bluuebunny/crossref_metadata_2025 huggingface dataset:
Student engagement and education in underwater sound through the build-a-hydrophone activity
Authors: Joy Kevin
Published: May 2014

The hydrophone is arguably the most basic sensor used in underwater acoustics. In its simplest form, this passive device provides scientists, and users of all kinds, the ability to listen to and capture natural and anthropogenic sounds that occur in our aquatic world. As part of an array, this technology enables humans to perform such underwater tasks as tracking submerged targets to studying volcanic and tectonic processes. The concept behind the do-it-yourself (DIY) activity to build a hydrophone is not unique. A simple Internet search will reveal numerous design options geared toward constructing a functional hydrophone. The Center for Ocean Sciences Education Excellence (COSEE), Technology and Engineering for Knowledge (TEK), at the University of Connecticut, has modified this activity to offer an alternative that is both affordable and functional, while minimizing the technical skills, tools, and time required for completion. The outcome has been the development of a hands-on student exercise that provides an avenue for engagement and learning in the Science, Technology, Engineering and Mathematics (STEM) fields, while offering a simple means to introduce the science of underwater sound.
Animations for communicating the science of underwater sound
Authors: Russell Daniel A.
Published: May 2014

For several years, the author has been developing a popular website (http://www.acs.psu.edu/drussell/demos.html) consisting of a variety of computer animations illustrating acoustic phenomena involving wave propagation and vibration [Russell, J. Acoust. Soc. Am., 114, 2308 (2003)]. This paper will showcase a subset of those animations, including several new animations of acoustic wave phenomena and interactive plots created using the Wolfram Mathematica Computable Document Format, which are specifically geared toward enhancing student understanding of the propagation and radiation of sound underwater. Animation examples will include wave propagation; sound radiation from simple sources; directivity patterns for dipoles, doublets, line arrays, and the baffled piston; refraction and reflection; absorption of sound; and waveguides.
Able sea chicks... Adventures in acoustical oceanography
Authors: Van Uffelen Lora J. , Wage Kathleen E.
Published: May 2014

The Able Sea Chicks blog introduces readers to the exciting field of acoustical oceanography by following two female scientists on research cruises to deploy acoustic moorings in the Philippine Sea. The blog was designed for a special outreach session at the April 2010 Acoustical Society of America meeting titled “Listen up and get involved,” aimed at girls involved in the Girl Scouts of America. This session included a video explaining some of the principles of underwater sound propagation and showcasing some of the instrumentation used to collect underwater acoustic data. A live, interactive question and answer session with the scientists onboard the research vessel followed the video presentation. Both the blog and the outreach session were designed to reach girls at the age where research has shown that interest in STEM fields begins to decline. The blog illustrates how ocean acoustics experiments are conducted using examples from the PhilSea10 deployment and recovery cruises. Blog posts cover topics such as hydrophone testing, signal processing, mooring deployment, and long-baseline navigation. Since 2010, the Able Sea Chicks video and blog have been used in other Girl Scout sessions at ASA and in talks for high school and university audiences.
Discovery of Sound in the Sea: Resources for regulators, policymakers, and other stakeholders
Authors: Vigness-Raposa Kathleen J. , Scowcroft Gail , Knowlton Christopher , Morin Holly
Published: May 2014

There is concern about the effects of underwater sound on marine life. These effects must be considered by users of underwater sound under several regulations. To fulfill regulatory requirements, up-to-date resources are needed on the potential effects of underwater sound, as well as fundamental science content. The Discovery of Sound in the Sea website (DOSITS; http://www.dosits.org) provides accurate scientific information on underwater sound at levels appropriate for all audiences, including the general public, K-12 teachers and students, college students, policy-makers, and professionals in industry, education, and the media. Content, such as the effects of sound on marine life, is based on peer-reviewed publications and has undergone rigorous review by the DOSITS scientific advisory panel. This talk highlights new resources for regulators, including structured tutorials and a “hot topics” feature. Structured tutorials provide a directed progression of sequential knowledge; the first one outlines the scientific process for determining the risk of exposure to underwater sound, answering the question “How do you determine if a sound source might affect a marine animal?” The “hot topics” feature allows for current, topical issues to be highlighted on the DOSITS front page, providing efficient access to new information, as well as foundational scientific content.
Study on the aerodynamic noise of centrifugal compressors with rotor static eccentricity
Authors: Wei Kaijun , Zuo Shuguang , He Huijuan , Wang Zhe
Published: May 2014

Centrifugal compressors are widely used in industrial applications. Rotor static eccentricity is among the common faults in centrifugal compressors, and it has a significant influence on the aerodynamic noise. This paper compares the amplitudes and frequency spectrums of the aerodynamic noise of a centrifugal compressor with no eccentricity and the same compressors with different static eccentricities. The aerodynamic noise is calculated by a hybrid method. Large eddy simulation is applied to solve the unsteady three-dimensional internal flow of the compressor, and the pressure fluctuations obtained by the simulation are considered as the noise sources. The far-field aerodynamic noise generated by the noise sources is predicted using the aeroacoustic finite element method. The results show that centrifugal compressors with eccentricity have higher noise level and different spectrum characteristics.
Different experimental methods for the measurement of broadband noise sources in ducts
Authors: Xu Kunbo , Qiao Weiyang
Published: May 2014

Aeroengine broadband fan noise is a major contributor to the community noise exposure from aircraft. It is currently believed that the dominant broadband noise mechanisms are due to interaction of turbulent wake from the rotor with the stator, and interaction of the turbulent boundary layers on the rotor blades with the trailing edge. Two different methods are presented that enable the separation of different broadband noise sources in turbomachinery ducts, one with respect to modal decomposition developed by DLR, and the other with focused beamformer technique developed in university of Southampton. Different measurement mechanisms are displayed to explain the merits, faults, and requirements.
Experimental validation of a new intensity estimation method
Authors: Christensen Benjamin , Thomas Derek C. , Gee Kent L. , Neilsen Tracianne B. , Stewart Menley
Published: May 2014

A new method of estimating acoustic intensity has recently been developed in efforts to improve acoustic measurements of launch vehicles [Thomas et al., “Methods for estimating acoustic intensity in rocket noise fields,” JASA 134(5) 4058–4058 (2013)]. This new method, known as the phase and amplitude gradient estimation (PAGE) method, improves upon the traditional finite difference p-p method for estimating acoustic intensity. The primary advantage of the PAGE method is that it allows for accurate intensity measurements over a larger frequency band. Under certain conditions, it is possible to unwrap the phase component of the PAGE method, allowing for accurate intensity estimates well above previous limitations. The advantages and limitations of the PAGE method are investigated experimentally by taking measurements of arrays of loudspeakers. Preliminary uncertainty analyses of both the PAGE and the finite difference p-p methods are also presented.
An experimental research on the wake flow structure of serrated cascade
Authors: Xu Kunbo , Weiyang Qiao
Published: May 2014

This study concerns the mechanisms of the turbulent broadband noise reduction for cascade trailing edge serrations while the inlet velocity changed from 30 m/s to 50 m/s. The turbulence spatio-temporal information are measured with 3D hot-wire. The experiment is carried out in the Northwestern Polytechnical University low speed open jet wind tunnel. It showed the spreading rate of the wake and the decay rate of the wake centerline velocity deficit increased with serrated edge compared to the straight edge, and the three components of velocity changed differently with serrated trailing edge. It is also found that the turbulence peak occurs further from the airfoil surface in the presence of the serrations, and the serrations widened the mix area, which allowed the flow mixed together ahead of the schedule.
An equivalent source model for the sound intensity in the vicinity of a high-performance military aircraft
Authors: Stout Trevor A. , Kent Gee L. , Neilsen Tracianne B. , Christensen Benjamin Y. , Thomas Derek C. , James Michael M.
Published: May 2014

The sound field of the F-22A Raptor has been measured extensively using arrays of microphones including an intensity probe to the sideline and rear of the aircraft. Recently, an equivalent source model (ESM) has been proposed that incorporates two arrays of monopole sources placed along the jet axis and their image sources to account for the hard ground. The monopole amplitudes in each case are described by a Rayleigh distribution, and one array includes a phase angle between the monopoles to produce correlated noise steered in a specific direction. The model closely replicates the measured SPL at the majority of measurement locations. However, agreement with energy-based quantities such as the vector acoustic intensity may not follow from agreement with the pressure measurements alone. Hence, in this work, the ESM’s ability to produce intensity estimates that match the measured F-22A intensity is evaluated. A new method of estimating acoustic intensity, known as the phase and amplitude gradient estimation (PAGE) method, is applied to the measured F-22A data. The PAGE method accurately estimates intensity in a larger bandwidth than the traditional finite difference method, expanding this discussion of the ESM's accuracy. In addition, sensitivity of ESM parameter selection, e.g., correlated source phase angle and peak amplitude location, on changes in the predicted intensity is examined. [Work sponsored by ONR.]
Development of a digital noise exposure system for research on noise induced hearing loss
Authors: Qin Jun
Published: May 2014

Over 30 million Americans suffer from noise induced hearing loss (NIHL). Many research projects demonstrated that different types of noise, even with equal sound energies, could produce different amounts of hearing loss. In this project, a novel digital noise exposure system has been developed for generating various noise signals (e.g., pure-tone, Gaussian, impulsive, and complex noise). The developed system can be used to study NIHL produced by different types of noise in an animal model. The system could produce impulse noise with peak sound pressure level (SPL) up to 160 dB, which effectively mimics the noise generated by a military weapon (e.g., M-16 rifle). In addition, continuous Gaussian noise with peak SPL up to 140 dB can be created, which is well above the 85 dB recommended exposure limit established by the National Institute for Occupational Safety and Health (NIOSH). The preliminary results of in an animal study showed significant permanent threshold shift (PTS) produced by 90 shocks impulse noise with peak SPL = 155 dB generated by the system. In summary, the digital noise exposure system replicates environmental noise allowing researchers to study hearing loss in a controlled situation.
An iterative approach to measurement of oblique acoustic absorption coefficient in three-dimensions
Authors: Hall Hubert S. , Vignola Joseph F. , Judge John , Turo Diego
Published: May 2014

The measurement of oblique acoustic absorption coefficient remains an ill-defined test. Unlike Kundt's tube (ASTM C384 and E1050) and Sabine room measurements (ASTM C423), there is no uniform test method or standard to follow. In-situ or specific application based testing is the industry norm for arbitrary angular absorption coefficient measurement. An iterative approach has been developed for a free/free/free/free panel in an anechoic chamber. Using numerical model predictions of the acoustic field, microphone measurement data are updated to account for a multitude of acoustic phenomena including diffraction from the panel. This method provides a quick measurement option that is flexible and transportable for many applications.
Infrasound wind noise reduction via porous fabric domes
Authors: Noble John M. , Alberts W. C. Kirkpatrick , Raspet Richard , Collier Sandra L. , Coleman Mark A.
Published: May 2014

Often, porous hose arrays or complex pipe rosettes are used for wind noise reduction in long-term infrasound monitoring situations. Both of these wind noise reduction schemes perform well in reducing the wind-generated noise at an infrasound sensor, but each can significantly filter the information reaching the sensor. Ideally, a wind noise reduction scheme should have a flat frequency response and a minimal attenuation over the desired frequency band. To that end, three fabrics, two porous and one non-porous, stretched in dome configurations have been investigated for wind noise reduction. Two porous fabric domes are found to perform comparably to a porous hose array but without its amplitude- and phase-altering properties.
Experimental research on a sound insulation structure based on embedded Helmholtz resonators
Authors: Zeng Xinwu , Gao Dongbao , Gong Changchao
Published: May 2014

The structures containing Helmholtz resonators (HRs) have been demonstrated to be one of the metamaterials with negative parameters. Due to its local resonance, the one- and two-dimensional structures were indicated having local resonant forbidden band, which are possible ways to attenuate low-frequency acoustic waves and oscillations. At the resonant frequency, the incident acoustic energy is almost localized around the resonator, when a local resonant gap (LRG) appears at the end of the structure. Since the existence of LRG is only associated with the properties of single resonator, a wide band gap can be produced using periodically arrayed HRs with gradually changed parameters. In this structure, neighbor forbidden gaps are overlapped. A sound insulation device is designed and implemented in this paper based on embedded Helmholtz resonators (HRs). This structure is feasible to insulate acoustic waves transmitting above it. Numerical and experimental results both show that a low-frequency wideband insulation area can be formed by circularly arrayed HRs. The working band can be adjusted by changing the resonant frequency of the HRs in each layer.
A “better ear” listening strategy for improving speech-in-noise understanding in bilateral cochlear implant users
Authors: Kan Alan , Litovsky Ruth Y.
Published: May 2014

For bilateral cochlear implant (BiCI) users, understanding a target talker in noisy situations is difficult. Current efforts for improving speech-in-noise understanding have focused on improving signal-to-noise ratio (SNR) by multi-microphone techniques and signal processing, with only moderate improvement in performance. BiCI users typically report having a “better ear” for listening and recent data collected in our lab has shown that they have an asymmetry in speech unmasking performance. This work proposes a novel listening strategy for improving speech-in-noise understanding by combining: (1) a priori knowledge of a “better ear” and having a BiCI user selectively attend to a target talker in that ear; with (2) signal processing that delivers the target talker to the “better” ear and the noisy background to the opposite ear. We compared performance on a speech-in-noise test with and without this “better ear” strategy, using a virtual auditory space created from individualized head-related transfer functions. Subjects showed an improvement of at least 6 dB SNR in the speech reception threshold when using the “better ear” strategy, demonstrating that the strategy can boost speech-in-noise understanding for BiCI users. The novelty of this strategy is that it can be easily applied to other devices aimed at improving hearing.
Preliminary evaluation of a physiologically inspired signal processing strategy for cochlear implants
Authors: Swaminathan Jayaganesh , Goldsworthy Raymond L. , Zurek Patrick M. , Léger Agnès C. , Braida Louis D.
Published: May 2014

An approach to developing a cochlear-implant (CI) signal processing strategy inspired by healthy auditory physiology is described, and preliminary perceptual measurements with CI patients are reported. Envelopes were derived from a phenomenological auditory-nerve model (Zilany and Carney, 2010) that has been rigorously tested against physiological responses to both simple and complex stimuli. The model captures several physiological properties associated with nonlinear cochlear tuning including compression, suppression, and neural adaptation that are known to be critical for the representation of speech in quiet and noise. The envelopes were imposed on constant-rate current pulses via a programmable interface. Performance with this “neural” strategy was compared against the listeners’ personal processor and a standard encoder (ACE) processor implementation. Sentence, consonant and vowel identification were obtained from 5 CI users in quiet and + 6 dB of continuous speech-shaped noise. Overall, performance with the neural strategy was intermediate between that with subjects’ personal processors and the standard ACE processor. The closeness of results with the neural strategy to the listeners’ personal processor was viewed as promising as no individual tailoring of parameters were performed, and the subjects had no training with the neural strategy. Results from modeling efforts and future research directions will be discussed. [Work supported by NIH-NIDCD (R43-DC013006).]
Speech understanding in adults and children for sentences with cochlear-implant simulated spectral degradation and shallow insertion depth
Authors: Dougherty Sara , Waked Arifi , Goupell Matthew J.
Published: May 2014

While previous studies have compared cochlear-implant simulated (i.e., vocoded) speech understanding between adults and children, they have been acute experiments that have not considered long-term adaptation or training effects. Normal-hearing adult listeners demonstrate significant improvement in vocoded speech understanding with training, particularly if there is simulated frequency-to-place mismatch. The purpose of this study was to compare how adults and children adapt to eight-channel sine-vocoded speech with 0, 3, and 6 mm of frequency-to-place mismatch. Twenty adults (>18 yrs) and ten children (8–10 yrs) were trained on vocoded speech understanding over a four-hour period. The stimuli were a closed set of 40 simple words from which five-word nonsense sentences were constructed. Speech understanding was measured in 45-trial blocks where no feedback was provided, followed by 30-trial blocks where visual and auditory feedback was provided. High variability existed within both groups. On average, children performed worse than adults. Over the first five testing/training blocks, children improved at a slower rate than adults. Some children showed minimal improvement over the testing, whereas most of the adults showed noticeable improvement. These results suggest that adults have developed neural mechanisms that can more effectively adapt to and process degraded and frequency-shifted speech.
Discrimination of inconsistent interaural time differences across frequency in simulated bilateral cochlear-implant users
Authors: Rodríguez Francisco A. , Goupell Matthew J.
Published: May 2014

When using highly controlled low-rate stimulation, interaural-time-difference (ITD) just-noticeable differences (JNDs) vary greatly across frequency and individual bilateral cochlear-implant (CI) users. It is unclear how across-frequency variability would affect JNDs in multi-electrode stimulation where there are channel interactions from current spread. Therefore, we simulated variable across-frequency ITD sensitivity and channel interactions in 10 normal-hearing listeners presented with a CI simulation that consisted of 100-pulse/s, 1.5-mm bandlimited pulse trains (PTs) in three frequency regions (middle frequency = 7.6 kHz). The simulated electrode spacing was varied by placing the low- and high-frequency PTs Δ = 0.75, 1.5, 3, or 4.5 mm from the middle frequency. Inconsistent ITD sensitivity across frequency was simulated by applying non-zero ITDs in one, two, or three PTs, while the remaining PTs had zero ITD. The pulses in the PTs were simultaneous, nearly simultaneous, or maximally separated. In a left-right discrimination task, JNDs increased for a decreasing number of dichotic PTs. JNDs increased for Δ<3 mm, suggesting an important role of channel interactions. JNDs increased for non-simultaneous stimulation likely due to a rate limitation effect. The relatively poorer listener performance from the factors in our simulations suggest a means to predict multi-electrode ITD performance in CI users.
Recovery from forward masking of vowels and consonants: Effects of age and hearing loss
Authors: Bologna William J. , Fogerty Daniel , Ahlstrom Jayne B. , Dubno Judy R.
Published: May 2014

Vowels contain slow periodic fluctuations that contrast with aperiodic noise, which may favor perception of vowels over consonants with simultaneous and forward noise maskers. Older adults with or without hearing loss may be poorer at coding these temporal periodicity cues, which may limit their performance. In this study, younger adults with normal hearing and older adults with normal and impaired hearing identified consonants or vowels in the initial or final position of noise-masked syllables. Syllable and masker duration and relative timing ensured that the final 10, 40, or 100 ms of the syllable occurred after masker offset. Individualized spectral shaping minimized confounding effects of reduced audibility in regions of hearing loss. Preliminary results for initial-position phonemes indicate vowels are less susceptible to simultaneous masking than consonants. Recognition of final-position vowels is facilitated by 40-ms delay, whereas final consonants require 100-ms delay or longer for similar improvement. Younger adults benefit most from these delays, older adults with normal hearing benefit less, and older adults with hearing loss benefit least. These findings suggest that age and hearing loss contribute to prolonged recovery from forward masking, and that vowels have greater resistance to forward masking than consonants. [Work supported by NIH/NIDCD and ASHA.]
An algorithm to improve speech recognition in noise for hearing-impaired listeners: Consonant identification and articulatory feature transmission
Authors: Healy Eric W. , Yoho Sarah E. , Wang Yuxuan , Apoux Frederic , Youngdahl Carla L. , Wang DeLiang
Published: May 2014

Previous work has shown that a supervised-learning algorithm estimating the ideal binary mask (IBM) can improve sentence intelligibility in noise for hearing-impaired (HI) listeners from scores below 30% to above 80% [Healy et al., J. Acoust. Soc. Am. 134 (2013)]. The algorithm generates a binary mask by using a deep neural network to classify speech-dominant and noise-dominant time-frequency units. In the current study, these results are extended to consonant recognition, in order to examine the specific speech cues responsible for the observed performance improvements. Consonant recognition in speech-shaped noise or babble was examined in normal-hearing and HI listeners in three conditions: unprocessed, noise removed via the IBM, and noise removed via the classification-based algorithm. The IBM demonstrated substantial performance improvements, averaging up to 45% points. The algorithm also produced sizeable gains, averaging up to 34% points. An information-transmission analysis of cues associated with manner of articulation, place of articulation, and voicing indicated general similarity in the cues transmitted by the IBM versus the algorithm. However, important differences were observed, which may guide the future refinement of the algorithm. [Work supported by NIH.]
An analysis of the acoustic characteristics of forest sounds
Authors: Kyon Doo-Heon , Bae Myung-Jin
Published: May 2014

The purpose of this study is to define the general acoustic characteristics of forest sounds. Thus, large-scale measurements and analyses were conducted throughout four seasons for three main mountainous areas in Korea. The results showed there were clear differences in the acoustic characteristics, depending on environments and seasons. As acoustic elements of a forest, there are sounds of water from waterfalls and streams and sounds from birds and insects, and even sounds caused by stepping on snow in winter or fallen leaves in autumn have effect on the seasonal acoustic characteristics. The frequency of forest sounds was about 20 dB(A) smaller than typical sounds occurring in a downtown area, and forest sounds showed flat frequency characteristics in general. Especially, it was investigated that the energy ratio of their ultra-high tone sound domains was only 0.1% of that of a downtown area, while the rate of forest sounds was about 50 times more than that of a downtown area, and there were large differences found in the other frequency bands. Based on the results above, such acoustic differences between forests and downtown areas might have effect on the human health in a long-term view.
Effects of level roving and overall level on correlation change discrimination in naïve and trained listeners
Authors: Goupell Matthew J. , Barrett Mary E.
Published: May 2014

Sensitivity to interaural differences is remarkably variable across individuals, even highly trained listeners. In experiment 1, we measured sensitivity to changes in interaural correlation in 28 naïve listeners to determine the expected inter-individual variability for this task. Stimuli were 65-dB-A, 10-Hz narrowband noises with a 500- or 4000-Hz center frequency. Stimuli were tested either without level roving, or with ±5 dB of level roving in an effort to force listeners to attend to binaural width/movement cues rather than binaural loudness cues. At 500 Hz without level roving, the performance for the naïve listeners was much worse than the performance for the highly trained listeners previously reported in the literature. The addition of level roving significantly degraded performance, suggesting a close relationship between detection of decorrelation and loudness. At 4 kHz with and without level roving, none of the naïve listeners could perform the task. In experiment 2, we measured correlation change sensitivity as a function of overall level in trained listeners. Performance improved as overall level increased to about 65 dB-A, then worsened for more intense levels. The data will be discussed in terms of confusions between width and loudness cues, and in terms of Weber’s law.
Gradual decay of auditory short-term memory
Authors: Mathias Samuel R. , Micheyl Christophe , Shinn-Cunningham Barbara
Published: May 2014

Recent work suggests that once a representation of a visual object is committed to short-term memory, its precision remains fixed until it finally disappears without trace—a phenomenon called “sudden death.” In the present study, we investigated whether auditory representations also experience sudden death. In three experiments, listeners discriminated either the pitches or loudnesses of pairs of tones separated in time by up to 10 s. The data were analyzed using a model that allowed us to estimate both the precision of pitch/loudness representations and the probability of sudden death as a function of the interstimulus interval from listeners’ psychometric functions. Contrary to recent findings from vision, we found that auditory representations were no more likely to “die” after 10 s than 0.5 s; instead, they declined in precision (or “decayed”) gradually over time. The results point to a qualitative difference between how auditory and visual representations are forgotten from short-term memory.
Induced loudness reduction as a function of inducer level
Authors: Epstein Michael , Florentine Mary
Published: May 2014

Induced loudness reduction (ILR) is a phenomenon that occurs when the loudness of a stimulus (test tone) is reduced by the presence of a proceeding sound (inducer tone) that is presented at a level higher than the stimulus. This effect depends on a number of parameters, including inducer level. Whereas a number of studies have examined the effects of inducer level, there is no comprehensive data set examining the effects of a wide range of inducer levels on a wide range of test-tone levels. The present study fills this void by examining the effects of ILR for 500-Hz inducers when matching affected 500-Hz test tones at fixed levels of 5 dB SL and 20, 40, 60, 80, and 90 dB SPL with unaffected 2500-Hz level-adjusted comparison tones. Inducers primarily had an effect on test tones lower than the inducer level. Moderate-level tones exhibited the most ILR. This is consistent with prior studies examining single inducer levels. In addition, low- and moderate-level inducers appear to slightly increase the loudnesses of sounds at higher levels, indicating that the loudnesses of sounds at virtually every sound level within the dynamic range of hearing can be altered by ILR. [Work supported by NIH/NIDCD 1R03DC009071.]
Factors affecting auditory streaming of random tone sequences
Authors: Chang An-Chieh , Heo Inseok , Lee Jungmee , Stoelinga Christophe , Lutfi Robert
Published: May 2014

As the frequency separation of A and B tones in an ABAABA tone sequence increases the tones are heard to split into separate auditory streams (fission threshold). The phenomenon is identified with our ability to ‘hear out’ individual sound sources in natural, multisource acoustic environments. One important difference, however, between natural sounds and the tone sequences used in most streaming studies is that natural sounds often vary unpredictably from one moment to the next. In the present study, fission thresholds were measured for ABAABA tone sequences made more or less predictable by sampling the frequencies, levels or durations of the tones at random from normal distributions having different values of sigma (0–800 cents, 0–8 dB, and 0–40 ms, respectively, for frequency, level, and duration). Frequency variation on average had the greatest effect on threshold, but the function relating threshold to sigma was non-monotonic; first increasing then decreasing for the largest value of sigma. Differences in the sigmas for A and B tones tended to reduce thresholds, but covariance in the A and B tones had little effect. The results suggest that the principles of perceptual organization underlying streaming may differ for predictable and unpredictable tone sequences.
Bidirectional audiovisual interactions: Evidence from a computerized fishing game
Authors: Bensussen Seth , Chou Kenny F. , Varghese Lenny A. , Sun Yile , Somers David C. , Sekuler Robert , Shinn-Cunningham Barbara G.
Published: May 2014

We used a specially designed computer game to examine behavioral consequences of audiovisual integration. Target stimuli (animated fish swimming across the computer screen) were modulated in size and/or emitted an amplitude-modulated sound. Modulations, visual or auditory, were at 6 or 7 Hz (corresponding to “slow” and “fast”). In one game, subjects were instructed to categorize successive fish as “slow” or “fast” based on the auditory modulations; in another game, they categorized fish based on visual modulation rate. In each game, subjects were instructed to ignore input from the task-irrelevant modality. In each game, modulations could be (1) present only in the modality of interest, (2) present and matching in both modalities, or (3) present but mismatched between modalities. While reaction times were similar across games, accuracy was highest when auditory modulation was the basis for categorizing fish. Accuracy and reaction times improved when cross-modal modulation rates matched, and worsened when modulation rates conflicted. Additionally, accuracy was more strongly affected by between-modality congruence/incongruence when subjects attended to visual modulations than when they attended to auditory ones. Results indicate that audiovisual integration is not entirely under volitional control, and that competition between sensory modalities adversely impacts perception in dynamic environments.
Behavioral and neural measures of auditory selective attention in blast-exposed veterans with traumatic brain injury
Authors: Bressler Scott , Choi Inyong , Bharadwaj Hari , Goldberg Hannah , Shinn-Cunningham Barbara
Published: May 2014

It is estimated that 15–20% of veterans returning from combat operations have experienced some sort of traumatic brain injury (TBI), a majority of which are the result of exposure to blast forces from military ordinance. Many of these veterans complain of complications when understanding speech in noisy environments, even when they have normal or near normal audiograms. Here, ten veterans diagnosed with mild TBI performed a selective auditory attention task in which they identified the shape of one of three simultaneously occurring ITD-spatialized melodies. TBI subject performance was significantly worse than that of 17 adult controls. Importantly, the veterans had hearing thresholds within 20 dB HL up to 8 kHz and brainstem responses indistinguishable from those of the controls. Cortical response potentials (measured on the scalp using electroencephalography) from correctly identified trials showed weaker attention-related modulation than in controls. These preliminary results suggest blast exposure damages cortical structures responsible for controlling selective auditory attention (modulating sensory representations in cortex), and represent a step toward developing novel diagnostic methods to assess functional consequences of specific patterns of blast TBI damage in individual patients.
A bilingual advantage in the irrelevant speech effect
Authors: Dorsi Josh , Dominique Simmons Dominique , Cook Theresa , Rosenblum Lawrence , Laleko Oksana
Published: May 2014

The bilingual advantage refers to superior performance by multilingual over monolingual individuals on a variety of linguistic, and non-linguistic cognitive tasks. Tasks that include the bilingual advantage are ones that involve attention control, such as the Simon or Stroop tasks, but not tasks that involve short-term memory (Bialyskok, 2009). The Irrelevant Speech Effect (ISE) is the finding that serial recall accuracy of visual list items declines when task irrelevant speech backgrounds are present (relative to the disruption of white noise backgrounds: Colle and Welsh, 1976; Jones and Macken, 1993). The theories that have been offered to account for ISE are of two kinds, those that assume recall disruption is a function of limited memory, and those that assume it is the result of diffused attention (Elliot, 2002). As the bilingual advantage is found in tasks of attention, but not those concerning memory, a bilingual advantage in the ISE would support an attention based theory. In order to test this prediction, bilingual and monolingual subjects performed serial recall tests against speech and white noise backgrounds. Preliminary results suggest that the size of ISE is substantially smaller in bilingual, than it is in monolingual participants supporting attention-based theories of the ISE.
Air conduction and bone conduction in aging mice
Authors: Chhan David , McKinnon Melissa L. , Rosowski John J.
Published: May 2014

Umbo velocity and Auditory Brainstem Response (ABR) measurements in aged mice suggest that there are functional changes in both the inner ear and middle ear for frequencies from 5 to 13 kHz (Doan et al., 1996). In this work, we use a combination of air conduction (AC) and bone conduction (BC) stimulation to better quantify the middle ear contribution to age-related hearing loss seen in mice. ABRs were recorded with AC and BC stimuli from BALB/c mice of four different age groups (1, 2, 8, and 12 months); mice of this strain are widely used as models for age-related hearing loss. Results show the threshold stimulus levels for both AC and BC increase as the mice get older, consistent with age-related hearing loss. At frequencies below 12 kHz, the age-related changes in thresholds for all age groups are similar for both stimuli: the AC-BC difference (the air-bone gap) is not statistically significant. This suggests in this frequency range, the hearing loss is primarily sensorineural. At 16 kHz, the air-bone gaps of the two oldest groups are statistically significant suggesting the middle ear contributes to the hearing loss. Thresholds at higher frequencies were not measurable in the two oldest groups.
Can frequency discrimination be indexed by electrophysiological measures
Authors: Wang Wen-Jie , Martin Brett A. , Long Glenis R. , Tan Chin-Tuan
Published: May 2014

Acoustic change complex (ACC) responses to quadratic frequency glides following a steady pure tone were obtained from eight normal-hearing listeners. The base frequencies were 500 and 1000 Hz. The glides either increased or decreased in frequency by 200 Hz with durations of 50, 100, and 200 ms. The N1 response to stimulus onset and glide onset were evaluated to determine if the elicited ACC was associated with the frequency change in the glides. In contrast to the onset N1, the ACC N1 was delayed. This delay was used to index the time for frequency glides to reach a potentially detectable frequency change. ACC N1 latency was longer for glides of longer duration. However, the instantaneous frequency at ACC N1 peak latency was relatively constant regardless of glide duration. The difference in ACC N1 latency across glide durations was diminished when expressed as stimulus frequency, ratio of frequency change to base frequency, or associated place shift along the basilar membrane. There was also an effect of base frequency. ACC N1 latency was evoked when the percentage change from the base frequency exceeded ~0.2%. There was no clear evidence that the ACC is dependent solely on the duration or rate of frequency change.
Auditory cortex membrane potential dynamics in mice during challenging and ethologically relevant natural soundscapes
Authors: McGinley Mathew J. , Castellucci Gregg A. , McCormick David A.
Published: May 2014

Natural sound environments present numerous spectro-temporally complex and highly non-linear sound sources that an animal must assess for relevance. The auditory cortex is thought to be important for the execution of challenging sensory processing tasks. Yet, recordings in the auditory cortex have largely been studied in response to simple or synthetic sounds, or single well-isolated natural sound patterns such as con-specific vocalizations. In particular, membrane potential s in auditory cortex have only been recorded in response to simple stimuli, to which they respond with large rapid depolarization from a stable hyperpolarized baseline voltage. To help determine the dynamics of membrane potential fluctuations in natural conditions, we played natural soundscapes to head-fixed awake mice on a cylindrical treadmill while recording membrane potentials of pyramidal neurons the primary auditory cortex of mice. Ambient sounds were recorded from forests or meadows, particularly near creeks and during sunrise or sunset when numerous species were active. Sound frequencies encompassing the entire hearing range of mice (1–250 kHz) were recorded using a AviSoft UltraSoundGate recording system with CM16/CMPA microphone.
Aligning digital holography images of tympanic membrane motion
Authors: Guignard Jeremie , Cheng Jeffrey T. , Ravicz Michael E. , Rosowski John J.
Published: May 2014

Modern studies of sound-induced tympanic membrane (TM) motion include digital holography and optical-coherence tomography. These techniques generate high-definition maps of the motion of human and animal TMs that define and quantify spatial patterns of motion. In order to compare such patterns quantitatively between multiple specimens, a common coordinate system is necessary. Differences in the relative position of the sensor and the measured sample induce differences in magnification, shape, and location of the resulting TM image. A registration algorithm was implemented as follows: (1) The plot of the phase of motion at low frequencies was filtered with a moving 2D variance filter to differentiate the membrane from the bone in both reference and moving images (a process called segmentation). (2) A routine calculated the covariance matrix of the segmented images to measure its second central moment. (3) The images were aligned and warped on the basis of their centroid and moment. The method is about ten times faster than state-of-the-art intensity based registration, and the average squared errors of the pixel intensities after registration are not statistically different. The method is an efficient strategy to align TM surface images.
Resonant vibration control of a cylindrical shell using distributed tuned mass dampers
Authors: Page Christopher , Avitabile Peter , Niezrecki Christopher
Published: May 2014

Often structure borne noise has a deleterious effect on the performance and aesthetics of many commercial and military systems. Solutions to structural noise problems can be broadly classified into three categories: active, semi-active, and passive. With ever increasing improvements in electronics and computing, much of the recent research emphasizes active and semi-active solutions. However, passive solutions remain favorable in practice as they are often less costly to implement and less imposing on the system architecture. The tuned mass damper (TMD) is a common passive approach. Tuned mass dampers are generally used to affect a single undesirable resonance or forcing frequency. In order to affect multiple resonances multiple tuned absorbers are typically required, causing the design and implementation to be prohibitively difficult. This paper develops the concept of a modally enhanced dynamic absorber (MEDA) as a dynamic design philosophy, whereby the mass of peripheral equipment can be utilized and employed in attenuating the structural response of a larger system in some desirable way. The concept is demonstrated numerically and experimentally on a steel cylindrical shell. The importance of TMD attachment location with respect to the mode being de-tuned, robustness of TMD damping, and the effect of TMD mistuning is studied.
Audio frequency response of glass flat panel distributed mode loudspeakers
Authors: Roessner Stephen , Anderson David , Bocko Mark F.
Published: May 2014

The development of glass flat panel distributed mode loudspeakers (DML’s) will enable compelling new applications, such as flat panel displays that double as loudspeakers, or in architectural and automotive applications, window glass that can produce sound or actively cancel environmental noise. In this paper we discuss the frequency response of glass flat-panel distributed mode loudspeakers. The density of panel bending modes, their quality factors (Q’s), and the modal radiation efficiencies determine a DML’s frequency response. For example, a 0.55 mm thick cover glass for a 55 in. television has more than 10 000 bending modes in the audible frequency range. The density of modes increases with frequency and above a threshold frequency, determined by the panel dimensions and the mode Q’s, the mode spacing becomes less than the width of the individual modes, so the overall frequency response approaches the smooth, flat response of an ideal pistonic loudspeaker. However, below this frequency, discrete modes produce prominent peaks in the frequency response. We present simulated and measured glass panel frequency responses, including the effect of a thin layer of trapped air behind a DML panel, and we discuss the need for high internal friction (low Q) glass.
Optimization of spherical numerical integration methods in reconstruction of sound field with spherical microphone array
Authors: Li Minzong , Jin Jiangming , Lu Huancai
Published: May 2014

Choosing an effective spherical numerical integration method is one of the key problems in reconstruction of incident sound field by using spherical microphone array based on spherical near-field acoustic holography (SNAH). Many different spherical numerical integration methods are reported, such as I.H.Sloan, Spherical-t design, J.Fliege, etc., each method applies different distributions of discrete positions and integral weights in the design of spherical microphone array. In the paper, the accuracies of sound pressure reconstruction results based on SNAH with different spherical numerical integration methods are examined and compared. It is found that spherical-t integration method can provide more options of discrete points, simple algorithmic due to equal weight of each point, and the same or higher precision of sound pressure reconstruction. Meanwhile, the expansion terms of reconstructed sound pressure has to be truncated at an optimal number in order to minimize the impact of spherical numerical integration error, the approach to choosing the optimal truncated term number with respect of the number of microphones in spherical numerical integration is discussed in details.
Optimal research on volume velocity-matching wave superposition method based on tripole
Authors: Wu Shaowei , Xiang Yang
Published: May 2014

A volume velocity-matching wave superposition method for sound prediction of vibrating structure based on tripole is proposed. In the method, the position of the equivalent sources has similar geometry shape with radiating structure using a scale coefficient K. The calculating cut-off frequency of structure under a meshing pattern is predicted by setting 1.5% volume velocity relative error limit using surface vibration velocity. The cut-off frequency is defined as the maximum value of all wavenumber ka, at which the relative error just exceeds 1.5% when K iterates through (0,1) with step K. Then, for calculating frequency range [k1,k2], the optimal position of the equivalent sources, in which the relative error of volume velocity is minimum, is determined in the calculating cut-off frequency range by searching K in set {K|K,ka≥k2}. The transfer matrix between pressure and surface volume velocity is constructed in the optimal position by using tripole as equivalent source. After that, the sound radiation of structure is predicted to get high precision prediction. The results of numerical simulation show that the method is good at predicting the sound radiation of structure and the predicting error is very low within the cut-off frequency range once the optimal position is determined.