bluuebunny/crossref_metadata_2025 huggingface dataset:
Autoregressive wave number estimation technique for range-dependent shallow-water waveguides with abrupt environmental variations
Authors: Souza Luiz L. , Frisk George V.
Published: October 2013

We extend the technique described by Becker and Frisk [J. Acoust. Soc. Am. 111, 2388 (2002)], which uses an autoregressive spectral (AR) estimator to infer characteristic modal wave numbers in shallow-water, range-dependent waveguides. Their method offers high resolution over relatively small range apertures, an improvement over the short-range Fourier transform technique originally employed. When the environment varies continuously, e.g., due to a sloping bottom, the assumption of constant modal eigenvalues over a range aperture, intrinsic to the AR estimator, becomes invalid, resulting in increased bias. Abrupt changes in the modal content of the field can be detected, but not measured. The present work extends this method by allowing the modal structure to change with range on a sample by sample basis, and yields AR coefficients that are functions of range, analogous to the time-varying AR (TVAR) estimators used for nonstationary time series analysis. By processing the data twice, with increasing and decreasing ranges (forward and backwards), and combining the results, both slow and abrupt changes in the eigenvalues can be measured.
Why seek fusion from cavitation: Molecular dynamic simulations and a detector capable of time correlated single neutron counting
Authors: Camara Carlos , Cousins Robert , Naranjo Brian , Putterman Seth , Merriman Barry , Ruuth Steven
Published: October 2013

The blackbody spectra, and similar sonoluminescence intensities of He and Xe bubbles suggest that the interior of a sonoluminescing bubble is highly stressed and dense. Molecular dynamic simulations indicate interior temperatures which are enhanced by thermal conduction and can approach 1 MK. Furthermore the gas passes through states where the mean free path is larger than the distance over which temperature varies and so calls into question the value of theories based on hydrodynamics. To search for rare fusion events a neutron detector with 25% total discriminated quantum efficiency has been built. It can time stamp neutron arrival and sonoluminescence to better than 1 ns and record tracks on the fly. [Work supported by DARPA.]
Chemical control of single bubble cavitation
Authors: Didenko Yuri T. , Suslick Kenneth S.
Published: October 2013

Sonochemistry would be ideally studied with a single bubble with known size pulsating in known acoustic pressure field. Single bubble cavitation provides the means to make such studies. The promise that single bubble cavitation brought to the quantitative measurements of chemical activity of cavitation, however, has not been previously fulfilled due to the very small amount of reacting gas within a single bubble (typically <10–13 moles). We have now quantitated the chemical reactions inside a single cavitating bubble and established an inventory of energy dissipation during bubble collapse. The yields of nitrite ions, hydroxyl radicals, and photons from a single cavitation bubble have now been measured, and the first true measures of energy efficiencies during acoustic cavitation have been determined. The energy efficiency of the formation of hydroxyl radicals from single bubble is comparable to that in multibubble cavitation. The energy efficiency of light emission, however, is much higher for SBSL. The observed rate of nitrite formation is in good agreement with the calculated diffusion rate of nitrogen into the bubble. The temperatures reached during single bubble cavitation in liquids with significant vapor pressures will be substantially limited by the endothermic chemical reactions of the polyatomics inside the collapsing bubble.
Infants’ detection in the presence of masker uncertainty
Authors: Leibold Lori J. , Werner Lynne A.
Published: October 2013

Most adults have difficulty detecting a fixed-frequency pure tone when the frequency content of the masker is varied on each presentation. This informational masking is thought to be the result of masker uncertainty. The purpose of the present study was to examine infant thresholds under conditions of masker uncertainty. The subjects were 7- to 9-month-old infants and 18- to 30-year-old adults with no risk factors for hearing loss. Detection thresholds were measured for a 300-ms, 1-kHz tone in the presence of a noise band (300–3000 Hz), a two-component constant-frequency masker, or a two-component random-frequency masker. Maskers repeated at 300-ms intervals throughout the testing session at an overall level of 60 dB SPL. The signal was played synchronously with one interval of the repeating masker. An observer-based testing method was used. Following training to 80%-correct criterion, detection thresholds were determined adaptively. For adults, the random-frequency masker condition produced the greatest amount of masking, ranging from 30–60 dB. For infants, the random-frequency masker also produced the greatest amount of masking, ranging from 60–80 dB. In contrast to the adults, infant thresholds were also elevated in the constant-frequency masker condition, ranging from 35–55 dB across subjects. [Work supported by NIDCD RO1 DC000396 and NRSA DC006122.]
Introductory remarks on neural modeling in speech perception research
Authors: Guenther Frank H.
Published: October 2013

This talk will provide an overview of several neural models that have been used to address experimental data on speech perception, including psychophysical and neurophysiological data. Although these models differ greatly in their scope and mathematical detail, all are geared toward providing a better understanding of the neural mechanisms underlying the processing of speech sounds.
Resonant cortical dynamics of speech perception
Authors: Grossberg Stephen
Published: October 2013

What is the neural representation of a speech code as it evolves in time? How do listeners integrate temporally distributed phonemic information into coherent representations of syllables and words? How does the brain extract invariant properties of variable-rate speech? This talk describes a neural model that suggests answers to these questions, while quantitatively simulating speech and word recognition data. The conscious speech and word recognition code is suggested to be a resonant wave, and a percept of silence a temporal discontinuity in the rate that resonance evolves. A resonant wave emerges when sequential activation and storage of phonemic items in working memory provides bottom-up input to list chunks that group together sequences of items of variable length. The list chunks compete and winning chunks activate top-down expectations that amplify and focus attention on consistent working memory items, while suppressing inconsistent ones. The ensuing resonance boosts activation levels of selected items and chunks. Because resonance occurs after working memory activation, it can incorporate information presented after intervening silence intervals, so future sounds can influence how we hear past sounds. The model suggests that resonant dynamics enable the brain to learn quickly without suffering catastrophic forgetting, as described within Adaptive Resonance Theory.
Using neural modeling and functional neuroimaging to study the neural basis of auditory object processing
Authors: Horwitz Barry , Husain Fatima T.
Published: October 2013

The neural basis of auditory object processing in the human cerebral cortex was investigated by combining neural modeling and functional neuroimaging. We developed a large-scale, neurobiologically realistic network model of auditory pattern recognition that relates neuronal dynamics of cortical auditory processing of frequency-modulated (FM) sweeps to functional neuroimaging data obtained using functional magnetic resonance imaging (fMRI). FM sweeps are ubiquitous in animal communication. Areas included in the model extend from primary auditory to prefrontal cortex. The electrical activities of the model neuronal units were constrained to agree with data from the neurophysiological literature regarding FM sweep perception. A fMRI experiment using stimuli and tasks similar to those used in our simulations was performed. The regional integrated synaptic activities of the model were used to determine simulated regional fMRI activities, and generally agreed with the experimentally observed fMRI data. Our results demonstrate that the model is capable of exhibiting the salient features of both electrophysiological neuronal activities and fMRI values that are in agreement with empirically observed data. These findings provide support for our hypotheses concerning how auditory objects are processed by primate neocortex. This type of approach offers the potential for understanding the neural basis of human speech perception.
Imaging with eigenfunctions of a scattering operator
Authors: Waag Robert C.
Published: October 2013

An inverse scattering method using eigenfunctions of a scattering operator is reviewed. The framework of the method encompasses the use of eigenfunctions for focusing and quantitative image reconstruction in arbitrary media. Scattered acoustic fields are described using an operator. The eigenfunctions correspond to far-field patterns of an effective source distribution. Incident-wave patterns specified by the eigenfunctions focus on the distribution. The eigenfunction focusing properties are employed to reconstruct an unknown scattering medium by using products of numerically calculated fields defined in terms of eigenfunctions. A linearized version of the method is equivalent to the filtered-backprojection method for Born inversion. The full range of frequencies present in an incident pulse waveform can be used in the method. The method is examined by using both calculated and measured data. In the calculations, an exact solution for scattering from nonconcentric cylinders was employed to obtain the scattered field. In the measurements, a novel ring-transducer system was used to obtain the incident and total fields. The results of simulations and experiments show that the method is accurate for objects large compared to the incident pulse center-frequency wavelength and that the point resolution of the method is about one-half the center-frequency wavelength.
Vector subspace methods in inverse acoustic wave scattering
Authors: Devaney Anthony J.
Published: October 2013

The problems of detecting, locating, and identifying acoustic scatterers embedded in a known inhomogeneous background from the multistatic data matrix collected from an arbitrary unstructured, mixed sensor array are addressed using the distorted wave Born approximation. Based on this formulation two classes of problems are considered: (i) detecting and locating a finite set of point scatterers, (ii) imaging an arbitrary distribution of point or extended scatterers. Problems of the first class are shown to admit solutions based on the singular value decomposition (SVD) of the multistatic data matrix considered as a linear mapping from the finite vector space of transmitter inputs to the finite vector space of receiver outputs. In this case the SVD of the data matrix is shown to lead directly to generalized time-reversal algorithms that allow super-resolution location estimation. Problems of the second class are also addressed using the SVD but this time it is based on the multistatic data matrix considered as a linear mapping from the Hilbert space of scatterer distributions to the finite vector space of receiver outputs. In this case a generalized form of the filtered backpropagation algorithm is derived and shown to lead to a minimum norm image of the scatterer distribution.
Detection of a weak target in the presence of loud moving noise sources
Authors: Sullivan Edmund J. , Candy James V. , Carey William M.
Published: October 2013

A previous study [J. Acoust. Soc. Am. 112 (2002)] treated the detection of a weak target in an adverse shallow-water environment with ambient noise and ‘‘known’’ interfering ships by applying a model-based adaptive (recursive) technique. The shallow-water environment and noise sources were represented by a normal-mode model directly incorporated into the model-based processor, thereby allowing their effects to be removed from the decision function prior to target detection. This previous work assumed that the interfering ships were motionless. Here, a general state-space framework is developed, that includes a dynamic model of the moving interferers together with the adaptive signal processor in a self-consistent formalism. The decision function is the so-called ‘‘Weighted Sum Square Residual’’ (WSSR), which evolves directly from the innovation sequence of the model-based processor. An example is shown for a low level sound source in Gaussian ambient noise with a strongly interfering moving ship. It is shown further that, upon detection, the innovation sequence can be used as ‘‘prefiltered’’ data providing input that can be used to track the source after its detection.
Model-based source localization in the 8–16 kHz band using the channel impulse response function
Authors: Hursky Paul , Siderius Martin , Porter Michael B. , McDonald Vincent K.
Published: October 2013

SignalEx tests have been conducted in a variety of shallow water coastal environments to relate the performance of acoustic communications systems to the prevailing oceanographic conditions. These tests have typically consisted of using a fixed receiver (one to four elements, spaced for diversity) and a transmitter drifting out to ranges beyond the minimum detectable level. During these tests, waveforms to probe the channel in the 8 to 16 kHz band were transmitted at regular intervals. These probe signals were used to test source localization algorithms at these high frequencies. Model-based source localization at these very high frequencies requires either very accurate modeling or algorithms inherently robust against model mismatch. Although the first couple of multipath arrivals can be stabilized from ping to ping, the later arrivals exhibit rapidly fluctuating amplitudes and times of arrival, due to the motion of the ocean surface, water column variability, and the varying bathymetry as the transmitter drifts in range. Measurements of the time-varying channel response and source localization results using the 8–16 kHz band SignalEx channel probes will be presented from sites at the New England Front and the Coronado Bank off San Diego.
Virtual acoustics for music practice rooms
Authors: Freiheit Ron
Published: October 2013

The use of virtual acoustics has provided a new level of practice experience for the musician. By integrating the sound isolation of music practice rooms with the signal processing of an active acoustic system (with time variant-gain before feedback) musicians can now benefit from the experience of practicing in multiple acoustic environments. Musicians select from various acoustics environments from a typical small practice room to that of a large space such as a sports arena. The variability of the acoustic environment allows the musician to hear clearly their intonation and articulation, which may be difficult to discern in a small practice room. To effectively communicate the various acoustics environments, the musicians must be immersed in the sound field of the active acoustics without being able to discern source locations of the speakers. The system must also be able to support the dynamic range of the musicians without presenting artifacts of its own such as system noise or audible distortion. This paper deals with the design constraints needed to meet these requirements as well the antidotal responses from musicians who have used these environments for practice.
Using miniature signal processing equipment in real-time brass performance
Authors: Plsek Thomas J.
Published: October 2013

Real time signal processing for brass instrument performance has been in use for more than 20 years now. It has been fraught with many problems not the least of which is the complexity, size, and expense of the equipment as well as the acoustical output of the instrument itself. One device which addresses these issues is the new Yamaha ST5: Personal Studio for brass instruments. By combining a Yamaha Pickup Mute, which very effectively minimizes the acoustical output of the instrument, with a battery powered unit small enough to be worn on a belt (ca. 5 in.×3 in.×1 in.), this system enables the performer to use such effects as reverb, delay, chorus, equalizer, pitch shifter, etc., that can be used in a wide variety of situations such as private practice, live concert performances, and recordings. By creatively managing the acoustic instrument and the miniature electronic equipment, a reasonably large array of musical resources become available to the performer enabling him/her to enhance existing performance environments, as well as find and develop new ones.
The use of delay in multitrack production
Authors: Case Alexander U.
Published: October 2013

Delay, inevitable whenever sound propagates through space, is too often the bane of the acoustician’s practice. An audible echo generally relegates a music performance hall—no matter how beautiful it otherwise might sound—to the lowest status. Multitrack music production on the other hand, with its aggressive use of overdubbing, editing, and signal processing, is not bound by those rules of time and space which determine the sound of a hall. In the recording studio, where music is synthesized for playback over loudspeakers, the delay is employed as a powerful, multipurpose tool. It is not avoided. It is in fact embraced. Echoes are used on purpose, strategically, to enhance the loudspeaker listening experience. Moreover, the humble delay is the basis for many nonecho effects. Flanging, chorus, and pitch shifting are delay-based effects regularly used in audio engineering practice. This paper discusses some of the more common delay-based effects, reviewing their technical structure, the psychoacoustic motivation behind them, and the musical value they create.
Music 4C, a multi-voiced synthesis program with instruments defined in C
Authors: Beauchamp James W.
Published: October 2013

Music 4C is a program which runs under Unix (including Linux) and provides a means for the synthesis of arbitrary signals as defined by the C code. The program is actually a loose translation of an earlier program, Music 4BF [H. S. Howe, Jr., Electronic Music Synthesis (Norton, 1975)]. A set of instrument definitions are driven by a numerical score which consists of a series of ‘‘events.’’ Each event gives an instrument name, start time and duration, and a number of parameters (e.g., pitch) which describe the event. Each instrument definition consists of event parameters, performance variables, initializations, and a synthesis algorithmic code. Thus, the synthetic signal, no matter how complex, is precisely defined. Moreover, the resulting sounds can be overlaid in any arbitrary pattern. The program serves as a mixer of algorithmically produced sounds or recorded sounds taken from sample files or synthesized from spectrum files. A score file can be entered by hand, generated from a program, translated from a MIDI file, or generated from an alpha-numeric score using an auxiliary program, Notepro. Output sample files are in wav, snd, or aiff format. The program is provided in the C source code for download.
Inside-in, alternative paradigms for sound spatialization
Authors: Bahn Curtis , Moore Stephan
Published: October 2013

Arrays of widely spaced mono-directional loudspeakers (P.A.-style stereo configurations or ‘‘outside-in’’ surround-sound systems) have long provided the dominant paradigms for electronic sound diffusion. So prevalent are these models that alternatives have largely been ignored and electronic sound, regardless of musical aesthetic, has come to be inseparably associated with single-channel speakers, or headphones. We recognize the value of these familiar paradigms, but believe that electronic sound can and should have many alternative, idiosyncratic voices. Through the design and construction of unique sound diffusion structures, one can reinvent the nature of electronic sound; when allied with new sensor technologies, these structures offer alternative modes of interaction with techniques of sonic computation. This paper describes several recent applications of spherical speakers (multichannel, outward-radiating geodesic speaker arrays) and Sensor-Speaker-Arrays (SenSAs: combinations of various sensor devices with outward-radiating multi-channel speaker arrays). This presentation introduces the development of four generations of spherical speakers—over a hundred individual speakers of various configurations—and their use in many different musical situations including live performance, recording, and sound installation. We describe the design and construction of these systems, and, more generally, the new ‘‘voices’’ they give to electronic sound.
Developments in self-noise towed-array inversion
Authors: Battle David J.
Published: October 2013

In preliminary experiments, self-noise geoacoustic inversion from towed-array data has been demonstrated as a feasible concept, whereby sonar performance predictions in shallow water can be based on locally measured geoacoustic parameters while a tow-ship is underway. In this approach, replica fields generated using a near-field propagation model are compared to measured source fields via a conventional matched-field processor (MFP). With parameter selection controlled by a combination of local and global search algorithms, MFP output power is used as a feedback signal to optimize the match between the modeled and actual acoustic environments. In addition to the desired geoacoustic parameters, the search space includes nuisance parameters such as unknown or poorly known depths, ranges and array perturbations. This paper discusses further results and practicalities of self-noise inversion using towed-arrays, including aspects of environmental sensitivity, propagation model selection, source signature acquisition and imaging.
Rapid geoacoustic inversion with a curved horizontal array
Authors: Fialkowski Laurie T. , Dacol Dalcio K. , Lingevitch Joseph F. , Kim Elisabeth
Published: October 2013

Real-time geoacoustic inversions with a towed array are of interest for rapidly characterizing the sediment properties over changing regions, and require an efficient and accurate forward propagation model. A wavenumber integration solution using the computationally efficient method of undetermined coefficients and the Fast-Field Program is implemented with a simulated annealing optimization method and a coordinate rotation technique. The source function of interest may be broadband ship self-noise or a controlled source. The parameters sought in the inversion include, but are not limited to, geoacoustic sediment properties and receiver array geometry; the water column is assumed known. The forward propagation method is efficient and accurate for very short ranges. Results are presented of simulations with a controlled source, as well as preliminary MAPEX2000 ship self-noise. [Work supported by ONR.]
Geoacoustic inversion for elastic bottoms: Matching model predictions and backscatter data from littoral limestone
Authors: Gragg Robert F. , Soukup Raymond J. , Gauss Roger C.
Published: October 2013

The scattering strength of the ocean bottom as a function of angle and frequency is of fundamental importance in predicting the performance of active sonar systems, particularly in littoral waters. In this work, we apply an elastic-bottom scattering model [the small-slope model of Gragg et al., ‘‘Rough interface scattering,’’ J. Acoust. Soc. Am. 110 (2001)] to the problem of matching 2–3.5 kHz acoustic backscatter from rough limestone sea floors off the Carolina Coast and in the Straits of Sicily. We use a simplex/annealing algorithm to determine model parameters that optimally fit the data. Analysis of an ensemble of such inversion runs addresses the following questions. How well can the elastic theory match data measured at sea? How sensitive is the theoretical prediction to the values of its inputs-frequency and the (seven) parameters that characterize the geoacoustics of the bottom material and the roughness of the surface? Results to date indicate more sensitivity to roughness than to geoacoustic parameters. This supports the importance of estimating in situ bottom roughness and the feasibility of using a physics-based model for that purpose. [Work supported by ONR.]
Beamforming beyond the λ/2 limit with microphone arrays
Authors: Moquin Philippe , Dedieu Stéphane , Goubran Rafik A.
Published: October 2013

One limitation of microphone arrays is that the inter-microphone spacing is restricted to λ/2 of the shortest wavelength (highest frequency) of interest. For an increase in frequency range, the array must either be made smaller (thereby losing low-frequency directivity) or the number of microphones must be increased (thereby increasing cost). The other problem is that the beamwidth decreases with increasing frequency and sidelobes become more problematic. This results in significant off-axis ‘‘coloration’’ of the signals. The extension of the working frequency range for an existing narrow-band (300–3 kHz) telephony microphone array to wide-band telephony (up to 7 kHz), without modifying its geometry and the number of microphones will be shown. Microphones are embedded in a diffraction structure that provides the desired directivity at high frequencies. To provide the desired directionality at lower frequencies, beamforming of the microphones is performed using digital signal processing techniques. The combination of beamforming and embedding the microphones in a diffraction structure that provides the desired directivity at high frequencies addresses the two weaknesses that arise in previous approaches: low-frequency directivity with small arrays and high-frequency difficulties that arise in conventional sensor arrays. [Work supported in part by Carleton University.]
Microphone matching for hybrid-order directional arrays in hearing aid applications
Authors: Warren Daniel M. , Thompson Steve C.
Published: October 2013

The ability of a hearing aid user to distinguish a single speech source amidst general background noise (for example, dinner table or cocktail party conversation) may be improved by a directional array of microphones in the hearing instrument. The theoretical maximum directivity index (DI) of a first-order pairing of microphones is 6 dB, and a second-order array of three microphones is 9.5 dB, assuming all three microphones have identical frequency responses. The close spacing of microphone ports in a hearing aid body means that directivity degrades rapidly with differences in microphone sensitivities. A hybrid of first- and second-order arrays can mitigate this effect, although close microphone matching is still necessary for high directivity. This paper explores the effect of microphone mismatch on the directivity of such arrays, and describes practical criteria for selecting matched microphones out of production batches to maximize a speech intelligibility weighted directivity index. [Work supported by Knowles Electronics, LLC.]
Calculation of uncertainty in calibration of microphones by the pressure reciprocity technique
Authors: Hanes Peter , Wu Lixue , Ohm Won-Suk , Wong George S. K.
Published: October 2013

At the primary level, acoustical measurement standards are realized through calibration of the sensitivity level of Laboratory Standard microphones by the reciprocity technique. The technique is described in International Standard IEC 61094-2, which allows for various implementations of the measurement method. The pressure sensitivity levels of a set of three microphones are determined from the electrical and acoustical transfer impedances of pairs of the microphones. The transfer impedances in turn depend on the design and performance of the measurement apparatus, the dimensions and acoustical properties of the microphones and the cavity that acts as an acoustical coupler between the microphones, and the prevailing environmental conditions. The uncertainty in the pressure sensitivity level depends on the uncertainties in these input quantities and on how the sensitivity level varies with changes in the input quantities. The ISO/IEC Guide Express: 1995 Guide to the Expression of Uncertainties in Measurement provides internationally agreed models and guidance for evaluating the expanded uncertainty of a measurement. The uncertainty model, the nature of the input variables, and the steps involved in the calculation of the expanded uncertainty are described for the realization of a particular implementation of the reciprocity technique.
High-frequency monopole sound source for anechoic chamber qualification
Authors: Saussus Patrick , Cunefare Kenneth A.
Published: October 2013

Anechoic chamber qualification procedures require the use of an omnidirectional monopole sound source. Required characteristics for these monopole sources are explicitly listed in ISO 3745. Building a high-frequency monopole source that meets these characteristics has proved difficult due to the size limitations imposed by small wavelengths at high frequency. A prototype design developed for use in hemianechoic chambers employs telescoping tubes, which act as an inverse horn. This same design can be used in anechoic chambers, with minor adaptations. A series of gradually decreasing brass telescoping tubes is attached to the throat of a well-insulated high-frequency compression driver. Therefore, all of the sound emitted from the driver travels through the horn and exits through an opening of approximately 2.5 mm. Directivity test data show that this design meets all of the requirements set forth by ISO 3745.
Introduction to auditory perception in listeners with hearing losses
Authors: Florentine Mary , Buus So/ren
Published: October 2013

Listeners with hearing losses cannot hear low-level sounds. In addition, they often complain that audible sounds do not have a comfortable loudness, lack clarity, and are difficult to hear in the presence of other sounds. In particular, they have difficulty understanding speech in background noise. The mechanisms underlying these complaints are not completely understood, but hearing losses are known to alter many aspects of auditory processing. This presentation highlights alterations in audibility, loudness, pitch, spectral and temporal processes, and binaural hearing that may result from hearing losses. The changes in these auditory processes can vary widely across individuals with seemingly similar amounts of hearing loss. For example, two listeners with nearly identical thresholds can differ in their ability to process spectral and temporal features of sounds. Such individual differences make rehabilitation of hearing losses complex. [Work supported by NIH/NIDCD.]
An introduction to hearing aids
Authors: Dyrlund Ole
Published: October 2013

This presentation reviews hearing-aid development from analog to advanced digital technology. A basic hearing aid consists of a microphone, an amplification circuit that provides a gain that varies with frequency to accommodate variations in hearing loss with frequency, and a small earphone. In recent years, hearing aid technology has developed rapidly. Digital hearing aids have become commonplace and their share of the marketplace is increasing rapidly. Therefore, the main focus of this talk is signal-processing schemes in advanced digital hearing aids, including microphones with digitally controlled directional characteristics, wide-dynamic-range compression in multiple channels that allow the compression characteristics to vary with frequency, noise reduction, and feedback cancellation. Each of these signal-processing functions help address the needs of individuals with hearing losses.
Beyond the hearing aid: Assistive listening devices
Authors: Holmes Alice E.
Published: October 2013

Persons with hearing loss can obtain great benefit from hearing aids but there are many situations that traditional amplification devices will not provide enough help to ensure optimal communication. Assistive listening and signaling devices are designed to improve the communication of the hearing impaired in instances where traditional hearing aids are not sufficient. These devices are designed to help with problems created by listening in noise or against a competing message, improve distance listening, facilitate group conversation (help with problems created by rapidly changing speakers), and allow independence from friends and family. With the passage of the Americans with Disabilities Act in 1990, assistive listening devices (ALDs) are becoming more accessible to the public with hearing loss. Employers and public facilities must provide auxiliary aids and services when necessary to ensure effective communication for persons who are deaf or hard of hearing. However many professionals and persons with hearing loss are unaware of the various types and availability of ALDs. An overview of ALDs along with a discussion of their advantages and disadvantages will be given.
Teaching hearing science to undergraduate nonscientists
Authors: Weiler Ernest M. , Boyce Suzanne , Steger Joseph
Published: October 2013

For those students interested in potential clinical careers in Speech Pathology, or Audiology, a knowledge of some of the scientific bases is important, but should not create a distaste for science. The authors have addressed themselves to these goals: (1) calculation of period, Hz, summation of two sine waves, phase and dB; (2) anticipating undergraduate Speech Science; (3) simple examples of hearing pathology; and (4) basic psycho-acoustical issues. The classic material of Harry Helson was used to elucidate issues of context in experimental science, and that of S.S. Stevens was used to exemplify psycho-acoustical formulas of common use. Four texts that have been tried on approximately 200 students were evaluated. Surprisingly, the best provided the fewest formulas, short study questions with answers, good examples, and a list of common terms. The next best was aimed at slightly more advanced students, but each chapter contained introductory material, examples, and definitions suitable for naïve undergraduates. The least satisfactory text provided excerpts of technical material with abrupt transitions, no examples, and only part of the definitions needed for the naïve student. Perhaps the most difficult teaching issue is to avoid demanding graduate-level science from those undergraduates with clinical aspirations.
Sonoluminescence and multi-bubble cavitation phenomena for selected research and industrial applications
Authors: Greenwood Larry , Olsen Khris , Good Morris , Bond Leonard , Posakony Gerald , Peters Timothy , Baldwin David , Wester Dennis , Ahmed Salahuddin
Published: October 2013

Single bubble sonoluminescence (SBSL), multi-bubble sonoluminescence (MBSL), multi-bubble sonochemiluminescence (MBSCL) and other high power ultrasound cavitation and noncavitating ultrasound process stream interaction phenomena are known to produce a wide range of both physical and chemical effects that depend upon the system and operating conditions employed. Three interacting regimes are under investigation (a) high power and high frequency (including noncavitating systems), (b) single bubble resonance/sonoluminescence and (c) multi-bubble high power sonochemical processing. In all cases these involve various reactors, including possible schemes for continuous material feeding and processing for selected chemical, nonaqueous fluids and biological research and industrial applications. High power sonochemical and noncavitating ultrasound processing applications and a review of literature pertaining to the potential of high power processing, including fusion are discussed. Work includes the investigation of acoustic fields in reactors, characterization of sonoluminescence spectra, the investigation of system parameters to control maximum bubble temperature and pressure, and acoustic energy partition into light and acoustic emission/shock waves. Effects of various chemical systems on multi-bubble luminescence are being investigated and will be reported. Work to date has emphasized the evaluation of both single and multi-bubble sonoluminescence, spectral measurements, acoustic emission measurements and the observation of a continuous bubble feed phenomenon.
Update and clarifications on experimental studies for nuclear emissions during acoustic cavitation
Authors: Taleyarkhan Rusi P. , West C. D. , Cho J. S. , Lahey R. T. , Block R. C. , Nigmatulin R. I.
Published: October 2013

A seminal discovery related to detection of nuclear emissions during acoustic inertial confinement fusion with deuterated acetone has been reported in Science (3/8/2002 issue). Nuclear emissions we measured included 2.5-MeV neutrons and tritium as would be expected from deuterium–deuterium nuclear fusion. These unmistakable statistically significant signatures were measured under conditions commensurate with degassed rapid condensation-induced implosion conditions only with the test fluid deuterated acetone. In these experiments bubble clusters are nucleated in tensioned degassed liquids with neutrons at the nanoscale level and are then made to grow by a factor of ∼100000 in size to the mm scale prior to implosive collapse. Similarly conducted control experiments with natural acetone did not result in any statistically significant nuclear emissions. Shock code simulations (discussed in a companion talk) corroborated these observations and provided insights into the physics of the overall process. Since the recent announcement of this discovery several world-wide researchers have contacted the authors for further clarifications in a variety of areas. The presentation will discuss these issues and questions, and will provide relevant explanations with supporting evidence.
Simulations of cochlear implant hearing using filtered harmonic complexes: Implications for concurrent sound segregation and pitch perception
Authors: Deeks John M. , Carlyon Robert P.
Published: October 2013

We studied concurrent sound segregation using cues similar to those available to cochlear implant listeners. Sixteen normally hearing subjects listened to mixtures of IHR sentences, processed using a simulation of implant hearing. Target sentences were bandpass filtered into six frequency bands between 1020 and 5000 Hz. The amplitude envelope in each band modulated a filtered, alternating-phase harmonic complex with F0=40 or 70 Hz. Each complex resembled a pulse train with pulse rate=2F0 and was filtered in the same way as the speech band that modulated it. The complexes consisted only of unresolved harmonics, whose pitch is processed in a way similar to that of electric pulse trains in implant hearing. The interferer was 3.5 s of time-reversed speech, processed in the same way as the target. After processing, targets were added to interferers at a 1.2-s delay, with SNR=9 dB. The target and interferer used complexes with either the same or different F0. Using a different F0 for the interferer benefited performance when the target F0=70 Hz, but had no effect when the target F0=40 Hz. Modifying the scheme by processing one sentence on only the odd-numbered channels and the other on the even-numbered channels impaired performance in all conditions.
The effects of simulated cochlear-implant processing on F0 discrimination
Authors: Qin Michael K. , Oxenham Andrew J.
Published: October 2013

Fundamental frequency (F0) information has long been thought to play an important role in perceptually segregating simultaneous and nonsimultaneous sources. The ability to discriminate different F0’s is thought to depend primarily on fine-structure information, in particular the information carried in peripherally resolved, lower-order harmonics. Current cochlear-implant users do not have access to such cues. While implant-processed stimuli can carry some periodicity information in the stimulus envelopes, the F0 discriminability associated with envelope periodicity is rather weak. This study investigated F0 discrimination of complexes with F0’s of 130 and 220 Hz in normal-hearing listeners using noise-excited vocoders to simulate cochlear-implant processing. F0 difference limens of tone complexes were measured as a function of processing condition (1, 4, 8, 24, 40 channels, and unprocessed) and environment (tone complex alone and in reverberation). Performance with vocoder processing was poorer than in unprocessed conditions, despite the relatively good frequency resolution with the highest number of channels. Vocoder processing was particularly detrimental to F0 discrimination in the reverberant conditions. The detrimental effects of vocoder processing may be due to the elimination of temporal fine-structure information or the poorer spectral representation of the lower-order harmonics. [Work supported by NIDCD Grant No. R01 DC05216.]
A neural network model of harmonic detection
Authors: Lewis Clifford F.
Published: October 2013

Harmonic detection theories postulate that a virtual pitch is perceived when a sufficient number of harmonics is present. The harmonics need not be consecutive, but higher harmonics contribute less than lower harmonics [J. Raatgever and F. A. Bilsen, in Auditory Physiology and Perception, edited by Y. Cazals, K. Horner, and L. Demany (Pergamon, Oxford, 1992), pp. 215–222; M. K. McBeath and J. F. Wayand, Abstracts of the Psychonom. Soc. 3, 55 (1998)]. A neural network model is presented that has the potential to simulate this operation. Harmonics are first passed through a bank of rounded exponential filters with lateral inhibition. The results are used as inputs for an autoassociator neural network. The model is trained using harmonic data for symphonic musical instruments, in order to test whether it can self-organize by learning associations between co-occurring harmonics. It is shown that the trained model can complete the pattern for missing-fundamental sounds. The Performance of the model in harmonic detection will be compared with experimental results for humans.
Effects of modulation phase on profile analysis in normal-hearing and hearing-impaired listeners
Authors: Rogers Deanna , Lentz Jennifer
Published: October 2013

The ability to discriminate between sounds with different spectral shapes in the presence of amplitude modulation was measured in normal-hearing and hearing-impaired listeners. The standard stimulus was the sum of equal-amplitude modulated tones, and the signal stimulus was generated by increasing the level of half the tones (up components) and decreasing the level of half the tones (down components). The down components had the same modulation phase, and a phase shift was applied to the up components to encourage segregation from the down tones. The same phase shift was used in both standard and signal stimuli. Profile-analysis thresholds were measured as a function of the phase shift between up and down components. The phase shifts were 0, 30, 45, 60, 90, and 180 deg. As expected, thresholds were lowest when all tones had the same modulation phase and increased somewhat with increasing phase disparity. This small increase in thresholds was similar for both groups. These results suggest that hearing-impaired listeners are able to use modulation phase to group sounds in a manner similar to that of normal listeners. [Work supported by NIH (DC 05835).]
Perceptual learning in frequency discrimination and amplitude-modulation rate discrimination, and generalization to fundamental frequency discrimination
Authors: Grimault Nicolas , Micheyl Christophe , Carlyon Robert P. , Bacon Sid P. , Collet Lionel
Published: October 2013

Fifteen subjects were trained during twelve 2-h sessions in either frequency discrimination with pure tones, or amplitude-modulation rate discrimination of noise bands. Thresholds for the discrimination of pure-tone frequency, harmonic complex tone fundamental frequency, and amplitude-modulation rate were measured before, during, and after training. Comparison of pre- and post-training thresholds revealed significant improvements in all conditions in both subjects trained in frequency discrimination and subjects trained in modulation rate discrimination. Training in frequency discrimination resulted in larger improvements in fundamental frequency discrimination when the test complexes contained resolved harmonics than when they were composed of unresolved harmonics. Training in modulation rate discrimination did not result in larger fundamental frequency discrimination improvements for unresolved than for resolved harmonics. The implications for models of pitch perception are discussed.
The aging middle ear: Wideband energy reflectance measurements
Authors: Feeney M. Patrick , Sanford Chris A.
Published: October 2013

Several anatomical studies have documented aging effects in the human middle ear. However, efforts to study the effect of aging using both low-frequency and multifrequency tympanometry to 2000 Hz have been inconclusive. This study examined energy reflectance at ambient pressure from 250 to 10,080 Hz in 40 young (M=22 years) and 34 elderly adults (M=72 years). All subjects had normal 226 Hz tympanometry and audiometric air-bone gaps of 10 dB or less. Reflectance measurements were obtained in a sound-treated booth using a digitally-generated wideband chirp as the probe stimulus delivered by a receiver in an ER-10C microphone. Each reflectance measurement consisted of a time-waveform average of the microphone response to 8 chirps. Three such one-third-octave reflectance responses were averaged to obtain an estimate of middle ear reflectance for one ear of each subject. The average reflectance for the elderly subjects was significantly lower from 794 to 2000 Hz with a maximum difference of 15% at 1260 Hz. A reflectance minimum occurred at 4000 Hz for both groups, but was about 15% lower for the young group. Results will be compared to published adult data using similar systems. [Work supported by the NIDCD Grant No. DC04129.]
Optimization of a state-switched absorber applied to a vibrating continuous system
Authors: Holdhusen Mark , Cunefare Kenneth
Published: October 2013

A state-switched device is a device that can instantaneously change one or more of its dynamic properties. A vibration absorber that can switch between discrete stiffenesses is termed a state-switched absorber. Using a maximum work extraction switching rule, the state-switched absorber has been shown to increase performance in lumped mass systems as compared to a classical tuned vibration absorber. Because the state-switched absorber can retune its resonance frequency during the response of the system, the bandwidth which the state-switched absorber is effective is larger than that of classical passive devices. The work at hand considers the effectiveness of vibration control using state-switched absorber attached to a continuous system subjected to a multifrequency excitation. The switching rule, tuning frequencies, and location of attachment on the continuous system are optimized for the state-switched absorber to achieve the lowest kinetic energy in the insolated system. Tuning and location optimization is also done for a classical tuned vibration absorber. The optimized performance of the state-switched absorber is compared to the optimized performance of the tuned vibration absorber for a number of different forcing cases.