bluuebunny/crossref_metadata_2025 huggingface dataset:
The time and place of cochlear implant pitch perception
Authors: Bissmeyer Susan R. , Hossain Shaikat , Goldsworthy Raymond L.
Published: April 2018

It is well established that cochlear implant users can perceive pitch associated with pulse train stimulation rate, but that rate sensitivity diminishes above 300 pps. However, there have been multiple reported cases of individuals sensitive to stimulation rates well above 300 pps. Furthermore, Goldsworthy and Shannon (2014) demonstrated that sensitivity to stimulation rate could be improved through psychophysical training. The goal of the present study is to identify factors affecting rate sensitivity in cochlear implant users, including age, etiology, and duration of deafness, electrode location, stimulation mode, neural health as measured by multipulse integration, and spatial tuning as measured by forward masked thresholds. Adult cochlear implant users serve as subjects in a 3-week protocol that includes 4 hours per week of electrode psychophysics conducted in the laboratory and 30 minute daily auditory listening exercises conducted at home. Results indicate that cochlear implant users with minimal psychophysical training consistently discriminate rate differences at least as high as 800 pps. Emerging trends show a group effect with a monopolar stimulation mode and basal cochlear location benefit. We will report on the effect of the factors affecting rate sensitivity for this ongoing study.
Examining the auditory cortex response to musical stimuli with differences in timbre and reverberation
Authors: Lawless Martin S. , Vigeant Michelle C.
Published: April 2018

In a previous functional magnetic resonance imaging (fMRI) study by the authors, the addition of reverberation was shown to degrade the cortical auditory response to musical stimuli. This degradation may occur since reverberation blurs the frequency content of the stimuli in time, making it more difficult for the brain to distinguish between distinct auditory objects. The current study aims to further establish the auditory response to reverberant stimuli by investigating if higher-order frequency content (timbre) affects the auditory cortex’s sensitivity to reverberation. Using an anechoic solo-instrumental trumpet motif, a sine-trumpet musical stimulus was generated by removing the harmonic content from the played notes while maintaining the attack of the instrument and rhythm of the musical passage. Auralizations of both motifs were created from eight simulated room conditions with reverberation times ranging from 0.0-7.2 s. Participants were recruited to rate the auralizations in terms of overall preference, perceived reverberance, and perceived clarity. Subsequently, the participants listened to a subset of the solo-instrumental and sine-trumpet auralizations in an MRI machine. Contrasts between the motifs and room acoustic conditions were analyzed to confirm the sensitivity of the auditory cortex to reverberation, as well as investigate the possible interaction effect between reverberation and timbre.
Observation on the variability of fitting by two experimenters at REAT and the agreement with objective measurement on three premolded earplugs
Authors: Liu Yufei
Published: April 2018

Real ear attenuation at threshold (REAT) has been recognized as the gold standard method to evaluate the attenuation of hearing protection devices (HPDs). It has been standardized in ANSI, ISO/EN and AS/NZS documents. The fitting of HPDs is one of the key factors of REAT measurement. The experimenter fit protocol of ANSI S3.19 is considered one precise approach to obtain the “best fit” of HPDs. A study of the attenuation obtained between the three fits for two experimenters on the same ten subjects on three premolded three-flange earplugs was conducted at 3M E-A-RCAL laboratory, including two models of Chinese brands earplugs and 3MTM E-A-RTM UltraFit earplug. Good agreement was shown between the REAT results that were obtained by the two experimenters. The REAT result of one Chinese brand earplug shows much higher variability within subjects. Comparing with the insertion loss (IL) measured by G.R.A.S 45CB, which has been corrected for bone conduction, occlusion effects and physiological noise, produced IL for two of the tested earplugs that did not showed good agreement with REAT results at most, but not frequencies. The discrepancies will be discussed.
Quantifying the insertion loss of hearing protection using a compressed gas shock tube
Authors: Argo Theodore F. , Greene Nate , Easter James , Tollin Daniel J. , Walilko Timothy J.
Published: April 2018

Compressed gas shock tubes provide a method of generating repeatable and consistent blast overpressure waves similar to those observed in combat and military training. Shock tubes of different diameters and lengths have been used to evaluate the level dependence of various types of hearing protection devices (HPDs) for sound pressure levels ranging from 132 to 192 dB. Insertion loss data was generated using mechanical test fixtures, post mortem human surrogates, and animal models in order to compare the respective auditory responses at these high pressures. The inclusion of animal models was possible since gas driven shock tubes can be housed in mobile laboratories and trailered to vivarium for physiological and behavioral observations. Measurements were conducted on a variety of HPDs and indicate that the responses of many advanced hearing protection devices are nonlinear at these sound pressure levels. Models of hearing protection and auditory injury have been updated and improved based upon quantification of HPD level-dependence. [Work supported by DOD grant W81XWH-15-2-0002.]
Anticipatory middle ear muscle contractions in damage-risk criteria
Authors: Flamme Gregory , Tasko Stephen M. , Deiters Kristy , Murphy William J. , Jones Heath , Ahroon William A. , Greene Nate
Published: April 2018

Anticipatory middle ear muscle contractions (MEMC) have been implemented as protective components of Damage-Risk Criteria (DRC) for impulsive noises. However, no studies have shown that anticipatory MEMC are pervasive among humans. This presentation describes a series of studies of the viability of assumed anticipatory MEMC obtained either through classical conditioning or while operating a model gun. Participants were adults with normal hearing, and the conditioning tasks varied on sensory modality and attention. Both between- and within-subjects designs were used. A conditioned response was defined as an MEMC occurring prior to the unconditioned stimulus and when only the conditioning stimulus was presented. These studies do not suggest that anticipatory MEMC should be included in DRC for impulsive noises.
Developing a method to assess noise reduction of firearm suppressors for small-caliber weapons
Authors: Murphy William J. , Flamme Gregory , Campbell Adam R. , Zechmann Edward L. , Stewart Michael , Tasko Stephen M. , Lankford James E. , Meinke Deanna K. , Finan Donald
Published: April 2018

Firearm suppressors have the potential to reduce the muzzle blast by diffusing the initial shock wave of a gunshot. Currently, the American National Standards Institute does not have any standards that specifically address suppressor measurements. A recent NATO standard, AEP 4875 has been proposed to characterize suppressor performance, but the scope of this standard does not include suppressor effects at the shooter’s ear. Additionally, the standard requires firing the weapon from an elevated platform 4 meters above the ground with microphones positioned with regular spacing of about 18 degrees at 5 meters from the muzzle. This study evaluated fourteen different firearms with and without a suppressor. Different loads of ammunition were used to vary the speed of the projectile. For ten of the guns, both supersonic and subsonic conditions were measured. Twelve microphones were positioned at 30-degree spacing in 3-meter ring at 1.5 meters above the ground. One microphone was positioned at 1 meter to the left of the muzzle and two microphones were positioned at 15 centimeters from the right and left ears. The suppressors were effective in reducing the peak sound pressure levels between 3 and 28 dB and 8-hour equivalent energy (LAeq8) between 2 and 24 dB.
How ordering effects change with cursor placements on scales when evaluating annoyance levels of transient environmental sounds
Authors: Zhang Yiyun , Davies Patricia
Published: April 2018

Sound ordering effects are the impacts of the last heard sound on the rating of the current sound when evaluating a group of signals in sequence. Previously, it was shown that such effects are more significant in evaluations of transient sounds than in evaluations of steady-state sounds. The goal of this study was to investigate whether initial cursor placement on the rating scale could strengthen or attenuate ordering effects. In the previous study, the cursor was placed in the middle for the first signal, and then was left at the place from the last rating. In this follow-up test with the transient sounds, the effect of three different cursor setting strategies was examined: (1) always at the left (indicating the least annoyance level), (2) at the place of the last rating (same as the previous test), and (3) randomly placed on the rating scale. Preliminary results show that the overall trends of sound annoyance ratings with the three cursor setting strategies are similar for most subjects, and results are similar to those in the previous test. However, for some subjects, left cursor placement tends to reduce those subjects’ annoyance ratings.
A wide-area system for tracking supersonic aircraft overflights and capturing their shock waves
Authors: Sanchez Gonzalo , Horonjeff Richard , Mestre Vincent , Fidell Sanford
Published: April 2018

An Internet-based, geographically-distributed prototype system for capturing sonic booms produced by NASA’s low boom flight demonstrator (LBFD) aircraft has been successfully field tested. The system is designed to track supersonic LBFD flights over distances of hundreds of miles so that interviewing of overflown populations can be synchronized with the time of arrival of their sonic booms in multiple, widely separated communities. A central server archives reports of reception of the demonstrator aircraft’s ADS-B signals from multiple field receivers, calculates shock wave arrival times at interviewing sites, captures and stores the acoustic waveform produced at the predicted time of arrival of the boom, coordinates interview start and stop time with sonic boom arrival times, uploads the captured waveforms on demand, and provides multiple, geographically-distributed analysts with password protected, remote access to the flight tracking and acoustic information in near-real time.
Multilevel analysis of recent noise social survey data including noise sensitivity
Authors: Rathsam Jonathan , Hayward Maurice E. , Gille Laure-Anne , Nykaza Edward T. , Wayant Nicole M.
Published: April 2018

Noise social surveys are typified by large variation in annoyance ratings for equivalent noise exposure. Some of this variation has previously been linked to individual attitudes, including noise sensitivity. A framework provided by multilevel statistical analysis can be used to study the relationships between noise annoyance and individual attitudes. The current effort applies multilevel analysis to two recent community surveys of impulsive noise for which noise-sensitivity was assessed via a baseline survey. In the results, the sizable annoyance variation among respondents is quantified. Multilevel models with random effect terms for slope and or intercept describe more variation than multiple regression models with only fixed effects for noise exposure and noise sensitivity.
Improvements in transaural synthesis with the Moore-Penrose pseudoinverse matrix
Authors: Shore Aimee , Hartmann William M.
Published: April 2018

Transaural synthesis using loudspeakers is a powerful technique for conducting binaural hearing experiments while avoiding myriad problems caused by headphones. It is the most promising method to test binaural hearing for listeners using cochlear implants. Transaural synthesis is typically implemented using crosstalk cancellation, a technique whereby the left-channel signal and right-channel signal are adjusted so that reproduction by two synthesis loudspeakers results in near perfect transmission of the target left and right signals to the left and right ears (or processor microphones). Crosstalk cancellation has enabled precise experimental control over the stimulus delivered to each ear. Unfortunately, the 2×2 matrix inversion required for crosstalk cancellation can occasionally introduce spuriously large amplitudes for specific frequencies into the adjusted signals. These large amplitudes lead to perceptually salient tones. We demonstrate through simulation and experiment in a real room that adding a third loudspeaker and solving the resulting 2×3 problem using the Moore-Penrose pseudoinverse matrix yields dramatically fewer large amplitudes in the resulting adjusted waveforms. We have also conducted experiments to investigate the 2×3 system's robustness to inadvertent listener motions.
Threshold interaural time differences under optimal conditions
Authors: Dietz Mathias , Thavam Sinthiya
Published: April 2018

Klumpp and Eady (1956, J Acoust Soc Am 28, p.859-860) reported preliminary data on human sensitivity to interaural time differences (ITD) with various stimuli. At 10 μs ITD the best discrimination of 79% correct was reported for band-pass filtered (150-1700 Hz) noise. Despite the preliminary nature, and presentation methods different from todays, the above is still the best available reference for optimal ITD discrimination. The goal of the current study is to systematically determine the stimulus and the experimental paradigm that results in the smallest threshold ITD and to provide an accurate reference value. We varied seven stimulus and procedure parameters: stimulus waveform, stimulation level, stimulus duration, adaptive versus constant stimulus procedure, alternative-forced-choice (AFC) procedure, inter-stimulus pause duration, and complete waveform versus ongoing ITD. The condition yielding the lowest threshold ITD was Gaussian noise band-pass filtered from 20 to 1400 Hz, presented at 70 dB SPL, with a short inter-stimulus pause of 50 ms, and an interval duration of 0.5 s. Averaged across 8 trained subjects, the threshold ITD for this condition at the 79% correct level was 7 μs. The influence of each parameter will be discussed together with the obstacles of accurately determining this value.
Importance of low frequency hearing for spatial release from masking
Authors: Srinivasan Nirmal Kumar , Schmidt Maxwell , Staudenmeier Alexis , Clark Kelli
Published: April 2018

Listeners tend to achieve better speech understanding when the masking talker(s) are spatially separated from the target talker; an effect known as spatial release from masking. Here, we present data on spatial release from masking experiment where the target and maskers are high pass filtered with a three frequency audiometric pure-tone threshold equal to 20 dB HL. Young normal hearing listeners were presented with CRM sentences in anechoic and reverberant listening environments. The target speech was presented simultaneously with two maskers that were either colocated (0° azimuth) or symmetrically separated from the target in azimuth ( ±15°). Reverberant listening environments were simulated using techniques described in Zahorik (2009) and reverberation times (T60) of 1s and 2s were used. Preliminary results indicate that the amount of release from masking reduced for filtered speech and this reduction was greater for reverberant listening environments as compared to anechoic environment.
Is there spatial release from listening effort in noise and reverberation?
Authors: Rennies Jan , Kidd Gerald
Published: April 2018

Spatial unmasking of speech in the presence of interfering sounds has been investigated mostly by assessing speech recognition performance (e.g, SRTs or percent correct). Typically, these experiments measure recognition under very difficult listening conditions in which optimal performance (i.e., obtained in quiet) is reduced. For normal-hearing listeners this usually means presenting the signal at low/negative SNRs to avoid ceiling effects. However, such conditions may not be representative of everyday listening and hence the generalization of the results to real-world environments is limited. However, several recent studies have concluded that listening effort may be more meaningful than intelligibility because it can vary across a wider range of SNRs including positive SNRs where intelligibility is at ceiling. Despite this advantage, relatively few studies have examined listening effort in realistic listening conditions. In particular, binaural effects in listening effort have not been systematically investigated. The goal of this study is to measure listening effort in normal-hearing listeners in conditions with systematically varying binaural parameters, involving one or more interferers and different degrees of reverberation. Listening effort is assessed using categorical listening effort scaling, and experimental data are compared to predictions of a binaural listening effort model derived from a binaural speech intelligibility model.
Using binaural beat sensitivity to explore mechanisms of bimodal temporal envelope beat sensitivity
Authors: Dirks Coral , Nelson Peggy , Oxenham Andrew J.
Published: April 2018

Current cochlear implant (CI) fitting strategies aim to maximize speech perception through the CI by allocating all spectral information across the electrode array without regard to tonotopic placement of each electrode along the basilar membrane. For patients with considerable residual hearing in the non-implanted ear, this approach may not be optimal for binaural hearing. This study aims to explore fitting procedures in which CI maps better complement information from the acoustic ear by reducing the frequency mismatch between them. We investigate the mechanisms of binaural temporal-envelope beat sensitivity in normal-hearing listeners using bandpass filtered pulse trains with parameters including stimulus level, filter bandwidth, filter slope, and spectral overlap using bandpass filtered pulse trains. We find the minimum baseline interaural timing difference and spectral mismatch that normal-hearing listeners can tolerate while maintaining their ability to detect interaural timing differences. Initial results consistently demonstrate maximum sensitivity to binaural beats when place of stimulation is matched across ears. The outcomes of this study will provide new information on binaural interactions in normal-hearing listeners and guide methodology for incoming single-sided-deafness patients as we adjust their CI maps in an effort to reduce the frequency-mismatch. [Work supported by NIH grant F32DC016815-01.]
Evaluating sound localization cues for premium hearing aids across multiple manufacturers
Authors: Diedesch Anna C. , Stecker G. Christopher , Gallun Frederick J.
Published: April 2018

Modern hearing aids employ complex signal processing to improve signal-to-noise ratios for people with impaired hearing. Some of this processing, however, can reduce interaural level differences (ILD), which could reduce sound localization abilities and thus interfere with communication in complex acoustic scenes. It has been speculated that interaural time differences (ITD) are also distorted by amplification, perhaps due to even very brief signal processing delays or multi-path acoustics. Previously we found little ITD distortion for a single set of hearing aids, except when “strong” directional microphones were enabled. To extend and replicate these results, premium hearing aids from four major manufacturers were compared. Aids were set to low-gain amplification and a range of different signal processing algorithms were systematically activated in a manner consistent with clinical practice. Binaural acoustical recordings were measured using an acoustic manikin on which the hearing aids were attached as would be the case for a typical patient. This presentation will describe recordings collected in anechoic and simulated rooms for which frequency-specific ITD and ILD have been extracted and compared to unaided recordings. [Work supported by the F.V. Hunt Postdoctoral Research Fellowship, NIH R01-DC011548 (GCS), R01-DC011828 (FJG), and the VA RR&D NCRAR.]
Integrating remote microphone signals with hearing-aid processing
Authors: Kates James , Arehart Kathryn
Published: April 2018

Remote microphones (RM) are used with hearing aids to provide speech signals free from noise and room reverberation. However, because the RM signal bypasses the acoustics of the head and pinna, it can provide an unnatural signal that is lateralized at the center of the head rather than localized at the talker. A structural binaural model is combined with synthesized early room reflections to provide the missing externalization cues. The effectiveness of this simulation has been demonstrated over headphones using linear amplification to compensate for the hearing loss. In a hearing aid, however, the RM processed signal is combined with the acoustic inputs at the hearing-aid microphones, and the processed hearing-aid output is combined with the reverberated speech that is transmitted through the vent or open fitting. The hearing-aid processing implemented in the experiments described in this presentation includes both linear amplification and wide dynamic-range compression. The externalization, spatial soundfield characteristics, and sound quality provided by the entire signal-processing system are evaluated for normal-hearing and hearing-impaired listeners.
Binaural intelligibility level difference with a mixed-rate strategy simulation
Authors: Leclère Thibaud , Kan Alan , Litovsky Ruth
Published: April 2018

In noisy situations, bilateral cochlear implant (BiCI) listeners demonstrate poorer spatial release from masking compared to normal hearing (NH) listeners. One reason for this difference could be the limited access for BiCI listeners to fine structure interaural time differences (ITDs) which are crucial for binaural unmasking. Clinical processors convey speech envelopes using pulsatile high-rate (~1000 Hz) carriers, and discard the original temporal fine structure. At high rates, BiCI listeners are not sensitive to ITDs, whereas low rates (< 300 Hz) result in better ITD sensitivity. To preserve both ITD sensitivity and well-sampled speech envelopes, high and low rates are needed. The present study examines whether a mixed rate strategy will enable binaural unmasking in NH listeners using a 16-channel mixed-rate vocoder. Binaural intelligibility level differences were calculated from speech reception thresholds measurements in five mixed-rate configurations. Targets had a 0-μs ITD and masking speech-shaped noises were co-located or separated by applying a 400-μs ITD to the low-rate channels. Results from this experiment will help determine the potential benefits of a mixed-rate strategy to restore spatial unmasking abilities of CI listeners in noisy situations.
The role of onset cues for lateralization of an auditory object in bilateral cochlear implant listeners
Authors: Thakkar Tanvi D. , Kan Alan , Litovsky Ruth
Published: April 2018

In bilateral cochlear implant (BiCI) listeners, a common strategy for improving sensitivity to interaural timing differences (ITDs) is to introduce low rate stimulation in some channels. Previous work showed that BiCI listeners can lateralize a single sound with a constant ITD at every pulse of a low rate stimulus. However, in reverberant environments, ITDs for a target sound can change over a short period of time. Previously we have shown that BiCI listeners had difficulty using the ITD in the first pulse of a low-rate pulse train to lateralize a stimulus when the subsequent pulses had ITDs that varied over time. Hence, we hypothesized that a more salient onset is needed in order to restore ITD sensitivity in BiCI listeners when using low rate stimulation. In this study, we measured the number of pulses with a consistent ITD that is needed at the onset in order for BiCI listeners to reliably perceive one sound in the correct location. Results suggest that BiCI listeners need at least five pulses with consistent ITDs in order to report a single auditory object that is correctly lateralized. These results have important implications for low rate strategies aimed at restoring ITD sensitivity to BiCI listeners.
The effect of movement, compression, and cochlear-implant processing on the output of cochlear implants
Authors: Archer-Boyd Alan , Carlyon Robert P.
Published: April 2018

Dynamic-range compression is used in cochlear implants (CIs) to compensate for the greatly reduced dynamic range of CI listeners. It has been found that unlinked bilateral compression in hearing aids (HAs) can affect the spatial location of sounds, reducing interaural level differences (ILDs) and causing sound sources to appear closer to 0° azimuth and/or more diffuse (Wiggins and Seeber, 2011). The compressors in CI processors are often much stronger than hearing aids (e.g., CI compression ratios of 12:1 vs. ≤3:1 in HAs). We investigated the effect of compression on monaural and binaural level cues during simulated rotational movements in the horizontal plane. Moving signals were created by combining static behind-the-ear hearing aid microphone impulse responses at successive angles. For sound levels above the compressor threshold, fast movements over wide angles (120°/s) altered the size and extent of monaural level changes and ILDs considerably, relative to those from those that would occur naturally and without compression. The relatively slow attack and release times of the compressor also had the effect of producing a “lag” in the level changes after the movement had stopped. Microphone directionality was also found to have an effect on the output of the compressor during movement.
Mechanisms underlying speech masking release in hybrid cochlear implant users
Authors: Tejani Viral , Brown Carolyn
Published: April 2018

Hybrid cochlear implant (CI) users utilize low-frequency acoustic and high-frequency electric stimulation in the implanted ear, outperform traditional CI users in speech perception in noise (Gantz et al., 2017), and show “masking release,” an improvement in speech recognition in fluctuating noise relative to steady noise (Turner et al., 2004). Improved performance in noise has been attributed (but not proven) to the spectral resolution and temporal fine structure (TFS) provided by acoustic hearing. Spondee recognition in two-talker and ten-talker babble was assessed in Hybrid CI users, and masking release was calculated as the percent difference in performance in two-talker (easy condition) and ten-talker (hard condition) babble. Spectral ripple density discrimination thresholds and fundamental frequency discrimination limens was also assessed, as these psychophysical measures reflect underlying spectral resolution and TFS and correlate with speech perception in noise in traditional CI users (Won et al., 2007; Goldsworthy et al., 2013; Goldsworthy, 2015; Jeon et al., 2015). Preliminary data suggest masking release is possible in Hybrid CI users, and that psychophysical measures and masking release may be correlated, suggesting a role of spectral resolution and TFS in masking release. The extent of residual hearing also appears to correlate with psychophysical and masking release performance.
Comparison of different implementations of the ACE cochlear implant signal processing strategy with an objective metric
Authors: Murta Bernardo , Chiea Rafael , Mourão Gustavo , Paul Stephan , Cordioli Julio A.
Published: April 2018

Research that seeks to compare outcomes of Cochlear Implant (CI) processing strategies must ensure that different implementations of a given strategy are safe, and means for their comparison are required. This work aims to analyze the sensitivity of an objective metric to the effects introduced by different sets of filter bank models, compressing functions, envelope detectors and other aspects in implementations of the same CI signal processing strategy. To this end a subset of signals containing pure tones, complex harmonics, exponential sweeps and speech signals is defined and processed by means of the same CI strategy (ACE) in two different implementations: a standard version from the Cochlear Nucleus Toolbox and another version implemented by the authors using published data (FSC). As expected, variations in the output are due to different aspects in the signal processing chain of each implementation. While the proposed metric is sensitive to this differences, its magnitude also depends heavily on the test signals used and specificity is very low. Therefore a more accurate, specific and sensitive protocol was developed using UTD's CCi-Mobile Research Interface to analyze the root of the differences. This includes a metric and performance criteria that are specific for certain signal processing parameters.
Infants' use of amplitude modulation cues in the segregation of concurrent vowels
Authors: Oster Monika-Maria , Werner Lynne A.
Published: April 2018

Infants have difficulties separating speech from competing sounds. One explanation is that infants do not use acoustic cues to sound source segregation as adults do. This study investigated 3- and 7-month-old infants’ and adults’ ability to use AM cues in concurrent vowel segregation. Seven American-English vowels were produced by a male and by a female talker. One male and one female vowel were randomly chosen and superimposed. A train of such vowel pairs was presented to listeners, who were trained to respond to the male target vowel /i:/ or /u:/. Target-to-nontarget ratios were chosen to equate performance across age groups. In the baseline condition, all vowels had flat envelopes. In the cue condition, male vowels were modulated with three envelopes from infant-directed vowels. The proportion of 3-month-old infants achieving an 80% correct criterion at a ratio of + 15dB was higher in the cue than in the baseline condition. For 7-month olds and adults d' could be estimated based on 15 target and 15 nontarget trials in each condition. Preliminary results indicate that both 7-month-old infants and adults have a higher d' in the cue than in the baseline condition. Thus, infants appear to use AM to segregate concurrent vowels.
Bianisotropic acoustic metasurfaces for wavefront manipulation
Authors: Goldsberry Benjamin M. , Lawrence Andrew J. , Haberman Michael R.
Published: April 2018

Materials with sub-wavelength asymmetry and long-range order have recently been shown to respond to acoustic waves in a manner that is analogous to electromagnetic biansiotropy [Sieck et al. Phys. Rev. B 96, 104303, (2017)]. A characteristic of these materials is the generation of dipolar scattering when subjected to a uniform time-varying pressure field. This behavior leads to a characteristic acoustic plane-wave impedance that has a preferential direction, known as a polarization, which results in direction-dependent magnitude and phase of an acoustic field scattered from bianisotropic acoustic media. These materials can therefore be used as acoustic metasurfaces to control the reflected or transmitted acoustic fields. This work presents the use of bianisotropic acoustic media for wavefront manipulation through analytical and finite element modeling and discusses potential acoustic analogues to existing bianisotropic electromagnetic metasurfaces. [This work was supported by ONR and the National Science Foundation.]
Wave control with “Time Materials”
Authors: Fink Mathias
Published: April 2018

Phononic crystals and Metamaterials are made from assemblies of multiple elements usually arranged in repeating patterns at scales of the order or smaller than the wavelengths of the phenomena they influence. Because time and space play a similar role in wave propagation, wave propagation is affected by spatial modulation or by time modulation of the refractive index. Here we emphasize the role of time modulation. We show that sudden changes of the medium properties generate instant wave sources that emerge instantaneously from the entire wavefield and can be used to control wavefield and to revisit the way to create time-reversed waves. Experimental demonstrations of this approach will be presented More sophisticated time manipulations can also be studied in order to extend the concept of phononic crystals in the time domain.
Reciprocity theorems and perception of non-reciprocal behavior in vibrating systems, imetamaterials, acoustic devices, and electroacoustic devices
Authors: Pierce Allan D.
Published: April 2018

Equations governing physical systems often admit a reciprocity corollary. Such systems include mechanical, acoustical, vibrational, and electroacoustic systems. There is no universally applicable succinct definition of reciprocity, although it is sometimes loosely stated that reciprocity means that, “if you see it, then it can see you.” There have been a considerable number of derivations of reciprocity theorems for different types of acoustical and mechanical systems published over the years --- notable names are Helmholtz, Rayleigh, Lamb, Lyamshev, Schottky, Foldy, Primakoff, Payton, de Hoop, Knopoff, Gangi, Arntsen, Carcione, and Fokkema. There appears to be no universally applicable derivation, and no universally applicable reciprocity principle. The emergence of metamaterial systems poses new challenges and opportunities in regard to reciprocity. Given one such system, one naturally asks if there is some reciprocity theorem that applies. If one perceives that there is no such theorem, then one might naturally ask whether the perception is wrong, and if it is simply the case that no one has yet bothered to try to derive one or that one is not sufficiently clever to find one. It may also be that there is an analogous system for which there is a theorem, and that the theorem applies for the present system in some approximate sense. If so, how well? Present paper examines several metamaterial systems introduced in the recent literature from the viewpoint point of reciprocity theorems.
Extreme anisotropy over an acoustic hyperbolic metasurface
Authors: Quan Li , Alu Andrea
Published: April 2018

The emergence of optical hyperbolic metamaterials has attracted significant attention in the last decade due to its extreme anisotropy, linked to largely enhanced local density of states. This concept has been also extended to acoustics, and acoustic hyperbolic metamaterials have been realized. However, the fabrication complexity of these metamaterials, their inherent sensitivity to loss and disorder, and the fact that the hyperbolic modes are not easily accessible, as they are trapped in the bulk of the material prevent the wide use of hyperbolic metamaterials for sound. Our research group has recently proposed the idea of hyperbolic metasurfaces, for which extreme anisotropy over a surface enables plasmons that propagate with similar unbounded local density of states, but restricted to surface propagation. In this work, we extend these findings to acoustics, simplifying the realization of hyperbolic propagation and observation, and at the same time preserving the exciting properties of hyperbolic metamaterials. By carefully designing the acoustic microstructure decorating a metasurface, we successfully confine acoustic waves traveling along the interface with air and observe hyperbolic dispersion and in principle unbounded local density of states. These properties may open exciting opportunities for acoustic and surface acoustic wave devices.
Distorting an impulse wave with phononic metamaterials– A numerical study
Authors: Dorvee Jason R. , Swearingen Michelle E. , Albert Donald G. , Muhlestein Michael , Krieger Megan A. , O'Daniel James L.
Published: April 2018

This numerical study is focused on the effects of disordered arrays as phononic metamaterials on the distortion of low-frequency impulses. Through numerical simulations we have been able to determine the characteristic properties of such phononic metamaterials as they correlate to the disruption a propagating impulsive wave. The fast 2-D FDTD model used allows for rapid reconfiguration of theoretical designs of: arrangements, geometries, and materials for the meta-atom; the results are informing evolving theory of the disruption of impulsive waves using metamaterials. With the ability to share CAD designs with the computational model, the layouts of arrangements from the numerical simulations can be ported to a 3D printer for the fabrication of physical models, for validation of the numerical model. These simulations serve as a planning tool for the configuration of sensors for both bench-scale and field testing. The model used and results of the numerical study are presented. [The work described in this document was funded under the US Army Basic Research Program under PE 61102, Project T22, Task 01 “Military Engineering Basic Research"; Task 02 “Material Modeling for Force Protection;” Project T25, Task 01 “Environmental Science Basic Research” and was managed and executed at the US Army ERDC.]
The influence of talker accent strength on accent categorization
Authors: Federspill Taylor , Bent Tessa
Published: April 2018

Listeners' identification accuracy for nonnative-accented words tends to decrease as the talker's accent strength increases. However, the influence of accent strength on accent categorization has not been investigated. We hypothesized that nonnative talkers with stronger accents would have more salient cues to their native languages and thus, listeners would show increased accent categorization accuracy. To test this hypothesis, monolingual American English listeners categorized sentences produced by 12 native and 24 nonnative talkers in a 12-alternative forced-choice task with 8 nonnative accent categories and 4 regional dialect categories. Three talkers from each language/dialect background with different accent strengths (i.e., mild, moderate, strong) were included. The talkers' accent/dialect strengths were determined by 20 listeners' ratings on a 9-point scale (nonnative talkers: 1 = no foreign accent and 9 = very strong foreign accent; native talkers: 1 = Standard American English and 9 = very strong regional dialect). For the nonnative talkers, preliminary results show a moderate correlation between accent categorization accuracy and accent strength. However, within specific language backgrounds, adherence to the expected accuracy pattern varied substantially. Future analyses will investigate how accent strength influences confusion patterns across accents and dialects. [Work supported by an Indiana University Hutton Honors College grant.]
Sentence recognition in noise: The interplay between talker intelligibility, familiarity, and linguistic complexity
Authors: Strori Dorina , Bradlow Ann , Souza Pamela
Published: April 2018

Foreign-accented speech recognition is typically tested with linguistically simple sentences, which offer a limited window into speech processing. In this study, participants transcribed simple (i.e. mono-clausal, canonical declarative syntax) and complex (i.e. multi-clausal, non-canonical syntax, and/or passive voice) sentences in noise at two signal-to-noise ratios. The sentences were spoken by three talkers: one native, one high and one low intelligibility L2 English speaker. Unsurprisingly, recognition accuracy dropped with decreasing intelligibility and signal-to-noise ratio. Interestingly, accuracy dropped with increasing linguistic complexity only for the native and high intelligibility talkers. For the low intelligibility talker, sentence complexity did not affect recognition accuracy. This pattern of findings was present in mixed and blocked experimental designs, indicating no effect of familiarity with the talker. We argue that listeners employ qualitatively different speech processing strategies for low versus high intelligibility talkers, and suggest that this difference may be driven by the talker’s realization of prosodic structure. While native and high intelligibility talkers effectively encode the contrast between complex and simple sentences, low intelligibility talkers may not convey sufficient prosodic information, resulting in the recognition of both sentence types as non-hierarchically organized word sequences. [Work supported by Knowles Hearing Center, NIH grants R01-DC012289 and R01-DC005794.]
Cognitive effort in comprehending non-native speech decreases with listening experience
Authors: Ogden Dave C.
Published: April 2018

Listeners adapt to non-native accents, as shown by improved accuracy transcribing a speaker's intended utterance during testing (Baese-Berk et al., 2013; Bradlow & Bent, 2008; Sidaras et al., 2009). However, this improvement does not address whether cognitive effort decreases with accent experience, or whether accented speech continues to require effort to compensate for atypical features. The present study addresses this question using pupil dilation, a physiological measure indexing cognitive effort (Sirois & Brisson, 2014). In a matched-guise experiment, 48 participants heard 90 sentences spoken in either a General American Accent (GAA) or a non-native accent (NNA) while an eye tracker measured pupil diameter. An equal number of sentences with words predictable from context (e.g., “Sugar tastes very sweet.”) and without (e.g., “It can’t be brown.”) were included. Participants transcribed (typed) each sentence. Pupil dilation while participants listened to each sentence was measured as the difference between average diameter for 100 ms immediately before and after the sentence played. Dilation decreased significantly more in the NNA than in the GAA condition, indicating that perception of accented speech becomes less effortful with experience. No effects of semantic predictability were observed. The relationship between effort and transcription accuracy will be discussed.
The effect of distributional training on the production of non-native vowel contrasts
Authors: Go Grechen , Campbell Heather , Levi Susannah V.
Published: April 2018

The current study tests production accuracy of a non-native vowel contrast following a modified perceptual learning paradigm. Previous research has found that adults can learn non-native sound categories with sensitivity to distributional properties of their input. In the current study, 34 native-English adults heard stimuli from /o/-/oe/ continuum. Half of the participants heard stimuli drawn from a bimodal distribution and half from a unimodal distribution. To support learning, we incorporated active learning with feedback, lexical support in the form of two images, and overnight consolidation. Production of this contrast was assessed through a repetition task at baseline and at the end of the experiment. Production accuracy is measured as the Euclidean distance between /o/ and /oe/ at baseline and post-training. Preliminary analyses suggest that the distance between these two vowels increased for both groups, indicating that listeners in both conditions were able to transfer perceptual learning to production. These results suggest a way to mitigate the disadvantages previously found for participants in the unimodal condition, by incorporating active engagement with the target stimuli using lexical support, accuracy feedback, and overnight consolidation.
Cross-term analysis in frequency-difference-based source localization methods
Authors: Worthmann Brian M. , Dowling David R.
Published: April 2018

In previous work, it has been shown that a quadratic product of frequency-domain acoustic fields at different frequencies but the same spatial location leads to an auxiliary field which may contain field information at frequencies below the original signal’s bandwidth (Worthmann, Song and Dowling, 2015, JASA 138, 3549-3562). This quadratic product, termed the frequency-difference autoproduct, has been shown to be valuable for beamforming and source localization in the presence of environmental mismatch and/or array sparseness. However, in a multipath environment, this quadratic product leads to undesired cross-terms. Bandwidth averaging procedures have been found to suppress some of their detrimental influences in some cases, but not all. Additionally, the poor dynamic range observed in frequency-difference beamforming and frequency-difference matched field processing are associated with the imperfect mitigation of these cross-terms. In this presentation, the nature of these cross-terms is analyzed, and signal processing tools are developed which attempt to robustly mitigate the detrimental effects of these cross terms. These signal processing tools can be used to potentially improve localization performance when using frequency-difference autoproduct-based source localization schemes. [Sponsored by ONR and NSF]
Investigation of reconfigurable antennas by foldable, e-textile tessellations: Modeling and experimentation
Authors: Zou Chengzhe , Chaudhari Shreyas , Alharbi Saad , Shah Hamil , Kiourti Asimina , Harne Ryan L.
Published: April 2018

Physical deformation mechanisms are emerging as compelling and simple ways to adapt wave propagation properties of antenna arrays in contrast to digital steering approaches acting on topologically fixed antennas. Concepts of physical reconfigurability also enable exceptional capabilities such as deployable and morphing antenna arrays that serve multiple functions and permit compact transport with ease. Yet, the emergent concepts lack broad understanding of effective approaches to integrate conformal, electrically conductive architectures with high-compliance foldable frameworks. To explore this essential interface where electrical demands and mechanical requirements may conflict, this research studies e-textile-based reconfigurable antenna arrays that conform to adaptable topologies by origami-inspired tessellations. The e-textiles leverage embroidered conductive threads along frameworks established on origami tessellations to permit large compliance at folding edges as needed while retaining the desired electromagnetic wave propagation characteristics. Computational modeling is used to guide experimental fabrications and validations. It is found that e-textile origami antenna arrays permit significant adaptation of wave radiation while maintaining the required mechanical robustness under folding sequences. These findings may motivate future concepts for reconfigurable antennas established upon physical deformation processes.
Integration of signal processing modules of hearing aids as real-time smartphone apps
Authors: Kehtarnavaz Nasser , Thibodeau Linda
Published: April 2018

Our research group has been working on developing an open source research platform for hearing improvement studies based on smartphones noting that currently no open source, programmable, and mobile research platform exists in the public domain for carrying out hearing improvement studies. Such a platform allows smartphones, which are already widely possessed and carried by people, to be used for the development and field testing of existing or new hearing improvement signal processing algorithms. A number of smartphones apps have already been developed by our research group for specific or individual modules of the speech processing pipeline of a typical digital hearing aid. This work covers the integration of several individual signal processing modules into an integrated smartphone app. The steps taken in order to enable the individual modules to run together in real-time and with low audio latency will be presented and a demo of the integrated app running on both iOS and Android smartphones will be shown.