted sound. Three orthogonal vibration responses of the cochleae were measured, by means of miniature accelerometers, in the frequency range 0.1–10 kHz. The exciter was attached to the temporal, parietal, and frontal bones, one at the time. In the transmission response to the ipsilateral cochlea, a profound low frequency antiresonance (attenuation) was found, verified psycho-acoustically, and shown to yield a distinct lateralization effect. It was also shown that, for the ipsilateral side, the direction of excitation coincides with that of maximum response. At the contralateral cochlea, no such dominating response direction was found for frequencies above the first skull resonance. An overall higher response level was achieved, for the total energy transmission in general and specifically for the direction of excitation, at the ipsilateral cochlea when the transducer was attached to the excitation point closest to the cochlea. The transranial attenuation was found to be frequency dependent, with values from −5 to 10 dB for the energy transmission and −30 to 40 dB for measurements in a single direction, with a tendency toward higher attenuation at the higher frequencies.
A transfer-matrix approach for estimating the characteristic impedance and wave numbers of limp and rigid porous materials
Authors: Song Bryan H. , Bolton J. Stuart
Published: July 2002

A method for evaluating the acoustical properties of homogeneous and isotropic porous materials that may be modeled as fluids having complex properties is described here. To implement the procedure, a conventional, two-microphone standing wave tube was modified to include: a new sample holder; a section downstream of the sample holder that accommodated a second pair of microphone holders and an approximately anechoic termination. Sound-pressure measurements at two upstream and two downstream locations were then used to estimate the two-by-two transfer matrix of porous material samples. The experimental transfer matrix method has been most widely used in the past to measure the acoustical properties of silencer system components. That procedure was made more efficient here by taking advantage of the reciprocal nature of sound transmission through homogeneous and isotropic porous layers. The transfer matrix of a homogeneous and isotropic, rigid or limp porous layer can easily be used to identify the material’s characteristic impedance and wave number, from which other acoustical quantities of interest can be calculated. The procedure has been used to estimate the acoustical properties of a glass fiber material: good agreement was found between the estimated acoustical properties and those predicted by using the formulas of Delany and Bazley.
The case of the missing pitch templates: How harmonic templates emerge in the early auditory system
Authors: Shamma Shihab , Klein David
Published: July 2002

Periodicity pitch is the most salient and important of all pitch percepts. Psychoacoustical models of this percept have long postulated the existence of internalized harmonic templates against which incoming resolved spectra can be compared, and pitch determined according to the best matching templates [J. Goldstein, J. Acoust. Soc. Am. 54, 1496–1516 (1973)]. However, it has been a mystery where and how such harmonic templates can come about. We present here a biologically plausible model for how such templates can form in the early stages of the auditory system. The model demonstrates that any broadband stimulus, including noise and random click trains, suffices for generating the templates, and that there is no need for any delay lines, oscillators, or other neural temporal structures. The model consists of two key stages: cochlear filtering followed by coincidence detection. The cochlear stage provides responses analogous to those recorded in the auditory nerve and cochlear nucleus. Specifically, it performs moderately sharp frequency analysis via a filterbank with tonotopically ordered center frequencies (CFs); the rectified and phase-locked filter responses are further enhanced temporally to resemble the synchronized responses of cells in the cochlear nucleus. The second stage is a matrix of coincidence detectors that compute the average pairwise instantaneous correlation (or product) between responses from all CFs across the channels. Model simulations show that for any broadband stimulus, a degree of high coincidence occurs among cochlear channels that are spaced precisely at harmonic intervals. Accumulating coincidences over time results in the formation of harmonic templates for all fundamental frequencies in the phase-locking frequency range. The model accounts for the critical role played by three subtle but important factors in cochlear function: the nonlinear transformations following the filtering stage, the rapid phase shifts of the traveling wave near its resonance, and the spectral resolution of the cochlear filters. Finally, we discuss the physiological correlates and location of such a process and its resulting templates.
One-hydrophone method of estimating distance and depth of phonating dolphins in shallow water
Authors: Aubauer Roland , Lammers Marc O. , Au Whitlow W. L.
Published: July 2002

Previous attempts at localizing cetaceans have generally used multiple hydrophone arrays and multichannel recording systems. In this paper, a low-budget localization technique using only one hydrophone is described. The time delays of the signals traveling via the surface and bottom reflection paths to the hydrophone, relative to the direct signal, are used to calculate the distance and the depth of a phonating animal. Only two additional measures, the depth of the bottom and hydrophone, have to be taken. The method requires relatively shallow waters and a flat bottom surface. Echolocating and burst pulsing Hawaiian spinner dolphins (Stenella longirostris) at the Waianae coast of Oahu, Hawaii, were localized over different bottom substrates. A tracking range of up to 100 m was achieved. The accuracy of the method is estimated by the total error differential technique. The relative distance estimation error is below 35% and the absolute depth error below 0.7 m, so that the location method is sufficiently precise for examining source levels in our study area. Because of its simplicity, the method ideally complements sound recordings and visual sightings of marine mammals and could lead to a better understanding of the nature and use of click trains by dolphins.
Efficient across-frequency integration: Evidence from psychometric functions
Authors: Hicks Michelle L. , Buus Søren
Published: July 2002

Across-frequency integration of complex signals was investigated by measuring psychometric functions [log (d′) versus signal level in dB SPL] for detection of brief and long signals presented in broadband noise. The signals were tones at 630, 1600, and 4000 Hz, and a nine-tone complex with components spaced at one-third-octave frequencies between 630 and 4000 Hz. The phase relationship of the components in the complex was varied such that adjacent components were in phase (at 0 degrees), 90, or 180 degrees out of phase. Signal durations (defined in terms of the number of cycles between the half-amplitude points of the Gaussian envelopes) of 4.7 and 150 cycles were tested. Results for six normal-hearing listeners showed that the slopes of the psychometric functions were steeper for the brief than for the long signals, and steeper for the tone complexes than for the tones, particularly for the brief signals. This suggests that the transformation from signal intensity to decision variable may be different for brief complex signals than for tonal signals and long complex signals. Thresholds obtained from the psychometric functions were in excellent agreement with those obtained with an adaptive procedure that employed three interleaved tracks. For the long signals, the threshold improvement for the tone complexes relative to a single tone was well described by a 5* log (n) integration rule. However, the threshold improvement for brief signals obeyed a more efficient integration rule of 7 to 8* log (n). A portion of this effect could be accounted for by the phase relationship of the tone complexes; thresholds for brief signals were lowest when the components were in phase at the envelope peak of the signal. This finding indicates that temporal synchrony across auditory channels may enhance detection of brief multi-tone complexes.