bluuebunny/crossref_metadata_2025 huggingface dataset:
Experimental study of the A-wave axial propagation inside a fluid filled cylindrical shell: Application to Gelification process monitoring by surface acoustic wave
Authors: Martinez Loic , Senouci Brahim , Serfati Stephane , Gindre Marcel , Le Huerou Jean-Yves , Griesmar Pascal
Published: October 2013

During gel formation, the material passes from the liquid state to the solidlike state. In order to characterize this liquid-to-solid transition, the low-frequency fluid-born plate mode A-wave is used. The A-wave has interesting properties to be nonattenuated—very dispersive at low-frequency thickness products (fd), and nondispersive for high fd (the Scholte wave is the high fd limit). The axial propagation of the A-wave is studied along a glass cylindrical shell in the mid to high fd range (0.5 MHzmm—1.5 MHzmm). The A-wave is generated and received at one extremity of the tube by a contact broadband transducer excited by a pulse. In the calibration test, the cylindrical shell is filled with water and surrounded by air outside. The A-wave dispersion curve is extracted by short time Fourier transform; comparison with the theoretical dispersion curves is very good. For the gelification monitoring, the water is replaced by the gel in its liquid state. A-wave time signals are recorded during the gelification process. The evolution of the longitudinal speed during the gelification time is deduced from the A-wave dispersion curves.
It’s about time: Presentation in honor of Ira Hirsh
Authors: Grant Ken
Published: October 2013

Over his long and illustrious career, Ira Hirsh has returned time and time again to his interest in the temporal aspects of pattern perception. Although Hirsh has studied and published articles and books pertaining to many aspects of the auditory system, such as sound conduction in the ear, cochlear mechanics, masking, auditory localization, psychoacoustic behavior in animals, speech perception, medical and audiological applications, coupling between psychophysics and physiology, and ecological acoustics, it is his work on auditory timing of simple and complex rhythmic patterns, the backbone of speech and music, that are at the heart of his more recent work. Here, we will focus on several aspects of temporal processing of simple and complex signals, both within and across sensory systems. Data will be reviewed on temporal order judgments of simple tones, and simultaneity judgments and intelligibility of unimodal and bimodal complex stimuli where stimulus components are presented either synchronously or asynchronously. Differences in the symmetry and shape of ‘‘temporal windows’’ derived from these data sets will be highlighted.
Parameter space of the dichotic speech experiment
Authors: Bilger Robert C. , Speaks Charles E.
Published: October 2013

The experiment in which different CV nonsense syllables are presented simultaneously to each ear and the listener is asked to identify both consonants often results in more correct responses to syllables presented to the right than to the left ear, the so-called right-ear advantage (REA). The REA is generally interpreted as an indication that the left hemisphere processes speech more efficiently than does the right hemisphere. As the experiment is usually conducted, however, it has the parameter space of 3 (i.e., requires three numbers to completely specify the results). Using data obtained independently in each of our labs, it will be shown that the three parameters necessary to uniquely specify the results of the experiment are the probability that the (1) first response is correct; (2) first response is to the syllable presented to the right ear; and (3) second response will be incorrect given that the first response was correct. The ear advantage (R, 0, or L) is predicted to the nearest percent using the first two parameters; but, as of this moment, the exact fourfold matrix for a listener cannot be predicted.
Sound absorption characteristics of cenosphere enriched cement and asphalt concrete
Authors: Tiwari Vikrant , Shukla Arun , Bose Arjit
Published: October 2013

A detailed experimental study has been conducted to determine the effect of addition of cenospheres on the acoustic properties of cement matrix and asphalt concrete. Cenospheres are hollow ceramic microballoons, being a waste product of thermal power plants, they are relatively inexpensive and their use has the added benefit of decreasing the strain on the environment. Specimens were made with a different volume fraction of cenospheres (0% to 70% in increments of 10) to observe the change in acoustic properties with an increase in the cenosphere content. Specimens of different sizes were tested to study the acoustic characteristics of the materials for a wide range (125 to 4000 Hz) of frequencies. The effect of an increase in the cenosphere content on the dilatational wave velocity in the specimens was also studied with the help of acoustic transducers. It was noted that, although the wave velocity remains relatively unaffected with a change in the cenosphere content, the addition of cenospheres improves the sound absorption characteristics of the cement matrix and asphalt concrete with a considerable decrease (up to 41%) in the density of the material. It has also been noted that the sound absorption of cement increases with the addition of cenosphere up to 40% volume fraction and starts decreasing with the further addition of cenospheres.
History of structural acoustics and vibrations in the Acoustical Society of America
Authors: Feit David , Strasberg Murray , Ungar Eric E.
Published: October 2013

Structural acoustics refers to the interaction of sound and structures—the response of structures to sound, the radiation of sound from vibrating structures, and the effect of the acoustic medium on the structural vibrations. Interest in these subjects increased greatly during the 1930s and 40s because of practical applications in the design of microphones and loud speakers used in telephones, radios, and electronic phonographs. The combination of electrical and mechanical systems lead to the use of electrical engineering concepts such as impedance, circuits, and electrical analogies, in the analysis of mechanical systems. In later years, much of the work dealt with various aspects of underwater structures, prompted by U.S. Navy interests. The field, which began with classical analytical mechanics applications, has progressed to new approaches, including statistical energy analysis, near-field acoustical holography, fuzzy structures, active control of vibrations, and smart materials. In recognition of these new developments, the name of the technical committee was changed in 1987 from ‘‘Shock and Vibration’’ to ‘‘Structural Acoustics and Vibration.’’
Passive localization algorithms
Authors: Sullivan Edmund J.
Published: October 2013

An acoustic source can be localized by exploiting the fact that, if the range is not too great, the wavefront of the incoming acoustic energy can be assumed to be circular. This is sometimes called ‘‘Wavefront Curvature Ranging.’’ Here, the estimation of range reduces to the estimation of the radius of a circle. In fact, if sufficient horizontal and vertical aperture is available, range, bearing and elevation can be estimated, given a minimum of four receivers. The technique is based on the measurement of the relative phases between pairs of receivers. In this paper, two methods of improving the performance of such localizers are investigated. In the case where the receivers are mounted on a moving platform, it is shown that exploitation of this motion can improve performance by utilizing information that is usually ignored in phase measurements. Also, since the motion introduces a time dependence in the true values of the time delays, a limit on the accuracy of their measurement is incurred. A recursive estimation process is proposed as a remedy for this problem. These techniques will be demonstrated by using simulated data.
Ocean tomography, inverse methods, and broadband ocean acoustics
Authors: Cornuelle Bruce D.
Published: October 2013

Ocean acoustic tomography, as proposed by Munk and Wunsch in 1979, and as implemented by the Ocean Tomography Group, uses ray travel times to estimate ocean sound speed and currents. Earlier work by Medwin (1970) and Hamilton (1977) used pulse travel times as measures of integrated sound speed along paths at short and long range, respectively. Munk and Wunsch (1979) recognized that broadband transmissions between many instruments could be used with inverse methods [Backus and Gilbert (1967); Liebelt (1967)] to reconstruct 3D ocean sound speed fields from the travel times along multiple paths. Inverse methods are widely used in ocean acoustics and in physical oceanography, and the modern challenge is to incorporate time dependence into the inverse methods to take advantage of the improving ocean dynamical models. In addition, the understanding of broadband acoustic propagation has improved to the point of refining the sensitivity kernel for travel time measurements beyond the simple geometrical optics of ray paths. This paper will review the evolving use of forward and inverse methods in acoustical oceanography, primarily with application to acoustic tomography.
Hot topics: Signal processing in acoustics
Authors: Candy James
Published: October 2013

Signal processing represents a technology that provides the mechanism to extract the desired information from noisy acoustical measurement data. The desired result can range from extracting a single number like sound intensity level in the case of marine mammals to the seemingly impossible task of imaging the complex bottom in a hostile ocean environment. Some of the latest approaches to solving acoustical processing problems including sophisticated Bayesian processors in architectural acoustics, iterative flaw removal processing for non-destructive evaluation, time-reversal imaging for buried objects and time-reversal receivers in communications as well as some of the exciting breakthroughs using so-called blind processing techniques for deconvolution are discussed. Processors discussed range from the simple to the sophisticated as dictated by the particular application. It is shown how processing techniques are crucial to extracting the required information for success in the underlying application.
On the behavior of solutions obtained by the HELS method inside the minimum sphere
Authors: Semenova Tatiana , Wu Sean F.
Published: October 2013

Previous experimental and numerical studies have shown that the Helmholtz equation least-squares (HELS) method [Wu, J. Acoust. Soc. Am. 107, 2511–2522 (2000)] can be an effective methodology for nearfield acoustical holography. However, results also demonstrated that the accuracy of reconstruction on a highly nonspherical surface might be unsatisfactory. The validity of the HELS method on these highly nonspherical surfaces has become a controversial topic, just like the Rayleigh hypothesis for acoustic scattering on a corrugated surface. In this paper, the validity of solutions obtained by the HELS method will be examined for acoustic radiation from an infinite column with a rectangular cross section. The acoustic field inside the minimal circle that encircles the column will be reconstructed. The convergence of solutions both on the surface and inside the minimum circle will be checked. Moreover, a modified HELS formulation will be used, which describes the acoustic pressure as a superposition of both out-going and in-coming spherical waves. The nature of the solutions thus obtained will be examined, and the dependence of convergence of the HELS solution on the validity of the Rayleigh hypothesis and on the measurement locations will be investigated. [Work supported by NSF.]
Convention facility acoustical design
Authors: Marsh David
Published: October 2013

Convention centers usually include a variety of meeting spaces with widely varying acoustical requirements. Exhibit halls with several hundred thousand square feet of contiguous space are common along with divisible ballrooms and smaller meeting rooms. Some even have upscale boardrooms with built-in audio–video capabilities rivaling modern corporate facilities. This paper provides an overview of appropriate acoustical design criteria for typical occupied spaces in a convention center in terms of background noise levels, sound isolation between adjacent spaces, and acceptable reverberation times. Examples are given from real-world projects.
Minehunting sonar system research and development
Authors: Ferguson Brian
Published: October 2013

Sea mines have the potential to threaten the freedom of the seas by disrupting maritime trade and restricting the freedom of maneuver of navies. The acoustic detection, localization, and classification of sea mines involves a sequence of operations starting with the transmission of a sonar pulse and ending with an operator interpreting the information on a sonar display. A recent improvement to the process stems from the application of neural networks to the computed aided detection of sea mines. The advent of ultrawideband sonar transducers together with pulse compression techniques offers a thousandfold increase in the bandwidth-time product of conventional minehunting sonar transmissions enabling stealth mines to be detected at longer ranges. These wideband signals also enable mines to be imaged at safe standoff distances by applying tomographic image reconstruction techniques. The coupling of wideband transducer technology with synthetic aperture processing enhances the resolution of side scan sonars in both the cross-track and along-track directions. The principles on which conventional and advanced minehunting sonars are based are reviewed and the results of applying novel sonar signal processing algorithms to high-frequency sonar data collected in Australian waters are presented.
Separation of musical instruments based on amplitude and frequency comodulation
Authors: Jacobson Barry D. , Cauwenberghs Gert , Quatieri Thomas F.
Published: October 2013

In previous work, amplitude comodulation was investigated as a basis for monaural source separation. Amplitude comodulation refers to similarities in amplitude envelopes of individual spectral components emitted by particular types of sources. In many types of musical instruments, amplitudes of all resonant modes rise/fall, and start/stop together during the course of normal playing. We found that under certain well-defined conditions, a mixture of constant frequency, amplitude comodulated sources can unambiguously be decomposed into its constituents on the basis of these similarities. In this work, system performance was improved by relaxing the constant frequency requirement. String instruments, for example, which are normally played with vibrato, are both amplitude and frequency comodulated sources, and could not be properly tracked under the constant frequency assumption upon which our original algorithm was based. Frequency comodulation refers to similarities in frequency variations of individual harmonics emitted by these types of sources. The analytical difficulty is in defining a representation of the source which properly tracks frequency varying components. A simple, fixed filter bank can only track an individual spectral component for the duration in which it is within the passband of one of the filters. Alternatives are therefore explored which are amenable to real-time implementation.
Separation of harmonic sounds using multipitch analysis and linear models for the overtone series
Authors: Virtanen Tuomas , Klapuri Anssi
Published: October 2013

A signal processing method for the separation of concurrent harmonic sounds is described. The method is based on a two-stage approach. First, a multiple fundamental frequency estimator is applied to find initial sound parameters which are reliable, but inaccurate and static. Second, time-varying sinusoidal parameters are estimated in an iterative algorithm. The harmonic structure is retained by keeping the frequency ratio of overtones constant over time. Overlapping harmonic components are resolved using linear models for the overtone series. In practice, the models retain the spectral envelope continuity of natural sounds. Simulation experiments were carried out using generated test signals, which were random mixtures of two to six notes from recorded natural instruments. The system is able to produce meaningful results in all polyphonies, the quality of separated sounds gradually degrading along with the polyphony. Some denoising algorithms were applied to suppress nonstationary noise component, such as drums in real-world music signals. However, the usability of the system for real musical signals is still quite limited.
Effect of thermal conductivity on acoustic streaming in a channel
Authors: Hamilton Mark F. , Ilinskii Yurii A. , Zabolotskaya Evgenia A.
Published: October 2013

Acoustic streaming generated in a viscous fluid with thermal conductivity is considered analytically and numerically. The fluid is confined to a channel formed by two parallel plates. The streaming pattern is similar to that in a viscous fluid without thermal conductivity. Specifically, an acoustic standing wave generates inner and outer streaming vortices that circulate in opposite directions. The outer vortex resembles Rayleigh streaming in a viscous fluid without thermal conductivity, and the inner vortex, confined mostly to the boundary layer region, is due to viscous stresses near the wall. Thermal conductivity is found to increase the streaming velocity in a gas by as much as 20%. Dependence of the viscosity coefficient on temperature is also taken into account, and this effect is found to reduce the streaming velocity. However, the first effect dominates the second, and therefore the net result is larger streaming velocities than in the absence of both effects. An analytical expression for acoustic streaming in both narrow and wide channels was derived and used to investigate the two effects individually. The analytical solution is in close agreement with numerical results based on the fully nonlinear equations of motion. [Work supported by ONR.]
Perceptual effects in auralization of virtual rooms
Authors: Kleiner Mendel , Larsson Pontus , Vastfjall Daniel , Torres Rendell R.
Published: October 2013

By using various types of binaural simulation (or ‘‘auralization’’) of physical environments, it is now possible to study basic perceptual issues relevant to room acoustics, as well to simulate the acoustic conditions found in concert halls and other auditoria. Binaural simulation of physical spaces in general is also important to virtual reality systems. This presentation will begin with an overview of the issues encountered in the auralization of room and other environments. We will then discuss the influence of various approximations in room modeling, in particular, edge- and surface scattering, on the perceived room response. Finally, we will discuss cross-modal effects, such as the influence of visual cues on the perception of auditory cues, and the influence of cross-modal effects on the judgement of ‘‘perceived presence’’ and the rating of room acoustic quality.
Effects of reverberation on speech segregation
Authors: Culling John F. , Toh Chaz Yee , Hodder Kathryn I.
Published: October 2013

Perceptual separation of speech from interfering noise using binaural cues and fundamental frequency (F0) differences is disrupted by reverberation [Plomp, Acoustica 34, 200–211; Culling et al., Speech Commun. 14, 71–96]. Culling et al. found that the effect of F0 differences on vowel identification was robust in reverberation unless combined with even subtle F0 modulation. In the current study, speech reception thresholds (SRTs) were measured against a single competing voice. Both voices were either monotonized or normally intonated. Each came from recordings of the same voice, but interfering sentences were feminized (F0 increased 80%; vocal-tract length reduced 20%). The voices were presented from either the same or from different locations within anechoic and reverberant virtual rooms of Culling et al. In anechoic conditions, SRTs were lower when the voices were spatially separated and/or intonated, indicating that intonated speech is more intelligible than monotonous speech. In reverberant conditions (T60=400 ms), SRTs were higher, with no differences between the conditions. A follow-up experiment introduced sentences with inverted F0 contours. While acceptable in quiet, these sentences gave higher SRTs in all conditions. It appears that reverberant conditions leave intonated speech intelligible, but make it unseparable, while monotonous speech remains separable but is unintelligible.
Effects of natural and artificial spatialization cues on segregation
Authors: de Cheveigné Alain , Gretzky Reinhard , Baskind Alexis , Warusfel Olivier
Published: October 2013

A series of experiments was performed to better understanding factors that determine the clarity or ‘‘transparency’’ of sound scenes, particularly those created by artificial means. Using a paradigm proposed by C. J. Darwin and R. W. Hukin [J. Exp. Psychol. 25, 617–629], subjects were presented with a target sentence containing one word (chosen among two) that they had to report. Simultaneously they were presented with a distractor sentence containing a second word, temporally aligned with the first. In the absence of segregation cues, subjects scored 50% correct on average. A higher score indicated that simultaneous segregation and sequential grouping mechanisms were both effective. Stimuli were presented by headphones using individual or dummy-head head-related transfer functions (HRTFs), and various combinations of source positions, room effects, and restitution techniques. [Work supported by the Cognitique Programme of the French Ministry of Research and Technology.]
The influence of reverberation on spatial release of masking in consonant identification
Authors: Devore Sasha , Shinn-Cunningham Barbara G. , Durlach Nathaniel I. , Colburn H. Steven
Published: October 2013

This study investigates how reverberation influences spatial release of masking in consonant identification. HRTFs were measured in two rooms with different reverberation characteristics from sources directly in front of and to the right of the listener at a distance of 1 m. These two sets of room HRTFs and pseudo-anechoic HRTFs (time-windowing out the reverberation) were used to simulate a speech target and speech-shaped masker over headphones in two spatial configurations. In both configurations, the target was directly ahead of the listener; the masker was either directly ahead or 45° to the right of the listener. Subjects were asked to identify the initial or final consonant (one of 12 stop and fricative consonants) in CVC nonsense syllables for different target-to-masker levels. Each spatial configuration and room condition was tested using binaural, monaural-left, and monaural-right headphone presentations. Consonant confusion matrices are analyzed to determine how specific acoustic features of consonants in different frequency regions are affected by reverberation and noise. Comparisons across the different conditions reveal how binaural and spatial cues influence consonant identification and how different acoustic environments affect both monaural and binaural performance. [Work supported in part by AFOSR Grant Nos. F49620-01-1-0005 and F49620-98-1-0442.]
Effect of mean flow on the analysis of noise and vibration in hydraulic lines
Authors: Hastings Mardi C.
Published: October 2013

Recent reports in the literature conclude that convective terms due to mean flow should be included in the analysis to predict noise and vibration in steel hydraulic lines containing compliant hoses [M. L. Munjal and P. T. Thawani, Noise Control Eng. J. 45, 235–242 (1997)]. In this study analytical predictions for noise and vibration were compared with experimental data obtained from a transmission line excited by a 10-vane pump source. The predictions were based on a transfer matrix analysis that included the fluid–structure interaction but neglected the convective terms associated with mean flow [M. C. Hastings and C.-C. Chen, J. Passenger Cars, SAE Trans. 102, 1762–1767 (1994)]. Experimental data and analytical predictions were in excellent agreement. These results indicate that mean flow convective terms are negligible and do not need to be included in the analysis of noise and vibration occurring in relatively low-power hydraulic assist systems (such as automotive power steering). A comparison of analytical results with and without mean flow convective terms for higher-power hydraulic systems indicates when the convective terms are needed for an accurate prediction of noise and vibration.
Telephone speech enhancement for hearing-impaired listeners using multi-channel compression
Authors: Natarajan Harikrishna P. , Krishnamurthy Ashok K. , Feth Lawrence L.
Published: October 2013

Elderly listeners with sensorineural hearing impairment usually have difficulty with telephone communication if they do not use a hearing aid or telephone amplifier. In many cases, these devices may not be readily available or maybe uncomfortable to use. We describe here an alternative strategy to enhance the speech by pre-processing the signal before sending it over the telephone line. The pre-processing is based on a multi-channel compression algorithm that modifies the critical band spectrum of the speech signal to bring it within the dynamic hearing range of the listener, while also compensating for the effects of the telephone line. We describe the algorithm and the results of a simulation study in which the Articulation Index (AI) is used to measure the intelligibility enhancement, and the Glasberg, Moore and Stone (1999) model is used to simulate the effects of the hearing loss.
Effect of preceding speech on nonspeech sound perception
Authors: Stephens Joseph D. , Holt Lori L.
Published: October 2013

Data from Japanese quail suggest that the effect of preceding liquids (/l/ or /r/) on response to subsequent stops (/g/ or /d/) arises from general auditory processes sensitive to the spectral structure of sound [A. J. Lotto, K. R. Kluender, and L. L. Holt, J. Acoust. Soc. Am. 102, 1134–1140 (1997)]. If spectral content is key, appropriate nonspeech sounds should influence perception of speech sounds and vice versa. The former effect has been demonstrated [A. J. Lotto and K. R. Kluender, Percept. Psychophys. 60, 602–619 (1998)]. The current experiment investigated the influence of speech on the perception of nonspeech sounds. Nonspeech stimuli were 80-ms chirps modeled after the F2 and F3 transitions in /ga/ and /da/. F3 onset was increased in equal steps from 1800 Hz (/ga/ analog) to 2700 Hz (/da/ analog) to create a ten-member series. During AX discrimination trials, listeners heard chirps that were three steps apart on the series. Each chirp was preceded by a synthesized /al/ or /ar/. Results showed context effects predicted from differences in spectral content between the syllables and chirps. These results are consistent with the hypothesis that spectral contrast influences context effects in speech perception. [Work supported by ONR, NOHR, and CNBC.]
Audiovisual integration in the absence of a McGurk effect
Authors: Brancazio Lawrence , Miller Joanne L. , Mondini Michèle
Published: October 2013

The McGurk effect, a change in perceived place of articulation due to an incongruent visual stimulus (e.g., auditory /pi/ with visual /ti/ perceived as /ti/), demonstrates the contribution of vision to speech perception. Interestingly, in a given experiment the McGurk effect typically does not occur on every trial. We investigated whether non-McGurk trials result from a failure to perceptually integrate auditory and visual information by simultaneously manipulating visual place of articulation and visual speaking rate. Previous work [Green and Miller, Percept. Psychophys. 38, 269–276 (1985)] has shown that the boundary along an auditory /bi/–/pi/ voice-onset-time (VOT) continuum occurs at a longer VOT when the auditory stimulus is paired with a slow rather than a fast visual /pi/. We paired stimuli from an auditory /bi/–/pi/ continuum with fast and slow versions of a visual /ti/, and subjects identified each item as /b/, /p/, /d/, or /t/. We found a rate effect on McGurk trials, with the /d/–/t/ boundary occurring at a longer VOT when the visual stimulus was slow rather than fast. Importantly, we found a comparable rate effect for the /b/–/p/ boundary on non-McGurk trials. This indicates that audiovisual integration occurs even in the absence of a McGurk effect. [Work supported by NIH/NIDCD.]
Models of vowel perception
Authors: Molis Michelle R.
Published: October 2013

The debate continues regarding the efficacy of formant-based versus whole spectrum models of vowel perception. Categorization data were obtained for a set of synthetic steady-state vowels and were used to evaluate both types of models. The models tested included various combinations of formant frequencies and amplitudes, principal components derived from critical-band spectra, and perceptually scaled LPC cepstral coefficients. The stimuli were 54 five-formant synthesized vowels that varied orthogonally in F2 (1081–2120 Hz) and F3 (1268–2783 Hz) frequency in equal 0.8 Bark steps. Fundamental frequency contour, F1 (455 Hz), F4 (3250 Hz), F5 (3700 Hz), and duration (225 ms) were held constant across all stimuli. Twelve speakers of American English (Central Texas dialect) categorized the stimuli as the vowels /I/, /U/, or /ε/. Results indicate that formant frequencies provided the best account of the data only when nonlinear terms were also included in the analysis. The critical-band principal components also performed reasonably well. While the principle of parsimony would suggest that formant frequencies offer the most appropriate description of vowels, the relative success of a richer, more flexible and more neurophysiologically implementable model still must be taken into consideration. [Work supported by NIDCD.] a)Formerly at Univ. of Texas at Austin.
Optimization of a three-dimensional Simplex procedure for speech in noise
Authors: Franck Bas A. M. , Koopman Jan , Dreschler Wouter A.
Published: October 2013

In this study we investigated means to enhance the reliability of a three-dimensional Simplex optimization procedure. The algorithms used to span the three dimensions are noise reduction, spectral tilting, and spectral enhancement, respectively. We considered the influence of an adaptive step size, different starting points, different types of noise, and the effect of an impaired spectral resolution. The subjects were asked to evaluate two consecutive, differently processed sentences on the evaluation criterion listening comfort. Three different noise types were used. We asked the normal-hearing subjects to listen to the stimuli with or without simulated impaired auditory filtering. The impairment was simulated by application of spectral smearing. For the subjects with normal filtering we applied a signal-to-noise ratio (S/N) of 0 dB. For the subjects with simulated impaired frequency resolution we applied S/N=5 dB. Due to several methodological innovations (especially when the step size is adaptive), the reliability of the Simplex procedure increased. Moreover, the reliability depends on the algorithms included and on the type of noise used.
Scattering of plane evanescent waves by cylindrical shells and wave vector coupling conditions for exciting flexural waves
Authors: Marston Philip L.
Published: October 2013

The coupling of sound to buried targets can be associated with acoustic evanescent waves when the sea bottom is smooth. To understand the excitation of flexural waves on buried shells by acoustic evanescent waves, the partial wave series for the scattering is found for cylindrical shells at normal incidence in an unbounded medium. The formulation uses the simplifications of thin-shell dynamics. In the case of ordinary waves incident on a shell, a ray formulation is available to describe the coupling to subsonic flexural waves [P. L. Marston and N. H. Sun, J. Acoust. Soc. Am. 97, 777–783 (1995)]. When the incident wave is evanescent, the distance between propagating plane wavefronts is smaller than the ordinary acoustical wavelength at the same frequency and the coupling condition for the excitation of flexural waves on shells or plates is modified. Instead of matching the flexural wave number with the propagating part of the acoustic wave number only at the coincidence frequency, a second low-frequency wave number matching condition is found for highly evanescent waves. Numerical evaluation of the modified partial-wave-series appropriate for an evanescent wave is used to investigate the low-frequency coupling of evanescent waves with flexural wave resonances of shells.
Low-frequency propagation and scattering from proud targets
Authors: Sammelmann Gary
Published: October 2013

This discussion focuses on the effects of placing a large aspect ratio target in the vicinity of an interface. It uses the spheroidal T-matrix description of scattering from a large aspect ratio target developed by Hackman to describe the scattering from large aspect ratio solids. A comparison of the effects of the interface on the form function and the directivity pattern of the scattering from the target at low frequencies is made. At sufficiently low frequencies the scattering from the target and its image add coherently, giving rise to a 6-dB increase in the form function of the proud target versus a volume target. At higher frequencies the scattering oscillates between constructive and destructive interface of the scattering from the target and its image. [Work supported by Codes 321OE and 32MIW of the ONR.]
Rough surface reverberation of surface-generated ambient noise in a nonuniform ocean waveguide
Authors: Liu Jin-Yuan , Hsueh Ping-Chang , Tsai Shern-Hsiung
Published: October 2013

Rough surface reverberation of surface-generated ambient noise in an ocean waveguide with a transition layer of continuously varying density and sound speed is considered in this analysis. The general formulation is based upon a theory for rough surface scattering in a stratified medium, in conjunction with the analytical solutions for wave equation in a nonuniform medium intended to simulate the sediment layer in a realistic ocean. Numerical results for various parameters for the reverberation field were obtained and analyzed. The model is intended to refine the previously proposed one [J.-Y. Liu, H. Schmidt, and W. A. Kuperman, J. Acoust. Soc. Am. 93, 753–769 (1993)], and will serve as a basis for analysis of the data obtained in a recent experiment. [Work supported by NSC of ROC.]
Passive phase-conjugate signaling for autonomous systems in the ocean
Authors: Hursky Paul , Siderius Martin , Porter Michael B. , McDonald Vincent K. , Rice Joseph A.
Published: October 2013

Acoustic communications is an attractive way to wirelessly network distributed assets, such as UUVs, autonomous sensors, and other systems in the ocean. We have been experimenting with modulation techniques based on passive phase-conjugation (PPC), which essentially uses the channel itself to refocus multipath arrivals, that would otherwise cause inter-symbol interference. We have also used Gold codes to isolate snapshots of the channel response, prior to the PPC operation. Our scheme can be used point-to-point, without arrays. This is a very simple demodulation technique which nevertheless is coherent and can overcome signal distortion due to multipath. Simplicity is very important, because simple algorithms consume less battery power, allowing the node to operate for a longer time in the water. We discuss details of our algorithms, including synchronization and Doppler compensation, and present the results of testing them in a variety of environments, including four SignalEx tests and the ASCOT01 experiment. We compare our PPC signaling scheme to frequency-hopped frequency-shift keying and direct sequence spread spectrum modulation using a rake receiver. All of these signaling schemes are being considered for low power, lightweight applications.
Time reversal, back propagation, matched field processors, correlation receivers, and the principles of radar/sonar signal design
Authors: Baggeroer Arthur
Published: October 2013

Time reversal and backpropagation have been demonstrated in several experiments. Similarly, while matched field processing (MFP) differs in terms of implentation—experimental vs computed replicas—both have two common properties: (i) they are based on a single, spatially coherent signal; and (ii) the conjugate transpose of the Green’s function and replica correlation are identical for self-adjoint systems. Hence, the principles for focusing and ambiguity plane properties of these processors are virtually identical to those for correlation receivers. The principles of optimal signal design for correlation receivers were the subject of much research for radar/sonar systems four decades ago and many of them seem to have been neglected in the analysis of time reversal, back propagation, and matched field processors. For example, time reversal from a point, a line array, or a random array of scatterers are duals of an impulse, a frequency modulated, and a pseudo-random noise signal, respectively. The equivalence and consequences of the time-bandwidth products for signals and array length wave number spread are demonstrated. The impact of sidelobes and multipath spread can be predicted. The generalizations of the important radar/sonar uncertainty principle, however, have yet been not demonstrated. This presentation reviews these optimal signal design principles and applies them to time reversal and MFP.
Reverberation and the Franssen effect
Authors: Whitmer William M. , Yost William A. , Dye Raymond H.
Published: October 2013

The Franssen effect, in which the location of a sudden-onset (transient) tone can occlude the location of a contralateral slow-onset (steady-state) tone, has been previously shown to occur only in a reverberant space. The nature of the reverberation required—how much and what kind—for the effect to occur has yet to be determined. To explore the role of reverberation in the Franssen effect, listeners were asked to identify the location(s) of transient/steady-state pure tones with frequencies ranging from 250–4000 Hz in a variety of real and virtual contexts: (a) limited (single reflective panel) reverberant enclosure; (b) discrete reflections in an anechoic room; (c) mannequin recordings of a reverberant enclosure presented over headphones; (d) impulse-response filtered signals presented over headphones; and (e) simulated 3-D reverberation via consumer-grade software. Results indicate that the Franssen effect can be a simple, effective means to judge the verisimilitude of a virtual environment. [Work supported by NIDCD.]
School cafeteria noise—The impact of room acoustics and speech intelligibility on children’s voice levels
Authors: Bridger Joseph F.
Published: October 2013

The impact of room acoustics and speech intelligibility conditions of different school cafeterias on the voice levels of children is examined. Methods of evaluating cafeteria designs and predicting noise levels are discussed. Children are shown to modify their voice levels with changes in speech intelligibility like adults. Reverberation and signal to noise ratio are the important acoustical factors affecting speech intelligibility. Children have much more difficulty than adults in conditions where noise and reverberation are present. To evaluate the relationship of voice level and speech intelligibility, a database of real sound levels and room acoustics data was generated from measurements and data recorded during visits to a variety of existing cafeterias under different occupancy conditions. The effects of speech intelligibility and room acoustics on childrens voice levels are demonstrated. A new method is presented for predicting speech intelligibility conditions and resulting noise levels for the design of new cafeterias and renovation of existing facilities. Measurements are provided for an existing school cafeteria before and after new room acoustics treatments were added. This will be helpful for acousticians, architects, school systems, regulatory agencies, and Parent Teacher Associations to create less noisy cafeteria environments.
Light-weight low-frequency loudspeaker
Authors: Corsaro Robert , Tressler James
Published: October 2013

In an aerospace application, we require a very low-mass sound generator with good performance at low audio frequencies (i.e., 30–400 Hz). A number of device configurations have been explored using various actuation technologies. Two particularly interesting devices have been developed, both using ‘‘Thunder’’ transducers (Face Intl. Corp.) as the actuation component. One of these devices has the advantage of high sound output but a complex phase spectrum, while the other has somewhat lower output but a highly uniform phase. The former is particularly novel in that the actuator is coupled to a flat, compliant diaphragm supported on the edges by an inflatable tube. This results in a radiating surface with very high modal complexity. Sound pressure levels measured in the far field (25 cm) using only 200-V peak drive (one-third or its rating) were nominally 74 6 dB over the band from 38 to 330 Hz. The second device essentially operates as a stiff low-mass piston, and is more suitable for our particular application, which is exploring the use of active controlled surface covers for reducing sound levels in payload fairing regions. [Work supported by NRL/ONR Smart Blanket program.]
Matching untrained singers using dynamic time warping
Authors: Kahrs Mark
Published: October 2013

Untrained singers lack precise control of pitch and timing, therefore any successful music search algorithm must accommodate inaccuracies in both dimensions. Furthermore, finding key signatures in untrained singers is not easy (or even plausible in some cases). Contours can be formed by up and down intervals and then matched against the database. An experimental system is described that matches an acoustic vocal input against a library of monophonic musical scores and returns a list of matched results. While partially successful, errors result from the detection of pitch on and off times as well as from pitch detection errors. Pitch detection is easiest during vowels, but the transitions from vowel to consonant can cause problems. In spite of these inaccuracies, the prototype algorithm using dynamic time warping (a dynamic programming algorithm) exceeded 95% accuracy for nine different singers for a simple tune library. However, errors computed by a difference metric can result in close misses, even for very dissimilar songs. A better metric is needed to clearly illustrate song differences.
Error signal sensing for active noise control in turbofan engine nacelles
Authors: Walker Bruce , Hersh Alan , Rice Edward
Published: October 2013

Blade passage harmonic tones are a significant component of turbofan engine noise under approach conditions. Active noise control (ANC) offers a tool for suppression of these tones, with the caveat that interference patterns in the sound field can lead to poor or even negative correlation between in-duct sound pressures and radiated sound power. Various modal decomposition and wavenumber separation schemes have been investigated in attempt to overcome this obstacle. The paper will discuss methods that have been applied to design and implementation of microphone arrays and signal processing techniques to provide unambiguous error signals for ANC systems in these complex acoustical environments and will present results of computer simulations and field tests. [Work sponsored by NASA Glenn and NASA Langley Research Centers.]
Development of shaped PVDF sensors for the measurement of volume displacement
Authors: Zellers Brian , Rozema Randall , Naghshineh Koorosh
Published: October 2013

To attenuate the sound radiated from a vibrating structure, active control can be implemented. When this radiated sound is in the lower frequencies, a reduction of the surface volume displacement of the vibrating structure is suitable for achieving attenuation within these lower frequencies. The sound power emitted from the surface is directly related to the surface volume displacement of the vibrating structure. Traditionally, measurement of the vibration of the structural surface is done using multiple point sensors. Previous work has shown that for a one-dimensional vibrating structure a single sensor made of polyvinylidene fluoride (PVDF) can be used in place of point sensors. Additionally, PVDF sensors have been developed to sense the surface volume displacement in a local area of interest by spanning only that area. However, it has been observed that at higher frequencies both the total and local sensors tend to exhibit reduced accuracy. Thus, a series of experiments was conducted to determine if the PVDF stress/charge coefficient varies as a function of frequency. Additionally, a simulation program was developed to predict sensor charge output for arbitrarily shaped sensors. This simulation, which allows us to develop sensor shapes and predict sensor placement errors, was verified experimentally.
The future of infant speech perception research: Gotta wear shades
Authors: Walley Amanda C.
Published: October 2013

The future of infant speech perception research is very bright in large part because of the legacy left to us by Peter W. Jusczyk. I will selectively review a number of studies, many of them conducted by Peter and his colleagues, that I consider to be especially interesting and/or provocative and that have helped to advance our understanding of the development of speech perception. For example, in two fairly recent studies, 4.5 to 6-month-old infants were shown to display some incipient spoken word recognition ability [D. R. Mandel et al., Psychol. Sci. 6, 314–317 (1995); R. Tincoff and P. W. Jusczyk, ibid. 10, 172–175 (1999)]. In effect, these studies have smashed the sound-meaning barrier that was, according to traditional estimates, placed somewhere between 9 and 13 months of age and have important implications for current developmental theory. This work is also illustrative of one trend apparent in recent research, namely, to accord greater attention to infants’ perception of ecologically-relevant stimuli and/or their perception of speech stimuli in ecologically-relevant contexts. Still, there is much to be learned about spoken word recognition beyond infancy and before adulthood if we are to arrive at a complete picture of this core aspect of spoken language processing.
Talker variation and word learning
Authors: Hollich George , Jusczyk Peter , Brent Michael
Published: October 2013

While infants must go beyond talker-specific information in recognizing a given word, regardless of the talker, they must also process talker-specific information in order to extract meaning from a particular sound source. Otherwise, for example, they could never recognize whether [hct] referred to a talker’s pronunciation of hot, hut, or even hat. This poster suggests that not only do infants process talker-specific information, but they also make use of it both to extract invariant properties in learning a new word and in recognizing talker-specific tokens faster. Using the splitscreen preferential looking paradigm, two studies were conducted that examined how talker-specific properties and variation among talkers could facilitate word learning. Results of study 1 indicated that word learning was facilitated in the case where infants heard different talkers. Thus, talker variation is critical for the extraction of invariant properties of a word. However, the results of study 2 indicated that talker-specific properties were encoded and used to help these infants recognize and learn the referents of these words. Given this evidence, it is suggested that infants appear to be using talker-specific information to form abstract representations of the invariant properties of words.