bluuebunny/crossref_metadata_2025 huggingface dataset:
Optimization of micro-loudspeaker radiation via finite element modeling
Authors: Bontomase Anthony R. , Burroughs Courtney B.
Published: February 2014

Measurements and predictions from numerical models for the response of and radiation from a micro-loudspeaker are compared. The numerical models are then used to examine various means for improving the performance of the loudspeaker radiation by varying the material properties of the loudspeaker. A commercial finite element code uses measured material properties to predict the diaphragm velocity distribution of lower order modes. Noncontact laser vibrometer measurements on the excited speaker are compared with the FEM predictions. The radiation from the loudspeaker, at frequencies of resonance, is predicted by a boundary element model. Diaphragm material properties (density, thickness, and elastic modulus) were varied and the sensitivity of the radiation to the material properties of the diaphragm determined.
Modeling of perforated absorbing silencers
Authors: Selamet Ahmet , Lee Iljae , Huff Norman T.
Published: February 2014

The acoustic characteristics of single-pass perforated ducts filled with absorbing material are studied experimentally and analytically. The transmission loss predictions from three models [lumped, one-dimensional decoupled, and three-dimensional boundary element method (BEM)] are compared with experimental data from an extended impedance tube setup in the absence of mean flow. Experimentally determined complex characteristic impedance and propagation constant are used to account for the wave propagation through the absorbing material. The perforation impedance suggested by Sullivan [J. Acoust. Soc. Am. 64, 207–215 (1978)] is modified to incorporate the effect of absorbing material. The results show that: (1) the effect of absorbing material on the perforation impedance needs to be taken into account for accurate predictions; (2) silencers filled with absorbing material exhibit significantly higher attenuation at higher frequencies than those without the filling material; (3) the three-dimensional BEM shows good agreement for the overall frequency range of interest (0–3200 Hz), while the one-dimensional approach is reasonable at relatively low frequencies; and (4) the lumped model, which treats the filled silencer as a Helmholtz resonator, may provide an approximate frequency for the location of peak attenuation.
Broadband, multimode, free-flooded, baffled circular ring projectors
Authors: Aronov Boris , Oishi Tetsuro , Reinhart Lawrence , Brown David A
Published: February 2014

In many underwater applications, it is required to have broadband unidirectional projectors with a small electrical Q and large electrical power factor. Cylindrical transducers made of circular piezoceramic rings with a part of the surface baffled can be employed to achieve unidirectionality in the horizontal plane over a broad frequency range. In our presentation, it is shown theoretically and experimentally that the operational frequency range of a projector can be extended by simultaneously exciting the 0 (breathing) and 1 (dipole) modes of ring vibration. In addition, the corresponding resonance modes of the internal volume of the free-flooded transducer can further enhance the frequency response. The results of an experimental investigation of the electroacoustical parameters of multimode fluid-filled cylindrical transducers are presented. [This work was supported in part by SBIR N99-011: ONR321SS,ONR36-(J.Rice), BTECH(IRaD), and ONR321SS(Lindberg).]
New and unusual percussion instruments
Authors: Rossing Thomas D.
Published: February 2014

Many new and unusual percussion instruments have been developed in recent years, and more are in the experimental stage. What is often termed ‘‘contemporary sound’’ makes extensive use of percussion instruments. We will describe the acoustical properties of a number of new percussion instruments made of wood, metal, glass, and stone. Normal modes of vibration and sound radiation will be emphasized.
Sounding clay: Pre-Hispanic flutes
Authors: Rawcliffe Susan
Published: February 2014

Ms. Rawcliffe makes and plays ceramic flutes and sound sculptures. Many were inspired by her explorations into ancient and wonderful wind instruments. To build a better flute, she studied construction methods of both contemporary and pre-Hispanic artists. The laws of acoustics dictate the possibilities for instrument construction within which design decisions are made according to cultural and individual preferences. She makes acoustical copies, learns to play them, then reinvests her insights, evolving through stages into new instruments, which inspire with their wonderful scales and evocative timbres. For 30 centuries, societies from the Olmecs to the Mayans and Aztecs developed a unique flute organology in a great diversity of form, timbre, and tunings. Many of the most complex innovations of the pre-Hispanic ceramists resulted in instruments of restricted pitch but rich timbre. They are not so much the ‘‘remains of a bygone art, as the sacred sound symbols of a now vanished cult.’’ In reviving this lost craft and science, Ms. Rawcliffe is able to play sounds that may not have been heard for 1000 years. For this presentation, using slides of her own and of ancient flutes, and performances on her flutes, she will discuss and illustrate the above issues.
The Woodstock Gamelan
Authors: Ayers Lydia , Horner Andrew
Published: February 2014

This paper considers the spectral properties of the Woodstock Gamelan, a 3-octave set of tubular chimes built by Woodstock Percussion in upstate New York. One of the main features of the instrument is expandability, and it includes 75 microtones in the middle octave. The justly tuned instruments of Harry Partch inspired the instrument. The Woodstock Gamelan has two types of aluminum tubes. Racks support the tubes down to Eb4, and these have a vibraphonelike timbre. The larger hanging tubes go down to G3, and have larger diameters than the rack tubes. The sound of the hanging tubes is similar to that of orchestral chimes. The Woodstock Gamelan has five exponentially decaying partials, and their frequency ratios are the same in the low and high registers. The frequency ratios measured were about 1, 2.69, 5.15, 8.38, and 12.08. The ratios are close to the just ratios of 1, 2.667, 5.333, 8.533, and 12. A Csound model for the Woodstock Gamelan has been developed. Listening tests show that the model produces tones nearly indistinguishable from the original. The model produces attractive related timbres by simple changes to the parameters.
Helioseismology—sound inside the sun
Authors: Hill Frank
Published: February 2014

In 1960, the solar surface was found to be moving radially with a period of 5 min. The cause of this effect was found to be the trapping of acoustic waves in the solar internal thermal gradient. Since the sound fills a resonant cavity, only specific oscillatory modes are present whose characteristics depend on the plasma properties. Thus, the solar interior can be remotely sensed by observing the trapping region’s upper boundary at the surface. This led to the field of helioseismology and produced new information about the solar internal rotation rate, composition, neutrino flux, convection zone depth, and internal changes associated with the solar activity cycle. It has also generated progress in remote sensing techniques (acoustic holography, ring diagrams, inversions), observing facilities (GONG Network, MDI instrument on SOHO), and insights into the coupling of turbulent convection and sound. Presumably, other stars also contain sound and their interiors could thus be probed, but so far no conclusive observations of these vibrations have been obtained. [Work supported by NSF and NASA.]
Global infrasonic monitoring of large meteoroids
Authors: ReVelle Douglas O.
Published: February 2014

The hypersonic meteoroid/bolide–atmosphere interaction can generate numerous phenomena, including strong shocks and infrasonic waves. For a nonfragmenting bolide, the line source, blast wave radius, Ro, is the product of the Mach number and the body diameter. For bolides reaching continuum flow, Ro can range from ∼1 m to many kilometers. To be detected, Ro must exceed ∼10 m [at a minimum source energy, Es, of ∼4.10(7) J]. Beyond 10 Ro, a frequency equal to Cs/(2.81 Ro) will be generated which will change during propagation due to effects of nonlinearity, absorption, and dispersion. Infrasonic arrays are now being routinely used to detect and locate such large near-Earth objects. Data from the CTBT IMS monitoring system will help refine estimates of the large body influx rate. These arrays have baselines of ∼1 to 2 km using microbarometers with response from ∼10 to 0.01 Hz. For ranges from 100 to 14000 km, such bolides have been observed previously with amplitudes from 0.02 to 16 Pa, with periods from 0.50 to 300 s, with total durations from 0.10 to 25 min and with computed acoustic efficiencies varying from 0.01% to ∼10%. The influx rate will be discussed as well as several recent bolides detected by Los Alamos arrays and by US DoD satellites.
Encoding of sinusoidal and natural speech tokens in the auditory nerve
Authors: Loebach Jeremy , Wickesberg Robert
Published: February 2014

Sinewave speech is a synthetic variation of natural speech that replaces natural formants with time varying sinusoidal waves. This synthesis preserves the general formant motion of the natural utterance while removing many spectral details; yet sinewave speech tokens can convey a linguistic message [Remez et al., Science 212, 947–950 (1981)]. This study examined the extent that phonetic information in sinusoidal speech is preserved at the level of the auditory periphery. We compared the responses of individual auditory nerve fibers in ketamine anesthetized chinchillas to the natural and sinusoidal tokens of the word /bal/. Comparisons of the first 100 milliseconds of the responses revealed high correlations over the first 20 milliseconds of the responses representing the initial consonant (r=0.82) despite low correlations between the stimuli themselves (r=0.18). Correlation coefficients decreased in the vowel portion of the responses (r=0.30). Spectral reduction in the sinusoidal token may produce the low correlation between the neural representations of the vowel. The preservation of temporal cues may explain the high correlation between the neural representations of the consonant despite spectral differences.
Consistency of the auditory nerve response to normal and whispered speech
Authors: Justin Stephanie , Wickesberg Robert
Published: February 2014

Multiple presentations of an acoustic stimulus are often used to study the encoding of sounds that in the natural environment are not repeated. This study examined how consistent the response of the auditory nerve is with respect to the number of repetitions. Naturally voiced and whispered forms of /ta/ were presented 50 times at 60 and 90 dB to ketamine anesthetized chinchillas. For each stimulus, global average peri-stimulus time (GAPST) histograms for the first 20 trials were compared to those for the last 20 trials. Correlation coefficients for the normal syllable were 0.87 and 0.72 during the consonant and vowel, respectively. In the whispered condition, correlation coefficients were 0.95 during the consonant, but only 0.44 during the vowel. Similar correlations were obtained with comparisons of GAPSTs for either the first or last 20 trials to those for all 50 trials. Correlations obtained from the GAPSTs were higher than those of individual responses for both speech conditions and at all intensities. Individual auditory nerve fiber correlations during the speech stimuli were variable. These results demonstrate that the use of many repetitions to achieve consistency in the response may not be necessary when one studies the ensemble response of the auditory nerve.
A low-frequency paradox in binaural pitch
Authors: Hartmann William M. , Zhang Peter Xinya
Published: February 2014

Binaural pitches, such as the Huggins pitch and binaural coherence edge pitch (BICEP), can be approached in terms of a model network of binaural neurons that are tuned in frequency and in interaural delay [e.g., Raatgever and Bilsen, J. Acoust. Soc. Am. 80, 429–441 (1986)]. The basic rule of binaural combination resembles addition. Binaural pitches appear as peaks or edge features in the frequency-delay plane. The characteristic frequency of a feature determines the pitch, and the internal delay of the feature determines the lateralization of the pitch image. Because delay lines become sparser with increasing delay, the model predicts that binaural pitches of very low frequency should be more difficult to hear the more they are lateralized. However, our Huggins pitch and BICEP discrimination and detection experiments show that the reverse is true. These experiments favor a model in which the basic rule of binaural combination resembles subtraction. However, the predictions of the subtractive model disagree with the observed lateralization. Furthermore, the low-frequency discrimination and detection experiments do not agree with a model in which additive and subtractive binaural combinations are alternative options. This is the low-frequency paradox. [Work supported by NIDCD.]
The binaural processing of double vowels: Experimental measurements and computational models
Authors: Akeroyd Michael A.
Published: February 2014

The degree to which the binaural processing of a ‘‘target’’ vowel is affected by the simultaneous presentation of a ‘‘distractor’’ vowel (of different ITD to the target) was studied using three pairs of synthetic vowels. The stimuli were an ‘‘er’’ (of 100-Hz F0) paired together with either an ‘‘ai,’’ ‘‘er,’’ or ‘‘oo’’ (all of 125-Hz F0). In Experiment 1 ITD thresholds for the target vowel were measured. The results showed that thresholds were larger in double-vowel pairs than in control, target-alone situations. Moreover, the size of the effect was dependent both on whichever vowel of the pair was the target and on the relative level of the two vowels. Computational modeling of the binaural correlograms of the stimuli showed that these results do not require any explicit segregation of frequency channels for the two vowels. In Experiment 2 the perceived lateralization of the target vowel (when given an ITD of 0 ms) was measured. The results showed that this lateralization was ‘‘pulled’’ slightly toward the distractor vowel. Further computational modeling showed that some segregation of the frequency channels in the binaural correlogram, perhaps based on the pitch information in the corresponding autocorrelogram, is required to account for this result.
The independence of adaptation to interaural time and level changes
Authors: Neelon Michael F. , Benedon Carrie M. , Jenison Rick L.
Published: February 2014

Changes in interaural time and interlevel differences (ITDs and ILDs) for a sound moving in the free field normally imply movement in the same direction. Research has shown that listeners can be adapted to and show aftereffects for each of these cues separately [W. H. Ehrenstein, Perception 23, 1249–1255 (1994)]. Psychophysical evidence of ILD to ITD conversion, however, raises the question whether adaptation is independent between these two channels. The present experiment explores their possible independence in order to better specify where in the auditory pathway their adaptation occurs. Listeners will be adapted to sounds with dynamically varying ITD and ILD cues alone, in concert or in conflict with each other (the latter conditions implying either the same or opposite direction of movement). ITD and ILD adaptation will be measured separately via test sounds changing in either of these cues alone. If aftereffects for one cue are unaffected by the implied movement direction of the other, then adaptation most likely lies early in the auditory pathway (e.g., the olivary bodies). However, if aftereffects show an influence of one cue on the other, some later physiological site of convergence may be involved in adaptation (e.g., the inferior colliculus).
The role of the precedence effect in sound source lateralization
Authors: Shub Daniel E. , Gilkey Robert H. , Colburn H. Steven
Published: February 2014

This work examines sound source lateralization of ongoing broadband noise targets in the presence of a single ongoing broadband noise jammer, which was either a simple simulated reflection of the target or was uncorrelated with the target. Identification thresholds of ±300-μs ITD targets and lateralization thresholds were determined. Identification thresholds of the reflection jammers were 3.2 dB lower (P=0.027) than the identification thresholds of the jammers that were uncorrelated with the targets. Lateralization thresholds of the reflection jammers were 2.85 dB lower (P=0.040) than the lateralization thresholds of the uncorrelated jammers. There was a slight trend, in agreement with the current understanding of the precedence effect, for both the identification and lateralization thresholds of the simple reflection jammers to be dependent on jammer ITD. The 0-μs ITD reflection jammer produced a threshold that was 1.9 dB lower (P=0.104) than the threshold of the 643-μs ITD reflection jammer. Additionally comparisons between the lateralization thresholds of normal-hearing listeners and two cross-correlation models were made. The models obtained lateralization thresholds as low as −13 dB, up to 10 dB better than normal-hearing performance. [Work supported by NIH Grant No. R01 DC00100.]
Localization and speech-identification ability of hearing-impaired listeners using phase-preserving amplification
Authors: Drennan Ward , Gatehouse Stuart , Howell Patrick , VanTasell Dianne , Lund Steven
Published: February 2014

Hearing-impaired listeners experience increased difficulties recognizing speech and localizing sounds in adverse environments. This study investigated the benefits of signal processing in binaural hearing aids designed to preserve cues that accompany spatial location. The ability of listeners to localize click-trains in noise was tested, along with their ability both to localize and to identify words in noise (a dual task). Listeners experienced two types of bilateral hearing aid fittings: (1) a custom fitting that provided appropriate gain while also matching the phase measured near the tympanic membranes without the hearing aids, and (2) a conventional fitting (using the same hearing aid device) that provided the same gain with noncustom, linear phase. Testing occurred for each fitting immediately and following 3-months listening experience using a within-listener, within-device, randomized, single blind, crossover design. A rating-scale questionnaire was administered to assess perceived speech- hearing and spatial abilities. In the dual task, the listeners exhibited superior localization ability for the phase-preserving fitting initially and after 3-months experience. This advantage did not occur for the click-train localization task. Listeners rated their spatial abilities higher with the phase-preserving fitting, although little improvement was observed or reported for speech hearing.
Auditory motion aftereffects with varying interaural phase difference
Authors: Kawashima Takayuki , Sato Takao
Published: February 2014

It has been known that the auditory motion aftereffect (MAE) is spatially specific [see, for example, D. W. Grantham, Acoustica 84, 337–347 (1998), exp. 3]. However, it still is not very clear which cues for sound localization are responsible for this spatial specificity of the MAE, since several possible cues, such as the spectral profile or interaural level difference, covaried in most past studies. In this study, MAE and its spatial specificity were investigated with sound motion created by using only interaural phase difference (IPD) as a first step to identify the responsible cues. We used the probe method with the method of constant stimuli. Either with or without adaptation to a moving tone, subjects were asked to judge the direction of probe tone motion (0.7 ms duration, either to the left or the right). The slope and the position of the psychometric functions were affected by the direction of the adapter’s motion direction, but only when IPD ranges (spatial ranges of motion) of the adapter and probe tones overlapped each other. These results suggest that the change in IPD is at least one of the cues which produce the spatial specificity.
Homogenization of a ribbed cylindrical shell over the full frequency range for efficient calculation of structural-acoustic behavior
Authors: Franzoni Linda , Bliss Donald
Published: February 2014

A homogenization method for complex structures, applicable for all frequencies, is described for a cylindrical shell with periodically spaced ribs. This approach has application to naval and aerospace structures. The homogenization method utilizes a local–global decomposition facilitated by adding and subtracting canceling smooth forces. The smooth global problem has an infinite-order structural operator, and periodic discontinuities are replaced by equivalent distributed suspension terms. The global problem can be solved very efficiently since all rapidly varying scales have been removed. The local problem, which provides transfer function information for the global problem, is solved separately and independently, except for amplitude information from the global problem. Once formulated, the self-contained global problem is solved first, and the local solution can be reconstructed afterwards. In the present work, the ribs are modeled as annular plates periodically attached to the cylindrical shell. The coupled shell equations are re-cast using the method of local–global homogenization, and are solved efficiently using a combination of symbolic manipulation and numerical methods. Comparisons withother solution methods are presented, and the extension to fluid loading is discussed. [Work sponsored by ONR.]
Visualization of the energy flow for a guided forward wave in and around a fluid loaded elastic cylindrical shell
Authors: Dean Cleon E. , Braselton James P.
Published: February 2014

Previous work [Cleon E. Dean and James P. Braselton, J. Acoust. Soc. Am. 107, 2921 (2000)] showed the energy flow for a fluid loaded elastic cylindrical shell at the resonance frequency. The results were equivocal since there are two counterpropagating guided waves. The current work uses a clever analysis due to Kaduchak and Marston [Gregory Kaduchak and Philip L. Marston, J. Acoust. Soc. Am. 98, 3501–3507 (2000)] to separate the two waves and show the energy flux associated with just one forward propagating surface guided wave.
Production, coproduction and perception of speech gestures
Authors: Carré René
Published: February 2014

An acoustic model was used to derive distinctive speech gestures that result in dynamic deformations of the area function of an acoustic tube. These deformations are efficient, i.e., they provide maximum acoustic contrasts. It can be demonstrated that the gestures generated by the model are consistent with those in natural speech production. Using sequential and/or parallel combination of gestures, V1V2 and V1CV2 utterances were generated and presented to listeners to assess the effect of a particular coproduction scheme on phonemic identification. Perception of some gesture combinations remained remarkably invariant despite considerable variations of coproduction parameters: In a V1CV2 token, for example, the onset of the consonant closure re/ the V1V2 transition can change to a large degree without altering the percept. In contrast, tokens such as [aya], obtained with two coproduced gestures (one tongue and one labial), require strict intergesture timing when [y] is just reached, in order for the French labial /y/ to be perceived. A comparison of the two classes of coproduction schemes and their perceptual consequences suggests possible ways of extracting, from formant trajectories (in the F1-F2 plane or the F1-F2-F3 solid) represented in the time domain, acoustic cues that may underlie a number of coproduced gestures.
What are the essential cues for understanding spoken language?
Authors: Greenberg Steven
Published: February 2014

Classical models of speech recognition (by both human and machine) assume that a detailed, short-term analysis of the acoustic signal is essential for accurately decoding spoken language. Several lines of evidence call this assumption into question: (1) intelligibility is relatively unimpaired when the frequency spectrum is distorted under a wide range of conditions (including cross-spectral asynchrony, reverberation, waveform time reversal and selective deletion of 80% of the spectrum), (2) the acoustic properties of spontaneous speech rarely conform to canonical patterns associated with specific phonetic segments, and (3) automatic-speech-recognition phonetic classifiers often require ca. 250 ms of acoustic context (spanning several segments) to function reliably. This pattern of evidence suggests that the essential cues for understanding spoken language are largely dynamic in nature, derived from (1) the complex modulation spectrum (incorporating both amplitude and phase) below 20 Hz, (2) segmentation of the speech signal into syllabic intervals between 50 and 400 ms, and (3) a multi-time-scale, coarse-grained analysis of phonetic constituents into features based on voicing, manner and place of articulation. [Work supported by the U.S. Department of Defense and NSF.]
Bayesian inference in architectural acoustics: Estimation of decay times in coupled spaces
Authors: Goggans Paul M. , Xiang Ning
Published: February 2014

Sound energy in coupled spaces decays at multiple rates under certain conditions. Identifying decay times in these coupled spaces often demands considerable effort. Generally, different portions of multi-rate decay functions obtained from measured data cannot always be distinctly recognized. A model-based parameter estimation approach within the Bayesian framework facilitates the evaluation of multiple decay times through an extension of the decay model established in a previous work [N. Xiang, J. Acoust. Soc. Am. 98, 2112–2121 (1995)]. Unlike gradient-based approaches, no careful estimation of initial values is required. Therefore, a robust algorithmic estimation of multiple decay times from experimentally measured decay functions shows advantages over the existing nonlinear regression approach.
Acoustic inversion via linearization and Bayesian multipath identification
Authors: Ma Xiaoqun , Michalopoulou Zoi-Heleni
Published: February 2014

A linearization approach to acoustic inversion is proposed employing distinct ray paths (direct arrival, first surface bounce, and first bottom bounce) for source localization and bathymetry and sound speed estimation. The ray path arrivals are selected from broadband, shallow water, synthetic data using a Bayesian time delay estimation scheme calculating posterior probability density functions of the delays in an efficient way. A linear system is then formed relating unknown parameters and arrival time data. The regularization method is used for the solution of the linear system [S. E. Dosso et al., J. Acoust. Soc. Am. 104, 846–859 (1998)] with excellent results. Also a simple least-squares approach for the solution of the system is implemented; results of the two approaches are compared. Finally, the linearization multipath based technique is successfully applied to real acoustic broadband data for source and receiver localization, and bathymetry and sound speed estimation. [Work supported by ONR.]
A Bayesian framework for perceptually motivated audio signal enhancement
Authors: Wolfe Patrick J. , Godsill Simon J.
Published: February 2014

Spurred by the success of perceptual models in audio coding applications, researchers have begun to address audio signal enhancement in a similar manner. Here a statistical model-based approach is presented that uses cost functions based on auditory perception. As noise reduction inevitably occurs at the expense of signal resolution, why not take advantage of human perception in order to optimize this trade-off? By mathematically incorporating the notion of perceived signal quality, Bayesian risk theory does just that. The Bayesian paradigm is shown to provide an ideal framework within which to formalize such an approach, as it is rigorous, powerful, and generalizable. At the same time it allows the incorporation not only of psychoacoustic optimality criteria, but also of additional nonperceptual prior information such as that concerning audio signal behavior in time and frequency. Importantly, it also permits sequential estimation for real-time noise reduction. Audio examples and the corresponding noise reduction software may be found at http://www-sigproc.eng.cam.ac.uk/∼pjw47. [Material by the first author is based upon work supported by a U.S. National Science Foundation graduate research fellowship.]
Scattering into the shadow zone
Authors: Munk Walter
Published: February 2014

We consider the scattering of sound into the shadow zone in long-range transmission by processes other than internal waves. Microfrontal activity measured by Dan Rudnick in the north Pacific appears to be a significant factor.
Modeling of sound propagation in sandy ocean sediments
Authors: Chotiros Nicholas P.
Published: February 2014

A physically accurate model of sound reflection and penetration into ocean sediments is the objective. Sandy sediments are of particular interest because current models are not compatible with acoustic measurements. The discrepancies of the fluid and elastic solid approximations are clearly demonstrable. The Biot theory has the potential to satisfy the objective. Stolls formulation requires 13 input parameters, which may be divided into three groups according to the accuracy with which they are known. The first group consists of tabulated physical constants; the second is less precisely known, and the third is not measurable. An inversion procedure was devised to estimate the immeasurable group from simple acoustic measurements — reflection loss, compressional and shear wave speeds, and attenuations. The imprecisely known group was handled in a probabilistic manner. The inversion results for water-saturated sand, using published laboratory and at-sea measurements, show a definite incompatibility between model and measurements. To find a solution, a couple of hypotheses were considered: (1) the inclusion of some proportion of the pore fluid within the frame and (2) the relaxation of the uniform, elastic frame assumption. The latter proved to be a viable solution. [Work supported by ONR, Ocean Acoustics.]
Quantitative characterization of the uppermost seabed stratum: Some results from the ISACS project
Authors: Caiti Andrea , Hovem Jens M. , Berntsen Bjarte
Published: February 2014

ISACS is a research initiative funded by the European Union in the years 1996–1999. The objective was to prove the feasibility of seafloor characterization by exploitation and integration of data from commercially available sonar equipment. The activities have focused on analysis of side-scan, multibeam and parametric sonar data, and have resulted in processing methods for classification, visualization, and quantitative characterization of the uppermost seabed stratum. Among these, two approaches have been implemented as stand-alone software packages for identification of bottom roughness, impedance and volume inhomogeneities by inversion of parametric sonar data. The first package, sirob, relies on a model-based approach in which the forward model is the prediction code boris. Bottom parameters are identified by minimizing the data-prediction discrepancy in a generalized time-frequency domain. The second package, farim, uses the coherent part of the backscattered time signal. The frequency shift in the data is used for finding the roughness, and roughness estimate is used to correct the impedance estimate found from the returned echo. farim is simpler and faster than sirob, but it determines a subset of the parameters identified by sirob. Results obtained on field data show good agreement among both methods and available ground truth.
Geoacoustic inversion hybrid model
Authors: Haire Mark S. , Knobles David P.
Published: February 2014

Simulated annealing (SA) or genetic algorithms are accepted optimization methods for geoacoustic inversion to obtain seabed parameters from measured acoustic data. These techniques permit optimizations where numerous local minima are present, but are computationally intensive. Gradient methods, such as the Levenberg–Marquardt (LM) approach, are extremely efficient at finding local minima, but typically are unable to locate the global minimum. To improve the computation speed and efficiency of SA, a ‘‘hybrid’’ algorithm combining SA and LM has been developed. The hybrid approach applies a LM algorithm after each temperature cycle of SA if an improved cost function value has been obtained during the cycle. The hybrid technique takes advantage of the fact that SA often examines a point that is in the proximity of the global minimum when the temperature is high. Factors of 3–8 increases in speed over SA alone are obtained with simulated data. Results will be presented from the application of the hybrid model to experimental data collected during the Area Characterization Test-I in the Gulf of Mexico. [Work supported by ONR.]
Expansion of the isoplanatic patch for correction of ultrasonic aberration by assignment of corrections based on similarity of propagation paths
Authors: Pilkington Wayne C. , Waag Robert C.
Published: February 2014

Aberration correction for ultrasonic b-scan images can be effectively implemented with less computation by employing cross correction, i.e., using a single set of aberration corrections computed for a reference aperture position to correct the transmit and receive focus at other points within an image region defined as the isoplanatic patch. An uncomplicated implementation of this approach compensates the signal at each location in the aperture with the aberration correction from the corresponding location in the reference aperture regardless of the aperture position. Alternatively, the signal at each location in the aperture may be compensated using the correction for the reference aperture signal with a propagation path most similar to the propagation path of the signal being corrected. This latter approach requires a different map of the reference aberration corrections for each aperture position, but the map can be implemented by simple shift operations. Measurements using a novel 2-D array system demonstrate that the assignment of corrections based on similarity of propagation paths can significantly improve the effectiveness of cross correction and enlarges the effective size of the isoplanatic patch.
Comparison between time reversal and spatio-temporal inverse filter application to focusing through a human skull
Authors: Tanter Mickael , Aubry Jean-Francois , Thomas Jean-Louis , Fink Mathias
Published: February 2014

Ultrasonic imaging systems capabilities are strongly dependent on the focusing quality of the ultrasonic beam. In the case of brain imaging, the skull strongly degrades the ultrasonic focusing pattern by introducing strong phase and amplitude aberrations of the wave-front. In previous work, this degradation of the beam focus had been partially corrected by coupling the time reversal focusing process to an amplitude compensation of the emission signals. In that case, the optimal focus was reproduced down to −20 dB, but the sidelobe level remained at about −25 dB. This elegant technique will be compared to another focusing technique recently developed in our laboratory, called spatio-temporal inverse filtering. Thanks to this method, based on the inversion of the propagation operator at each frequency within the bandwidth of our transducers, experimental focusing through the skull is now comparable to optimal focusing in a homogeneous medium. Those two methods not only differ theoretically, but also suffer differently from all the experimental defects, such as the limited bandwidth of the transducers or the limited aperture of the arrays. A comparson of the results obtained with both techniques in water and through a human skull clearly highlights the advantages and the drawbacks of each method.
Real-time additive synthesis with envelope parameter stremas: an alternative to sampling
Authors: Haken Lippold , Fitz Kelly , Christensen Paul
Published: February 2014

A sampling synthesizer uses a set of recordings for sound synthesis. A shortcoming of the sampling technique is the inability to produce complex spectral changes associated with the performer’s actions during a note. Additive synthesis, on the other hand, represents each sound as a collection of sine wave components, or partials. A wide variety of modifications are possible with additive synthesis, including frequency shifting, time dilation, cross synthesis, and sound morphing. Our real-time additive synthesis uses a set of recordings, like a traditional sampling synthesizer. Unlike a traditional sampling synthesizer, our additive synthesis models timbres as collections of bandwidth-enhanced partials (sine waves with noise) with time-varying parameters. The timbres are encoded in envelope parameter streams that can be manipulated in real time. These envelope parameter streams, which provide amplitude, frequency, phase, and bandwidth information for each partial, are the counterpart of sample streams in sampling synthesis. The talk will include demonstrations using the continuum fingerboard.
A hybrid additive-wavetable synthesis model for the horn
Authors: Horner Andrew , Ayers Lydia
Published: February 2014

Is it possible to synthesize the French horn well enough to pass for the real thing? Most commercial synthesizers and soundcards are limited to producing timbres that are ‘‘horn-like’’ or ‘‘clarinet-like.’’ It is much more difficult for a model to be flexible enough to allow different types of articulations, phrasings, and tone colors. This work presents a hybrid additive-wavetable synthesis model for the horn using genetic algorithm optimization. The parameters for the model include fundamental frequency, overall amplitude, vibrato amount, attack and release times, brightness, and articulation shape. The model gets brighter as the amplitude increases, and the brightness decreases as the amplitude decreases by scaling each partial with a set of exponentially related amplitude envelopes. The model is implemented in the Csound software synthesis language. After implementing the model, several solos and orchestral excerpts that are commonly required in auditions were synthesized. The model is particularly effective on Baroque and classical excerpts.
Noise generation by rotating blades in a riding lawnmower
Authors: Skinner Christian , Burroughs Courtney B.
Published: February 2014

Measurements presented at the Acoustical Society of America Meetings in Newport Beach, CA in December 2000 showed that the dominant source of noise radiated from our riding lawnmower is the mower deck and that most of this noise is radiated directly as airborne noise from under the mower deck. In this paper, results from measurements of unsteady pressures on the tips of mower blades and baffling under the deck are presented. Frequency spectra are presented along with fluctuations in the pressure versus blade position. Measurements were conducted on three types of blades (flat, standard, and mulching) located at each of the three blade positions under the deck. To quantify the affects of the interaction between blades, measurements were conducted with one instrumented and two noninstrumented blades installed, and with only the instrumented blade installed. Results from these measurements are presented and discussed.
Performance of a high-frequency thermoacoustic refrigerator
Authors: Abdel-Rahman Ehab , Symko Orest G. , Frates Angela M.
Published: February 2014

A thermoacoustic refrigerator pumped at an acoustic frequency of 4.4 kHz has been investigated for optimal geometric configuration for heat pumping. The device consists of a piezoelectric driver with a random stack in a wave cylindrical resonator with air at 1 atm as the working fluid. The temperature difference between the heat exchangers, at each end of the stack, has been studied as a function of acoustic intensity, stack position, stack length, and stack surface area. Results show that with sound levels of 160 dB the C.O.P. can be optimized to be around 3. Such performance strongly depends on the stack length with a compromise between effective stack area and quality factor of the resonator. Results will also be presented on the efficiency of the driver as a function of load.
Behavioral and physiological studies of binaural detection with reproducible noise in rabbit
Authors: Carney Laurel H.
Published: February 2014

This presentation will review studies of binaural detection in rabbit. Behavioral results include detection experiments with narrow band and wide band reproducible noise maskers using the Pavlovian conditioning technique. These results showed similar trends across bandwidths and binaural conditions as in a parallel study in human listeners. Comparison of the responses to the ensemble of reproducible noises across bandwidths demonstrates the influence on performance of spectral components that are outside a narrow frequency band (approximately a critical bandwidth) surrounding the tone. Physiological responses of neurons in the inferior colliculus to the ensemble of reproducible noises will be presented. These responses can be compared to decision variables provided by computational models that are being developed to predict the behavioral results. Strategies for ongoing combined studies of behavior and physiology related to detection will be presented. [Work supported by NIH-NIDCD.]
Energy source identification using reactive structural intensity
Authors: Lima Alves Paulo Sergio , Franca Arruda Jose Roberto
Published: February 2014

The analysis of wave propagation phenomena in structures, especially energy flux, can be a powerful approach to noise and vibration control problems. The active mechanical intensity corresponds to the period-averaged value and is related to the propagative wave field. The reactive intensity is related to the reverberance of energy in the structure, which originates the standing waves or mode shapes. Properly separated into its wave components, reactive intensity maps agree well with operational modes, indicating the location of the nodal lines. On the other hand, active intensity is the quantity usually addressed when solving vibration and noise problems, since it provides information about the main energy flow paths, thus enabling identification of the energy sources and sinks. For highly reverberant structures, however, measuring active intensity becomes awkward, and intensity plots fail to indicate the region where the energy is injected or dissipated in the structure. A new technique to localize energy sources based on the divergence of the reactive intensity and the distribution of the potential and kinetic energy densities within the structure is presented. Numerical results are given for a beam with a point excitation using finite-element and spectral element models based on the Bernoulli–Euler beam theory.
Application of fuzzy structures analysis to aircraft panels
Authors: Sparrow Victor W. , Buehrle Ralph D.
Published: February 2014

In 2000 a research implementation of Christian Soize’s fuzzy structures analysis procedures were implemented by Sparrow in the commercial finite element program MSC/NASTRAN [MSC Software Corp. (Los Angeles, CA)] using its direct matrix abstraction program (DMAP) capability coupled with MATLAB [Mathworks, Inc. (Natick, MA)]. The background and implementation details of the new DMAP code were recently given [AIAA paper 2001–1320, Non-Deterministic Approaches Forum, 42nd AIAA/ASME/ASCE/AHS/ASC Structures, Structural Dynamics, and Materials Conf., Seattle, WA, 2001]. The present talk will describe the application of the new code to two prototypical aircraft panels: a small 0.96 by 0.19 m panel with one longitudinal stringer and a larger 1.8 by 1.2 m panel including six longitudinal stringers and four equally spaced frame stiffeners. The smaller and larger panels were modeled in NASTRAN as having approximately 7900 and 60000 degrees of freedom, respectively. In these studies the attachment points between the stringers and skin were assumed to be fuzzy. Results will be described comparing the fuzzy and nonfuzzy finite element predictions to experimental vibration results. [Work supported by NASA Research Cooperative Agreement NCC-1-382.]
An M-mass vocal-fold excitation model and its relation to source–tract interaction
Authors: Huang Jun , Levinson Stephen , Davis Don , Slimon Scott
Published: February 2014

In this work, we propose an M-mass model of the vocal-fold excitation for articulatory speech synthesis. The vocal fold in this model is composed of M coupled oscillators. Each oscillator consists of a mass, a spring stiffness, and a dumper. The adjacent masses are coupled by a spring stiffness. The dynamic response of the M-mass model is described by a set of second-order ordinary differential equations called the equations of the motion. This M-mass model is combined with the Navier–Stokes equations to provide numerical solutions in an iterative way. First, the glottal volume velocity in the current iteration is used to determine the aerodynamic force exerted on each mass. Then the glottal area is calculated by solving the equations of motions given the aerodynamic force. This glottal area information is used to compute the glottal volume velocity in next iteration. We use this M-mass model to compute the glottal excitation signals including volume velocity and particle velocity. Experimental results show that we can observe ripples in the positive glottal opening interval of the particle velocity and volume velocity, which is strong evidence of source–tract interaction during speech production.
The effects of vowel quality and pitch on spectral and glottal flow measurements of the voice source
Authors: Epstein Melissa A. , Payri Blas G.
Published: February 2014

In efforts to evaluate voice quality, the voice source is frequently assessed using measurements of the voice spectrum and/or measurements of the glottal flow. In normal dynamic speech, though, speakers not only change their voice quality, but also their vowel qualities (i.e., they say different words) and their pitch (i.e., they use different intonation contours). The purpose of this study is to explore how vowel quality and pitch affect both spectral and glottal flow measurements of source characteristics. In the experiment, speakers are asked to say several words on three pitches each. Since these are real words, speakers should produce them in their normal, but undefined, voice quality. Samples are selected from the center of each vowel. Differentiated glottal flow is obtained by inverse filtering and is then matched with an LF model. The voice spectrum is evaluated using the measurements H1-H2, H1-A1 and H1-A3 (as proposed by Stevens and Hanson 1995). These measurements are then compared across vowels, pitches and speakers. [Work supported by the French Ministry of Foreign Affairs and NIH/NIDCD.]