bluuebunny/crossref_metadata_2025 huggingface dataset:
Spectral loudness summation for short and long signals as a function of level
Authors: Anweiler Anne-Kristin , Verhey Jesko L.
Published: May 2006

The equal-loudness-ratio hypothesis (ELRH) assumes that the loudness ratio between equal-level long and short signals with the same spectrum is independent of level and spectrum. Thus, it predicts the same amount of spectral loudness summation for equal-level short and long signals. To investigate how duration and level affect spectral loudness summation, loudness functions for 10- and 1000-ms bandpass-filtered noise centered at 2kHz were obtained for normal-hearing listeners using categorical loudness scaling. The bandwidth was varied from 200 to 6400Hz. From the loudness functions the level difference between the 3200-Hz-wide reference signal and the equally loud test signals was determined for reference levels in the range from 25 to 105dB SPL. In addition, for two reference levels (45 and 65dB SPL), this level difference was measured using a loudness-matching procedure. Both procedures showed similar results for the two reference levels. Over a wide level range spectral loudness summation is larger for the short than for the equal-level long signals. This is in contrast to the predictions of the ELRH. However, the data at moderate levels are consistent with the expectations of a modified ELRH which assumes a smaller loudness ratio for broadband than for narrow-band signals.
Enhanced contrast for vowels in utterance focus: A cross-language study
Authors: Hay Jessica F. , Sato Momoko , Coren Amy E. , Moran Cheryl L. , Diehl Randy L.
Published: May 2006

The present study examined several potential distinctiveness-enhancing correlates of vowels produced in utterance focus by talkers of American English, French, and German. These correlates included possible increases in vowel space size, in formant movement within individual vowels, and in duration variance among vowels. Each language group enhanced the distinctiveness of vowels in [+focus] context but used somewhat differing means to achieve this. All three groups used spectral differences, but only German talkers used durational differences, to enhance distinctiveness. The results suggest that the amount of distinctiveness enhancement of a vowel property in [+focus] context is positively related to the between-category variation of that property in [−focus] context. Thus, consistent with the theory of adaptive dispersion, utterance clarity appears to vary directly with information content.
Using self-organizing maps to recognize acoustic units associated with information content in animal vocalizations
Authors: Placer John , Slobodchikoff C. N. , Burns Jason , Placer Jeffrey , Middleton Ryan
Published: May 2006

Kohonen self-organizing neural networks, also called self-organizing maps (SOMs), have been used successfully to recognize human phonemes and in this way to aid in human speech recognition. This paper describes how SOMS also can be used to associate specific information content with animal vocalizations. A SOM was used to identify acoustic units in Gunnison’s prairie dog alarm calls that were vocalized in the presence of three different predator species. Some of these acoustic units and their combinations were found exclusively in the alarm calls associated with a particular predator species and were used to associate predator species information with individual alarm calls. This methodology allowed individual alarm calls to be classified by predator species with an average of 91% accuracy. Furthermore, the topological structure of the SOM used in these experiments provided additional insights about the acoustic units and their combinations that were used to classify the target alarm calls. An important benefit of the methodology developed in this paper is that it could be used to search for groups of sounds associated with information content for any animal whose vocalizations are composed of multiple simultaneous frequency components.
Detailed study of a traveling wave thermoacoustic refrigerator driven by a traveling wave thermoacoustic engine
Authors: Dai Wei , Luo Ercang , Zhang Yong , Ling Hong
Published: May 2006

Thermoacoustic systems have very attractive features and possible wide applications in many areas, especially for cooling purposes. Inventions of traveling wave thermoacoustic engines and traveling wave thermoacoustic refrigerators with work recovery capability have greatly improved the thermodynamic efficiencies of the thermoacoustic systems. To fully utilize the advantages of traveling wave systems, a traveling wave work-recoverable thermoacoustic refrigerator has been designed and built that is driven by a traveling wave thermoacoustic engine, which is aimed at domestic refrigeration purposes. So far, the lowest temperature of −64.4°C and 250W cooling power at −22.1°C are obtained by the system with 3.0MPa helium gas and 57.7Hz working frequency. Heat input into the system is 2.2kW. Simulations based on linear thermoacoustic theory have also been done for comparison with experimental results, which shows reasonable agreement within a certain pressure wave amplitude range and cold end temperature range. The comparisons meanwhile imply some serious loss mechanism inside the traveling wave thermoacoustic refrigerator.
Directional loudness in an anechoic sound field, head-related transfer functions, and binaural summation
Authors: Sivonen Ville Pekka , Ellermeier Wolfgang
Published: May 2006

The effect of sound incidence angle on loudness was investigated using real sound sources positioned in an anechoic chamber. Eight normal-hearing listeners produced loudness matches between a frontal reference location and seven sources placed at other directions, both in the horizontal and median planes. Matches were obtained via a two-interval, adaptive forced-choice (2AFC) procedure for three center frequencies (0.4, 1, and 5kHz) and two overall levels (45 and 65dB SPL). The results showed that loudness is not constant over sound incidence angles, with directional sensitivity varying over a range of up to 10dB, exhibiting considerable frequency dependence, but only minor effects of overall level. The pattern of results varied substantially between subjects, but was largely accounted for by variations in individual head-related transfer functions. Modeling of binaural loudness based on the at-ear signals favored a sound-power summation model, according to which the maximum binaural gain is only 3dB, over competing models based on larger gains, or on the summation of monaural loudness indices.
Transient response of a hollow cylinder with radial and axial material inhomogeneity
Authors: El-Raheb Michael
Published: May 2006

The transient response of a hollow finite cylinder with radial and axial material inhomogeneity to a pulse of short duration is analyzed. The problem is solved in two steps. Radial inhomogeneity is treated first by dividing the cylinder into coaxial segments each with a constant modulus E which is allowed to vary from segment to segment. Transfer matrices relating variables at the two radial interfaces of a segment combine to satisfy continuity of stress and displacement at these interfaces. The Galerkin method is then utilized to treat the general case with both radial and axial inhomogeneity adopting the eigenfunctions of the radially inhomogeneous cylinder as trial functions. Features of both static and transient responses resemble that for a weakening material along the axis: displacement increases and stress reduces in proportion to the reduction in modulus. Spatial attenuation of transient stress cannot be replicated when geometric dimensionality is reduced from 3-D axisymmetric to 2-D plane-strain.
Early learners’ discrimination of second-language vowels
Authors: Højen Anders , Flege James E.
Published: May 2006

It is uncertain from previous research to what extent the perceptual system retains plasticity after attunement to the native language (L1) sound system. This study evaluated second-language (L2) vowel discrimination by individuals who began learning the L2 as children (“early learners”). Experiment 1 identified procedures that lowered discrimination scores for foreign vowel contrasts in an AXB test (with three physically different stimuli per trial, where “X” was drawn from the same vowel category as “A” or “B”). Experiment 2 examined the AXB discrimination of English vowels by native Spanish early learners and monolingual speakers of Spanish and English (20 per group) at interstimulus intervals (ISIs) of 1000 and 0ms. The Spanish monolinguals obtained near-chance scores for three difficult vowel contrasts, presumably because they did not perceive the vowels as distinct phonemes and because the experimental design hindered low-level encoding strategies. Like the English monolinguals, the early learners obtained high scores, indicating they had shown considerable perceptual learning. However, statistically significant differences between early learners and English monolinguals for two of three difficult contrasts at the 0-ms ISI suggested that their underlying perceptual systems were not identical. Implications for claims regarding perceptual plasticity following L1 attunement are discussed.
Accuracy of an acoustic location system for monitoring the position of duetting songbirds in tropical forest
Authors: Mennill Daniel J. , Burt John M. , Fristrup Kurt M. , Vehrencamp Sandra L.
Published: May 2006

A field test was conducted on the accuracy of an eight-microphone acoustic location system designed to triangulate the position of duetting rufous-and-white wrens (Thryothorus rufalbus) in Costa Rica’s humid evergreen forest. Eight microphones were set up in the breeding territories of 20 pairs of wrens, with an average intermicrophone distance of 75.2±2.6m. The array of microphones was used to record antiphonal duets broadcast through stereo loudspeakers. The positions of the loudspeakers were then estimated by evaluating the delay with which the eight microphones recorded the broadcast sounds. Position estimates were compared to coordinates surveyed with a global-positioning system (GPS). The acoustic location system estimated the position of loudspeakers with an error of 2.82±0.26m and calculated the distance between the “male” and “female” loudspeakers with an error of 2.12±0.42m. Given the large range of distances between duetting birds, this relatively low level of error demonstrates that the acoustic location system is a useful tool for studying avian duets. Location error was influenced partly by the difficulties inherent in collecting high accuracy GPS coordinates of microphone positions underneath a lush tropical canopy and partly by the complicating influence of irregular topography and thick vegetation on sound transmission.
Iterative simulation of elastic wave scattering in arbitrary dispersions of spherical particles
Authors: Doyle Timothy E.
Published: May 2006

A numerical modeling approach was developed to simulate the propagation of shear and longitudinal waves in arbitrary, dense dispersions of spherical particles. The scattering interactions were modeled with vector multipole functions and boundary condition solutions for each particle. Multiple scattering was simulated by translating the scattered wave fields from one particle to another with the use of translational addition theorems, summing the multiple-scattering contributions, and recalculating the scattering using an iterative method. The approach can simulate 3D dispersions with a variety of particle sizes, compositions, and volume fractions. To test the model, spectra and wave field images were generated from ordered and disordered microstructures containing up to several thousand particles. The model predicted wave propagation phenomena such as refractive focusing and mode conversion. The iterations converged for many particle configurations, but did not converge or only partially converged for certain conditions, specifically large particle dispersions (>100 particles) at short wavelengths (λ∼particle diameter). Incorporating viscoelastic damping into the matrix properties reduced these numerical instabilities. The model is currently constrained by these convergence limitations and by the computation of sufficiently high multipole order for large numbers of particles. The theory and initial results for the model are presented.
Acoustic structures in the alarm calls of Gunnison’s prairie dogs
Authors: Slobodchikoff C. N. , Placer J.
Published: May 2006

Acoustic structures of sound in Gunnison’s prairie dog alarm calls are described, showing how these acoustic structures may encode information about three different predator species (red-tailed hawk—Buteo jamaicensis; domestic dog—Canis familaris; and coyote—Canis latrans). By dividing each alarm call into 25 equal-sized partitions and using resonant frequencies within each partition, commonly occurring acoustic structures were identified as components of alarm calls for the three predators. Although most of the acoustic structures appeared in alarm calls elicited by all three predator species, the frequency of occurrence of these acoustic structures varied among the alarm calls for the different predators, suggesting that these structures encode identifying information for each of the predators. A classification analysis of alarm calls elicited by each of the three predators showed that acoustic structures could correctly classify 67% of the calls elicited by domestic dogs, 73% of the calls elicited by coyotes, and 99% of the calls elicited by red-tailed hawks. The different distributions of acoustic structures associated with alarm calls for the three predator species suggest a duality of function, one of the design elements of language listed by Hockett [in Animal Sounds and Communication, edited by W. E. Lanyon and W. N. Tavolga (American Institute of Biological Sciences, Washington, DC, 1960), pp. 392–430].
Three-dimensional simulation of electron-acoustic detections
Authors: Gao Chun-ming , Zhang Shu-yi , Zhang Zhong-ning , Shui Xiu-ji
Published: May 2006

In electron-acoustic (EA) detection, a periodically modulated electron beam is impinging on a sample to excite thermo-acoustic signals, and a PZT transducer bonded on the back surface of the sample is used to detect the thermo-acoustic signals. Based on the conditions of the EA detection experiment, theoretical analyses of 3-D temperature distributions in the sample, 3-D elastic fields in the sample and piezoelectric transducer, and piezoelectric output signals of the transducer are given. Generally, the vibration modes and the electrical output signals of the transducer are hard to calculate because of the complicated structures, i.e., complicated boundary conditions, thus a finite element method (FEM) is used to simulate the electron-acoustic signals. According to the corresponding FEM model, the frequency spectra of the detected EA signals can be calculated. Meanwhile, the related experimental measurements for several homogeneous samples are carried out. Comparing the theoretical simulated results with those of the experiments, it shows that both are in good agreement with each other. Finally, using this FEM model, the dependences of the electron-acoustic signals on the thermal and elastic properties of several samples are investigated.
The influence of circumferential edge constraint on the acoustical properties of open-cell polyurethane foam samples
Authors: Tsay Huoy-Shyi , Yeh Fung-Huei
Published: May 2006

In this paper, a three-dimensional Laplace domain finite element model with four-node tetrahedral porous elements is developed and applied for the investigations. Without the incident acoustic field, the exact acoustical properties of a laterally infinite planar foam, which is backed with a rigid plane, are accurately predicted by assuming the circumferential edge constraints as rollers. It is found that the rigidity of the circumferential edge constraint has a profound effect on the predictions. If the circumferential edge constraints are all fixed, the amplitude of the first sound absorption peak is increased, and the peak is moved to a lower frequency as compared with the exact solutions. Measurements obtained by the two-microphone impedance tube method are bounded by the predictions of samples with roller as well as with fixed constraints. They are much closer to that predicted with fixed edge constraints. Upon increasing the diameter-to-thickness ratio of the sample, the influence of the fixed edge constraint is reduced. The measured acoustic results can be the representatives of the laterally infinite planar foam from which the sample was cut only if the roller edge constraint of the sample is managed or the diameter-to-thickness ratio of the sample is increased.
An experimental analysis of the pressures and flows within a driven mechanical model of phonation
Authors: Kucinschi Bogdan R. , Scherer Ronald C. , DeWitt Kenneth J. , Ng Terry T. M.
Published: May 2006

The production of voice is related to the flow of air through the glottis, whose time-dependent shape is defined by the motion of the vocal folds and the translaryngeal pressure. A scaled dynamically similar experimental apparatus that mimics the motion of the vocal folds was designed and built, such that both the glottal diameter and glottal angle change during a motion cycle. This motion is more realistic than in other reported dynamic models. The motion of the folds can be driven at different frequencies. The glottal flow takes place at a constant inlet pressure, mimicking the lung pressure. The transglottal pressure difference and flow rate were measured over the motion cycle. Satisfactory agreement was obtained for identical cases by numerically solving the two-dimensional, incompressible Navier-Stokes equations. Both experimental and numerical data showed that the glottal flow rate and transglottal pressure were affected by the oscillation frequency of the vocal folds. Flow visualization showed that the glottal flow patterns, which are a potential source of aero-acoustic sound, are influenced by the oscillation frequency. However, glottal flow resistance depended to a lesser extent on vocal fold oscillation frequency for the portion of the cycle when the glottis was divergent.
Broadband sound reflection by plates covering side-branch cavities in a duct
Authors: Huang Lixi
Published: May 2006

When a segment of a rigid duct is replaced by a plate backed by a hard-walled cavity, grazing incident sound waves induce plate bending, hence sound reflection. The mechanism is similar to the drumlike silencer with tensioned membranes [L. Huang, J. Acoust. Soc. Am. 112, 2014–2025 (2002)]. However, the logarithmic bandwidth over which the reflection occurs is much wider than that of a drumlike silencer of the same cavity geometry, the typical difference being nearly one octave band. The difference in the silencing performance is explained in terms of the intermodal acoustic interference between the odd and even in vacuo vibration modes. For a given cavity volume, the widest stopband for noise in air is obtained by using long plates with two free lateral edges parallel with the duct axis. The optimal material should be stiff and light, and the critical parameter is found to be the ratio of the Young’s modulus over the cube of density. Typically, this ratio is 250 times higher than those of common metallic materials like aluminum alloys, but it is within the reach of existing ultralight foam materials or composite beams with a light core.
Adaptive wave field synthesis with independent radiation mode control for active sound field reproduction: Theory
Authors: Gauthier Philippe-Aubert , Berry Alain
Published: May 2006

Sound field reproduction finds applications in music or audio reproduction and experimental acoustics. For audio applications, sound field reproduction can be used to artificially reproduce the spatial character of natural hearing. The general objective is then to reproduce a sound field in a real reproduction environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. For classical WFS, the room response thus reduces the quality of the physical sound field reproduction. In this paper, adaptive wave field synthesis (AWFS) is analytically investigated as an adaptive sound field reproduction system combining WFS and active control with a limited number of reproduction error sensors to compensate the response of the listening environment. The primary point of this paper is the definition of AWFS. Therefore, the fundamental behavior of AWFS is illustrated by analytical considerations and simple free-field simulation results. As demonstrated, AWFS is fundamentally related to WFS and “Ambisonics.” The paper introduces independent adaptive control of sound field reproduction on the basis of radiation modes, via the singular value decomposition of the transfer impedance matrix. Possible practical independent control of radiation modes for AWFS is discussed.
Evaluation of a boss model and subtraction technique for predicting wideband scattering phenomena in room acoustics
Authors: Natsiopoulos Georgios , Kleiner Mendel
Published: May 2006

Scattering from a hard hemisphere on an infinite plane is studied experimentally and theoretically in a laboratory environment using a subtraction technique. With this technique, the scattered field is determined by subtracting the transient responses from the plane with and without the boss present. Scattering for different combinations of transducer and hemisphere positions are then classified into a number of cases, denoted in-plane/off-plane and back-/forward/mixed scattering. Theoretical and experimental results are compared and evaluated from an auralization point of view using a one-third octave band smoothing of the contribution from the hemisphere. With the smoothing used, average intensity level discrepancies typically less than 3dB are obtained for a 6 octaves wide ka-interval, where a equals the hemisphere radius.
Measurements of the anisotropy of ultrasonic attenuation in freshly excised myocardium
Authors: Baldwin Steven L. , Marutyan Karen R. , Yang Min , Wallace Kirk D. , Holland Mark R. , Miller James G.
Published: May 2006

Echocardiography requires imaging of the heart with sound propagating at varying angles relative to the predominant direction of the myofibers. The degree of anisotropy of attenuation can significantly influence ultrasonic imaging and tissue characterization measurements in vivo. This study quantifies the anisotropy of attenuation of freshly excised myocardium at frequencies typical of echocardiographic imaging. Results show a significantly larger anisotropy than previously reported in specimens of locally unidirectional myofibers. Through-transmission radio frequency-based measurements were performed on specimens from 12 ovine and 12 bovine hearts. Although ovine hearts are closer in size to human, the larger size of bovine hearts offers the potential for specimens in which myofibers are more nearly unidirectionally aligned. The attenuation coefficient increased approximately linearly with frequency. The mean slope of attenuation with frequency was 3–4 times larger for propagation parallel than for perpendicular to the myofibers. At perpendicular insonification, slopes between ovine and bovine myocardium were approximately equal. However, attenuation in bovine specimens was larger for angles approaching parallel. The difference in results for parallel appears consistent with what might be expected from increased myofiber curvature associated with smaller lamb hearts. Quantitative knowledge of anisotropy of attenuation may be useful in understanding mechanisms underlying the interaction of ultrasound with myocardium.
Acoustic impedance of perforations in contact with fibrous material
Authors: Lee Iljae , Selamet Ahmet , Huff Norman T.
Published: May 2006

Acoustic impedance of perforations in contact with fibrous material, as well as air alone, is experimentally determined in the absence of mean flow. Different porosities (2.1, 8.4, 13.6, and 25.2%) and hole diameters (0.249 and 0.498cm) are applied for perforations, along with two fiber filling densities (100 and 200kg∕m3) to illustrate the effect of variations in such parameters on the perforation impedance. A modified impedance tube setup is developed for the measurement of the acoustic impedance of perforations in contact with fibrous material, and the air alone. The complex wave-number and characteristic impedance of the fibrous material are also experimentally obtained in this study to utilize such properties for the calculation of acoustic impedance of perforations in contact with the material. The experimental results show that both resistance and reactance of the perforations in contact with air decrease as the porosity increases. It is also shown that the fibrous material significantly increases both the resistance and reactance. For a given filling density, both resistance and reactance decrease as the porosity increases, except for the specific samples with highest porosity (25.2%).
Sources of excess noise in silicon piezoresistive microphones
Authors: Dieme Robert , Bosman Gijs , Nishida Toshikazu , Sheplak Mark
Published: May 2006

This paper provides evidence for electrical 1∕f noise as the dominant source of excess noise in piezoresistive microelectromechanical systems (MEMS) microphones. In piezoresistors, the fundamental noise sources may be divided into frequency independent thermal noise and frequency dependent 1∕f excess noise dominating at low frequencies. Noise power spectra are presented for both commercial and research-prototype MEMS piezoresistive microphones as a function of applied voltage bias for both free and blocked membranes. The contributions of various mechanical and electrical noise sources are compared using a lumped noise equivalent circuit of the piezoresistive microphone. The bias dependence and membrane independence of the output noise indicate that the primary source of the excess noise is electrical in origin.
Passive imaging of underground acoustic sources
Authors: Norton Stephen J. , Carr Bradley J. , Witten Alan J.
Published: May 2006

The problem of imaging the intensity distribution of a subsurface acoustic source from passive measurements is considered. The source may be spatially extended, and is assumed to be random and spatially incoherent. Two passive imaging algorithms are examined and shown to be closely related. The first algorithm is based on time-domain migration in which signals recorded by an array of receivers are backpropagated to multiple source locations. The second algorithm is based on backprojecting cross-correlating signals computed between all pairs of receiving elements. When the latter algorithm is applied in two dimensions, the correlation peak can be shown to be the line-integral of the source intensity along a hyperbolic path whose foci are the receiving points. This suggests a filtered-backprojection procedure in which the backprojection is performed along multiple hyperbolic paths. The algorithm is easily generalized to three dimensions, in which the cross-correlated signals are backprojected over hyperboloids. Images using simulated and field data are presented as an illustration.
Speech perception in gated noise: The effects of temporal resolution
Authors: Jin Su-Hyun , Nelson Peggy B.
Published: May 2006

Previous investigations have suggested that hearing-impaired (HI) listeners have reduced masking release (MR) compared to normal hearing listeners (NH) when they listen in modulated noise. The current study examined the following questions that have not been clearly answered: First, when HI listeners are amplified so that their performance is equal to that of NH listeners in quiet and in steady noise, do HI listeners still show reduced MR with modulated noise when compared to NH listeners? Second, is the masking release the same for sentences and CV syllables? Third, does forward masking significantly contribute to the variability in performance among HI listeners? To compensate for reduced hearing sensitivity for HI listeners, the spectrum levels of both speech and noise were adjusted based on the individual hearing loss. There was no significant difference between the performance of NH listeners and that of HI listeners in steady noise and in quiet. However, the amount of MR for sentences and for CV syllables was significantly reduced for HI listeners. For sentence recognition, the amount of MR seemed to be more related to hearing sensitivity for low-to-mid frequencies than to forward masking. In contrast, forward masking thresholds appear to be a major contributor to the amount of MR for syllable recognition.
An active headrest for personal audio
Authors: Elliott Stephen J. , Jones Matthew
Published: May 2006

There is an increasing need for personal audio systems, which generate sounds that are clearly audible to one listener but are not audible to other listeners nearby. Of particular interest in this paper are listeners sitting in adjacent seats in aircraft or land vehicles. Although personal audio could then be achieved with headsets, it would be safer and more comfortable if loudspeakers in the seat headrests could be actively controlled to generate an acceptable level of acoustic isolation. In this paper a number of approaches to this problem are investigated, but the most successful involves a pair of loudspeakers on one side of the headrest, driven together to reproduce an audio signal for a listener in that seat and also to attenuate the pressures in the adjacent seat. The performance of this technique is investigated using simple analytic models and also with a practical implementation, tested in an anechoic chamber and a small room. It is found that significant attenuations, of between 5 and 25 dB, can be achieved in the crosstalk between the seats for frequencies up to about 2kHz.
Measuring the free field acoustic impedance and absorption coefficient of sound absorbing materials with a combined particle velocity-pressure sensor
Authors: Lanoye R. , Vermeir G. , Lauriks W. , Kruse R. , Mellert V.
Published: May 2006

Acoustic surface impedance of sound absorbing materials can be measured by several techniques such as the impedance tube for normal impedance or the Tamura method for normal and oblique surface impedance. In situ, the acoustic impedance is mostly measured by use of impulse methods or by applying two-microphone techniques. All these techniques are based on the determination of the sound pressure at specific locations. In this paper, the authors use a method which is based on the combined measurement of the instantaneous sound pressure and sound particle velocity. A brief description of the measurement technique and a detailed analysis of the influence of the calibration, the source type, the source height, the sound incidence angle, and the sample size are included.
Paranasal sinus ventilation by humming
Authors: Granqvist Svante , Sundberg Johan , Lundberg Jon O. , Weitzberg Eddie
Published: May 2006

The gas nitric oxide (NO) is generated in the human paranasal sinuses and can be measured in nasally exhaled air. During humming, a marked increase in exhaled NO content has been observed. The acoustic phenomenon responsible for this evacuation of NO gas from the sinuses was analyzed. A tube model was constructed with a syringe containing NO gas attached radially. This tube was excited with an air stream modulated by a sine wave. Increased evacuation was observed whenever the syringe was not located at a pressure node of the exciting sine wave. A computer model of the system showed a good matching of observed pressure versus frequency data in the syringe resonator. The results thus suggest that the alternating pressure in the nasal cavity forces the air plug in the ostium of the paranasal sinus resonators to vibrate, thus expelling from the cavity NO gas, which is transported to free air by the exhalatory air stream.
Phonological versus phonetic cues in native and non-native listening: Korean and Dutch listeners' perception of Dutch and English consonants
Authors: Cho Taehong , McQueen James M.
Published: May 2006

We investigated how listeners of two unrelated languages, Korean and Dutch, process phonologically viable and nonviable consonants spoken in Dutch and American English. To Korean listeners, released final stops are nonviable because word-final stops in Korean are never released in words spoken in isolation, but to Dutch listeners, unreleased word-final stops are nonviable because word-final stops in Dutch are generally released in words spoken in isolation. Two phoneme monitoring experiments showed a phonological effect on both Dutch and English stimuli: Korean listeners detected the unreleased stops more rapidly whereas Dutch listeners detected the released stops more rapidly and/or more accurately. The Koreans, however, detected released stops more accurately than unreleased stops, but only in the non-native language they were familiar with (English). The results suggest that, in non-native speech perception, phonological legitimacy in the native language can be more important than the richness of phonetic information, though familiarity with phonetic detail in the non-native language can also improve listening performance.
Using an artificial neural network to classify black-capped chickadee (Poecile atricapillus) call note types
Authors: Dawson Michael R. W. , Charrier Isabelle , Sturdy Christopher B.
Published: May 2006

The “chick-a-dee” call of the black-capped chickadee (Poecile atricapillus) contains four note types, A, B, C, and D that have important functional roles. This provides strong motivation for studying the classification of acoustic components of the call into different note types. In this paper, the spectrograms from a sample of A, B, and C notes (370 in total) were each described as a set of 9 summary features. An artificial neural network was trained to identify note type on the basis of these features, and was able to obtain better than 98% accuracy. An internal analysis of this network revealed a distributed code in which different hidden units generated high activities to different subsets of notes. By combining these different sensitivities, the network could discriminate all three types of notes. The performance of this network was compared to a discriminant analysis of the same data. This analysis also achieved a high level of performance (95%). A comparison between the two approaches revealed some striking similarities, but also some intriguing differences. These results are discussed in terms of two related issues: developing a research tool for note classification, and developing a theory of how birds themselves might classify notes.
The influence of spectral, temporal, and interaural stimulus variations on the precedence effect
Authors: Dizon Roberto M. , Colburn H. Steven
Published: May 2006

The precedence effect describes phenomena that are believed to aid localization of sounds in reverberant environments. These phenomena relate to the emphasis given to the first-arriving or preceding sound. In this paper, experiments are described which study precedence using stimulus parametrizations spanning temporal, spectral, and interaural dimensions. Subjects report the sidedness of headphone stimuli comprising a source and single reflection placed symmetrically with respect to midline. Most of the experiments use long-duration noises with the onset and offset time-of-arrival differences windowed out from the combined lead and lag stimulus, thus requiring the subject to lateralize using cues in the ongoing portion of the stimuli where the lead and lag overlap completely. A similar experiment using click stimuli is included for comparison. The influence of spectral content is studied by varying either the bandwidth or the center frequency. Dependence on interaural cues is investigated by using either ITDs or IIDs to induce laterality in the individual lead and lag components. Results indicate that precedence continues into the ongoing portion of long-duration stimuli and is robust to the removal of initial onsets, to reduction of bandwidth, and to the choice of interaural cue used to induce laterality in the lead and lag.
Determination of the nonlinear parameter by propagating and modeling finite amplitude plane waves
Authors: Chavrier F. , Lafon C. , Birer A. , Barrière C. , Jacob X. , Cathignol D.
Published: May 2006

The acoustic nonlinear parameter, B∕A, is an important piece of data whenever high intensity pressure fields are under consideration. In this work, an alternative method is proposed to measure this parameter. First, the method involves measuring the sound velocity and nonlinear waveform distortion of a finite amplitude plane wave propagating through a medium, Butanediol, whose density and attenuation law have been preliminarily determined. Measurements were performed in the nearfield of a piston where plane wave propagation regime exists. Impulse response of the hydrophone was determined and pressure waveforms were obtained by a convolution process. Then, the method involves modeling, in time domain and under experimental conditions, the theoretical nonlinear waveform distortion and fitting it to the experimental results by adjusting the B∕A parameter. Comparative measurements were performed using the technique of parametric interaction. The respective results for the two methods (B∕A=11.0±10% and 10.9±5%) are in a good agreement despite a smaller degree of accuracy for the proposed method.
Discrimination of nonlinear frequency glides
Authors: Thyer Nick , Mahar Doug
Published: May 2006

Discrimination thresholds for short duration nonlinear tone glides that differed in glide rate were measured in order to determine whether cues related to rate of frequency change alone were sufficient for discrimination. Thresholds for rising and falling nonlinear glides of 50-ms and 400-ms duration, spanning three frequency excursions (0.5, 1, and 2 ERBs) at three center frequencies (0.5, 2.0, and 6.0kHz) were measured. Results showed that glide discrimination was possible when duration and initial and final frequencies were identical. Thresholds were of a different order to those found in previous studies using linear frequency glides where endpoint frequency or duration information is available as added cues. The pattern of results was suggestive of a mechanism sensitive to spectral changes in time. Thresholds increased as the rate of transition span increased, particularly above spans of 1 ERB. The Weber fraction associated with these changes was 0.6–0.7. Overall, the results were consistent with an excitation pattern model of nonlinear glide detection that has difficulty in tracking signals with rapid frequency changes that exceed the width of an auditory filter and are of short duration.
The radiation of atmospheric microbaroms by ocean waves
Authors: Waxler Roger , Gilbert Kenneth E.
Published: May 2006

A two-fluid model, air over seawater, is used to investigate the radiation of infrasound by ocean waves. The acoustic radiation which results from the motion of the air/water interface is known to be a nonlinear effect. The second-order nonlinear contribution to the acoustic radiation is computed and the statistical properties of the received microbarom signals are related to the statistical properties of the ocean wave system. The physical mechanisms and source strengths for radiation into the atmosphere and ocean are compared. The observed ratio of atmospheric to oceanic microbarom peak pressure levels (approximately 1 to 1000) is explained.
Fast calculation system specialized for head-related transfer function based on boundary element method
Authors: Otani Makoto , Ise Shiro
Published: May 2006

Recently, development of a numerical calculation of the head-related transfer function (HRTF) has been conducted using a computer model of a human head and the boundary element method. The reciprocity theorem is incorporated into the computational process in order to shorten the computational time, which is otherwise very long. On the other hand, another fast HRTF calculation method for any source position, which is realized by calculating factors independent of the source position in advance, has been suggested by the authors. Using this algorithm, the HRTF for any source position can be obtained in a few seconds with a common PC. The resulting HRTFs are more precise and are calculated faster than those by using the reciprocity theorem. However, speeding the process up even further is required in order to respond to a head movement and rotation or to moving sources during binaural sound reproduction. In this paper, a faster calculation method by incorporating a time domain operation into the authors’ previous algorithm is proposed. Additionally, the new formulation, which eliminates the extra computational time in the preprocess, is proposed. This method is shown to be faster than the previous ones, but there are some discrepancies at higher frequencies.
Signal to noise ratio analysis of maximum length sequence deconvolution of overlapping evoked potentials
Authors: Bohórquez Jorge , Özdamar Özcan
Published: May 2006

In this study a general formula for the signal to noise ratio (SNR) of the maximum length sequence (MLS) deconvolution averaging is developed using the frequency domain framework of the generalized continuous loop averaging deconvolution procedure [Özdamar and Bohórquez, J. Acoust. Soc. Am. 119, 429–438 (2006)]. This formulation takes advantage of the well known equivalency of energies in the time and frequency domains (Parseval's theorem) to show that in MLS deconvolution, SNR increases with the square root of half of the number of stimuli in the sweep. This increase is less than that of conventional averaging which is the square root of the number of sweeps averaged. Unlike arbitrary stimulus sequences that can attenuate or amplify phase unlocked noise depending on the frequency characteristics, the MLS deconvolution attenuates noise in all frequencies consistently. Furthermore, MLS and its zero-padded variations present optimal attenuation of noise at all frequencies yet they present a highly jittered stimulus sequence. In real recordings of evoked potentials, the time advantage gained by noise attenuation could be lost by the signal amplitude attenuation due to neural adaptation at high stimulus rates.
Time-domain impedance boundary conditions for surfaces with subsonic mean flows
Authors: Li X. D. , Richter C. , Thiele F.
Published: May 2006

The time-domain impedance boundary conditions of Tam and Auriault are extended for impedance boundaries with subsonic mean flows. This extension requires an effective impedance. A model for the effective impedance is proposed, which guarantees the continuity of particle displacement over an infinitely thin shear layer connecting the fluid to the impedance boundary as described by Myers. Together with the effective impedance model, the time domain impedance boundary condition renders the requirement of measuring the effective impedance under flow conditions unnecessary. To demonstrate the feasibility of the extended boundary conditions, a numerical validation using the NASA Grazing Incidence Tube measurements of Jones et al. is carried out. The assumed velocity on the infinitely thin shear layer connecting the flow over the surface to the locally reacting wall is shown to have a strong influence on the performance of the liner. Another important parameter identified is the shape of the mean flow profile, which not only causes convective effects but also may excite possible flow instabilities. It is demonstrated that satisfactory results can be obtained using the extended time domain impedance boundary condition based on the correct effective impedances and mean flow profiles.
Simultaneous latency estimations for distortion product otoacoustic emissions and envelope following responses
Authors: Purcell David W. , Van Roon Patricia , Sasha John M. , Picton Terence W.
Published: May 2006

The purpose of this research was to simultaneously estimate processing delays in the cochlea and brainstem using the same acoustic stimuli. Apparent latencies were estimated from ear canal measurements of 2f1-f2 distortion product otoacoustic emissions (DPOAEs), and scalp recordings of the f2-f1 envelope following response (EFR). The stimuli were equal level tone pairs (65dB SPL) with the upper tone f2 set at either 900 or 1800Hz to fix the initiation site of the DPOAE and EFR. The frequency of f1 was swept continuously between frequency limits chosen to keep the EFR response between 150 and 170Hz. The average DPOAE latencies were 9.6 and 6.2ms for f2=900 and 1800Hz, and the corresponding EFR latencies were 12.4 and 8.8ms. In a control condition, a third (suppressor) tone was added near the DPOAE response frequency to evaluate whether the potential source at fdp was contributing significantly to the measured emission. DPOAE latency is the sum of both inward and outward cochlear delays. The EFR apparent latency is the sum of inward cochlear delay and neural processing delay. Neural delay was estimated as approximately 5.3ms for both frequencies of stimulation.
The Protease Complex Consisting of Dipeptidyl Peptidase IV and Seprase Plays a Role in the Migration and Invasion of Human Endothelial Cells in Collagenous Matrices
Authors: Ghersi Giulio , Zhao Qiang , Salamone Monica , Yeh Yunyun , Zucker Stanley , Chen Wen-Tien
Published: May 2006

Dipeptidyl peptidase IV (DPP4/CD26) and seprase/fibroblast activation protein α are homologous type II transmembrane, homodimeric glycoproteins that exhibit unique prolyl peptidase activities. Human DPP4 is ubiquitously expressed in epithelial and endothelial cells and serves multiple functions in cleaving the penultimate positioned prolyl bonds at the NH2 terminus of a variety of physiologically important peptides in the circulation. Recent studies showed a linkage between DPP4 and down-regulation of certain chemokines and mitogenic growth factors, and degradation of denatured collagens (gelatin), suggesting a role of DPP4 in the cell invasive phenotype. Here, we found the existence of a novel protease complex consisting of DPP4 and seprase in human endothelial cells that were activated to migrate and invade in the extracellular matrix in vitro. DPP4 and seprase were coexpressed with the three major protease systems (matrix metalloproteinase, plasminogen activator, and type II transmembrane serine protease) at the cell surface and organize as a complex at invadopodia-like protrusions. Both proteases were colocalized at the endothelial cells of capillaries, but not large blood vessels, in invasive breast ductal carcinoma in vivo. Importantly, monoclonal antibodies against the gelatin-binding domain of DPP4 blocked the local gelatin degradation by endothelial cells in the presence of the major metallo- and serine protease systems that modified pericellular collagenous matrices and subsequent cell migration and invasion. Thus, we have identified a novel mechanism involving the DPP4 gelatin-binding domain of the DPP4-seprase complex that facilitates the local degradation of the extracellular matrix and the invasion of the endothelial cells into collagenous matrices. (Cancer Res 2006; 66(9); 4652-61)