bluuebunny/crossref_metadata_2025 huggingface dataset:
Wind noise reduction at infrasound frequencies using large domes
Authors: Talmadge Carrick L.
Published: April 2018

One ongoing challenge to the use of infrasound for long-range monitoring applications (e.g., remote explosions or natural sources of infrasound such as tornados, volcanos or hurricanes) continues to be the high noise floors experienced at many permanent infrasound stations. International Monitoring System (IMS) infrasound stations typically address this problem using large pipe arrays as wind noise filters. However, these arrays generate significant distortion of higher frequency signals, making them non-ideal signals with significant frequency content above a few Hz. Further, they do not produce sufficient noise reduction during windy (typically daytime) measurements, making their utility generally limited to very quiet wind-noise periods (typically nighttime). Here we report on the development history of a 20-foot dome intended for permanent infrasound installations, and on measurements collected from arrays at the University of Mississippi Biological Field Station and at the Sandia National Laboratory Facility for Acceptance, Calibration, and Testing (FACT) site. The 20-foot dome at the FACT site is collocated with an IMS 18-m rosette filter and a University of Alaska Fairbanks experimental array, and more than a year of data has been collected, allowing for detailed comparisons of the efficacy of these different wind-noise reduction schemes.
Temporal dynamics of the generator of stimulated otoacoustic emissions
Authors: Verhulst Sarah , Altoè Alessandro , Raufer Stefan , Charaziak Karolina , Shera Christopher
Published: April 2018

The early emission work of David Kemp investigated temporal properties of click-evoked otoacoustic emissions (CEOAEs). Suppressor clicks were positioned up to 10 ms before or after the evoking (test) click and reduced the emission amplitude. Intriguingly, suppressors that preceded the test click by 1-2 ms were more effective than simultaneously presented suppressors. This observation seems not to support the hypothesis that emission generation operates on the basis of an instantaneous gain/suppression mechanism, for which simultaneously presented suppressors should be most effective. Kemps’ observations thus left the field with an important OAE generation question that inspired several OAE studies. Kemps’ results can be explained on the basis of (i) a non-instantaneous gain mechanism at the OAE generation site, or (ii), complex temporal interactions of basilar-membrane impulse responses (BM IRs) that operate with an instantaneous nonlinearity, but yield the observed non-instantaneous suppression properties. We investigated these dynamics using a nonlinear transmission-line model of the human cochlea and found that maximal CEOAE suppression can occur for preceding clicks even when the cochlear nonlinearity is kept time-invariant, supporting hypothesis (ii). Additionally, we used the frequency-dependence of CEOAE suppression to quantify human BM IR duration and cochlear filter tuning, yielding QERB estimates of 13.8 F[in kHz]0.22.
Intracochlear evidence on generation and propagation of distortion product otoacoustic emissions in gerbil cochlea
Authors: Dong Wei
Published: April 2018

Otoacoustic emissions (OAEs, Kemp, 1978) are complex signals appearing as a sum of multiple components relating to different cochlear source regions and different active mechanisms thus making it difficult to ascribe them to specific hearing frequencies. Improving the utility of OAEs requires understanding the origin of these multiple components and extracting the information they carry to the ear canal. The current presentation aims to provide direct intracochlear observations on OAE generation and their transmission. Two- or three-tone evoked pressure responses were simultaneously measured in the gerbil ear canal and scala tympani. Distortion product otoacoustic emissions (DPOAEs) and their intracochlear sources as DPs were analyzed. Results demonstrated: 1) DPOAEs come mainly from the f2 peak region, and extend further basal with increases in primary-tone intensities. These findings are consistent with several notions including: the theoretical prediction of “wave-fixed” or nonlinear generation, and the location of cochlear amplifier; and 2) DPs post-generation may travel both forward to their own best frequency places and in reverse to the stapes. The reverse propagation of DPs was mainly through reverse traveling waves, even though compression waves may also contribute to DPOAE generation, especially, to those produced close to the stapes in the basal cochlear region.
The role of the tectorial membrane in cochlear mechanics
Authors: Grosh Karl , Nankali Amir , Sasmal Aritra
Published: April 2018

The tectorial membrane (TM) is strategically located in the mammalian cochlea, anchoring the apical pole of the outer hair cell stereocilia and hovering just above the sensory inner hair cell stereocilia. Genetic mutations of TM-specific proteins cause nonsyndromic deafness in humans while experiments in mutant mice clearly demonstrate the impact that manipulations of TM structural proteins have on hearing thresholds and cochlear sensitivity. This direct evidence shows the importance of this cochlear structure. Mathematical models (with Zwsilocki as an early proponent) have long incorporated the TM but have not included a complete electrical and fluidic coupling. We show that an electromechanical model of the active processes in the cochlea combined with a fluid-structure model that strongly couples the basilar membrane and the TM to the cochlear fluids is crucial in understanding the low and high frequency response in a base-to-apex model of the cochlea. Specifically, we show that the onset of nonlinearity at high frequencies is due to a radial resonance of the uncoupled TM and investigate the influence of fluid-loading on the TM and TM viscoelasticity on the low-pass behavior seen in the apex. A connection of this new wave bearing mechanism to otoacoustic emissions will be discussed.
Localization of adventitious respiratory sounds
Authors: Henry Brian , Royston Thomas J.
Published: April 2018

In a recent publication [Henry and Royston, J. Acoust. Soc. Am., 142, 1774–1783 (2017)], an algorithm was introduced to calculate the acoustic response to externally introduced and endogenous respiratory sounds within a realistic, patient-specific subglottal airway tree. This work is extended using an efficient numerical boundary element (BE) approach to calculate the resulting radiated sound field from the airway tree into the lung parenchyma taking into account the surrounding chest wall. Within the BE model of the left lung parenchyma, comprised of more than 6,000 triangular surface elements, more than 30,000 monopoles are used to approximate complex airway-originated acoustic sources. The chest wall is modeled as a boundary condition on the parenchymal surface. Several cases were simulated, including a bronchoconstricted lung that had an internal acoustic source introduced in a bronchiole, approximating a wheeze. An acoustic source localization algorithm coupled to the BE model estimated the wheeze source location to within a few millimeters based solely on the acoustic field at the surface. Improved noninvasive means of locating adventitious respiratory sounds may enhance our understanding of acoustic changes correlated to pathology, and potentially provide improved noninvasive tools for the diagnosis of pulmonary diseases that uniquely alter acoustics.
The effect of visual representation of a room on perceived reverberation
Authors: Schutte Michael , Ewert Stephan D. , Wiegrebe Lutz
Published: April 2018

In everyday environments, sound reflections from walls and other objects arrive at the listener’s ear in addition to the direct sound radiated from a source. Humans have evolved mechanisms to perceptually suppress distracting early reflections, summarized as the precedence effect. Visual information of the surroundings influences the effectiveness of this mechanism. It has also been shown that similar compensation effects can occur for later reflections and reverberation, and that humans can estimate acoustic properties of a room from the visual impression. Taking these findings together, we hypothesize that the visual impression of a room affects the perception of its reverberation. The hypothesis was tested in a highly immersive audio-visual virtual reality environment consisting of a horizontal loudspeaker array in an acoustically treated chamber and a head-mounted display. In a magnitude estimation paradigm, subjects judged the perceived degree of reverberation in conditions where the simultaneously presented acoustic and visual stimuli were either matched regarding the room, sound source azimuth, and sound source distance, or diverged in one of those aspects. Audio-only control conditions served as a baseline. Preliminary results from six subjects suggest a predominant reliance on auditory input when quantifying perceived reverberation even in audio-visually conflicting conditions.
A correlation-based binaural localization model with reflection identification using neural networks
Authors: Deshpande Nikhil , Braasch Jonas
Published: April 2018

This model identifies a lateralized direct sound and single reflection. Anechoic speech is lateralized using a head-related transfer function. The speech is then time-delayed and lateralized to a different angle. The model then utilizes a binaurally-integrated cross-correlation/auto-correlation mechanism (BICAM) to analyze the lead/lag stimulus and generate a band-limited binaural activity pattern. This output is analyzed to calculate the time delay of the reflection, and the model then uses a neural network to estimate the lateralization of both the direct and reflected sound. From here, the model can remove the reflected copy of the sound, and extract the original raw, anechoic speech. [This work was supported by the HASS Fellowship at Rensselaer Polytechnic Institute, and the National Science Foundation Grant No. NSF BCS-1539276.]
Assessing cognitive and perceptual abilities by manipulating task requirements when listening to speech in noise
Authors: Miller Rachel E. , Fogerty Daniel
Published: April 2018

Speech recognition involves both cognitive and perceptual processing that can be differentially affected by the task requirements. This study assessed the degree to which different cognitive and perceptual abilities explained speech recognition using the Coordinate Response Measure (CRM). Speech recognition was assessed by varying the requirements of attention (i.e., selective or divided attention), and by assessing perceptual processes related to speech glimpsing (i.e., steady-state noise versus speech-modulated noise). Additional cognitive and perceptual tasks assessed cognitive interference, working memory, and amplitude modulation detection at 4 and 10 Hz. Results demonstrated that masking release (was most associated with modulation detection, indicating that the ability to take advantage of momentary dips in the masker (i.e., glimpsing) requires better amplitude modulation detection abilities. In contrast, when attentional constraints were imposed, a measure of cognitive interference was most predictive of performance for reporting keywords spoken by both the target talker (i.e., accurate responses) and competitor (i.e., intrusion responses). Interestingly, our measure of working memory was not correlated with any of the CRM tasks. These results suggest that conditions in which the CRM task is administered (i.e. manipulating noise or attention) can differentially weight separate perceptual and cognitive skills. [Work supported, in part, by NIH/NIDCD.]
Listening to degraded speech can cause listeners to “wait and see”
Authors: Gianakas Steven P. , Winn Matthew
Published: April 2018

The “wait and see” approach is described as waiting until hearing the end of a sentence or word before committing to a response. Although people with normal hearing (NH) perceive speech in a very rapid incremental fashion, listeners with cochlear implants (CIs) appear to utilize the wait and see approach in tests of word recognition. Specifically, they not only present with lower accuracy, but do not respond (even with eye gaze) until the utterance is fully or nearly complete. This implies that in addition to poorer accuracy scores, CI listeners might be adopting a different, more reluctant approach to word recognition. To better understand why listeners may adopt a reluctant listening strategy, we examined word recognition with a spectrally degraded speech signal in NH listeners, where eye gaze was tracked as a real-time indicator of commitment to a response. Preliminary results show a decrease in accuracy and a delay in response time for NH listeners when the signal is vocoded. However, the delays were not as extreme as those observed in listeners with CIs. These findings would suggest listeners hearing in a severely degraded condition may change their approach to listening, but only after extensive experience adjusting to the signal.
ejωt and the time domain in Introductory Engineering Acoustics
Authors: Grosh Karl
Published: April 2018

Introductory classes in engineering acoustics traditionally focus on frequency domain solutions rather than those in the time domain. While this is somewhat historical in nature, it is also true that extensive coverage of analytic techniques for solving space-time partial differential equations is challenging in one semester. This is not a desirable situation because qualitative and quantitative understanding of the link between the frequency and time domains is often elusive to the new learner of acoustics. Such a deficit stands as a barrier to their deep understanding of the material as well as a knowledge gap of ubiquitous and important techniques. Naturally, computational tools to perform these manipulations are available along with high fidelity digital to analog conversion on students’ personal computers. Students now have the capability to play back their simulations to their own ears to compare to experiment. I will present one approach to teaching these concepts to students with no assumed signal processing background using Fourier transforms. I will show how to weave a relatively complicated frequency domain simulation throughout the class and how this model problem can be demonstrated. This same problem also serves to introduce the effects of nonlinear distortion and clipping in transducers.
Classroom demonstration of acoustic landmine detection using a clamped soil plate oscillator
Authors: Cartron Jenna M. , Korman Murray S.
Published: April 2018

In landmine detection some mines (constructed out of plastic) are difficult to detect using ground penetrating radar. In acoustic landmine detection, one can detect a vibration profile across the surface of the soil (sand) due to the interaction between the soil column and the elastic top plate. Table top classroom demonstrations can bring to life idealized field experiments. The soil plate oscillator (SPO) apparatus models an idealization of an acoustic landmine detection experiment. The clamped circular elastic plate models the top plate of a plastic buried mine while the soil column supported by the elastic plate models the burial depth. An amplified sinusoidal chirp from a sweep spectrum analyzer drives an AC coil placed below a small magnet attached to the underside of the clamped plate—causing vibration. A laser Doppler vibrometer measured the rms particle velocity at 13 scan locations across the 4.5 inch diameter soil surface in sweeps between 50 Hz and 850 Hz. Results show a fundamental mode shape at 184 Hz (with a central peak) and a second mode at 488 Hz (with peaks to the left and right of the null) corresponding to a second vibration mode.
Effects of congenital blindness on the mismatch responses to Mandarin lexical tone, consonant, and vowel
Authors: Feng Jie , Wu Xinchun , Liu Chang
Published: April 2018

With limited visual input, blind individuals rely more on auditory information to perceive the environmental stimuli. In this study, we investigated the effect of congenital blindness on the Mismatch Negativity (MMN) to Mandarin lexical tones, consonants and vowels using the oddball paradigm. The study included 9 congenitally blind subjects and 10 normal-sighted subjects without hearing impairment, while they are matched on age and verbal IQ. For lexical tone processing, no significant difference was observed between the two groups in MMN latencies or amplitudes, while both groups had larger Mandarin tone MMN on the right and central than on the left. For consonant processing, the MMN amplitude was larger on the left and central than the right only for blind participants, whereas MMN latencies of consonants in both groups were shorter on the right than the left. For vowel processing, blind group had longer MMN latencies than control group. These results suggested that blind individuals had similar pre-attentive lexical tone processing, but they relied more on the left and central hemisphere to differentiate consonants, and it took them more time automatically to process different vowels than the sighted individuals, possibly resulting from the compensatory mechanism of neural reorganization in blind subjects.
Monitoring L-Dopa induced dyskinesias (LID) using speech acoustic measures in patients with Parkinson's disease
Authors: Wang Emily , Verhagen Metman Leonard A. , Zhan Andong , Blodgett Carly
Published: April 2018

More than 90% of patients with Parkinson's disease (PD) treated with levodopa for over 5 years will develop motor fluctuations and involuntary movements known as levodopa-induced dyskinesias (LID). Since LID commonly manifested in the head-neck and trunk areas speech is inevitably affected. In this study, eight acoustic speech measures derived from three speech tasks were selected to examine changes in speech acoustics associated with LID. Twenty subjects with PD, ten during the medication-off state without dyskinesia, and ten during the medication-on state with dyskinesia completed the speech tasks. Results revealed that three speech acoustic measures demonstrated sensitivity to changes associated with LID. In order to capture the changes associated with LID throughout medication cycles, ten PD subjects were provided an Android phone with HopkinsPD, an Android phone application, installed. One-minute speech and video samples were simultaneously collected at three time points over a medication cycle (before, peak, and end dose), twice a day (first morning dose and first afternoon dose) and repeated over three days (every other day) at a patient’s home. The results were discussed in light of the long-term goal to develop a reliable at home LID Smart-phone monitoring system that uses speech acoustic measures to monitor LID real-time remotely.
Vowel dynamics in persons with aphasia
Authors: Niziolek Caroline A. , Kiran Swathi
Published: April 2018

Although aphasia is primarily considered a disorder of language, higher-level grammatical and word-finding deficits are often accompanied by lower-level motoric deficits in the stable production of consonants and vowels. This project investigates the extent to which persons with aphasia (PWA) use feedback to monitor their speech by examining online formant movements during vowel production. PWA and age-matched controls produced 200 repetitions of three monosyllabic words (“eat”, “Ed”, “add”) while neural activity was recorded using magnetoencephalography (MEG). This “speak” condition was interleaved with a “listen” condition in which recorded audio from the speak trials was played back to participants through earphones, serving as a baseline for the neural response. Vowel centering was defined as 2D formant (F1-F2) movement toward the vowel median over the time course of the syllable, lessening acoustic deviation. PWA had greater acoustic variability than controls at vowel onset, but both groups exhibited vowel centering, significantly decreasing variability over the course of the syllable. A hemispheric shift in neural responses to self-produced speech may accompany this corrective movement in PWA. These analyses inform theories of feedforward and feedback influences on speech production in aphasia.
The role of rhythm in adaptation to dysrhythmic speech, continued
Authors: Borrie Stephanie A. , Lansford Kaitlin
Published: April 2018

Rhythm cues have been shown to be important for deciphering speech in adverse listening conditions, even when the rhythm cues are corrupted. In an initial attempt to document the relationship between rhythm perception and processing of a naturally dysrhythmic speech signal, we found that listeners with expertise in rhythm perception were not advantaged with initial intelligibility of ataxic dysarthria but were significantly advantaged with adaptation to the degraded speech signal [Borrie, Lansford, and Barrett, Journal of Speech, Language, and Hearing Research, 60, 561–570 (2017)]. We speculated that listeners with skills in rhythm perception are better able to exploit experience (familiarization) with the degraded speech, learning something useful about corrupted rhythm cues for subsequent processing. This current study investigated whether the relationship between rhythm perception and perceptual adaptation of dysrhythmic speech, observed in our previous study with ataxic dysarthria, holds for other forms of dysrhythmic speech. Here, we replicate our original study with two different forms of dysarthria: the largely irregular and unpredictable speech of hyperkinetic dysarthria secondary to Huntington’s disease, and the relatively regular and predictable speech of hypokinetic dysarthric speech secondary to Parkinson’s disease. Results will bear on models of dysrhythmic speech perception as well as clinical practice.
The impact of combined degradations: Intelligibility of disordered speech in noise
Authors: Yoho Sarah E. , Borrie Stephanie A.
Published: April 2018

In the current study, the intelligibility of dysarthric speech in background noise was determined. Speech-shaped noise was mixed with neurologically healthy (control) and disordered (dysarthric) speech at a series of signal-to-noise ratios. In addition, bandpass filtered control and dysarthric speech conditions were assessed to determine the effect of noise on both naturally and artificially degraded speech. Both the amount of noise and type of speech significantly impacted intelligibility, but there was no interaction between the two factors. Thus, it appears that there is no differential effect of noise on dysarthric speech relative to control speech. Despite this lack of interaction, it is important to note that the intelligibility of dysarthric speech was substantially lower than the intelligibility of control speech at each of the signal-to-noise ratios. This supports the idea that patients with dysarthria and their communication partners should be advised to select a favorable listening environment for communication. Lastly, large-scale online crowdsourcing via Amazon Mechanical Turk was utilized to collect data for the current study. Findings and implications for this data collection approach will be discussed.
A planar array of differential microphones for 3D sound capture
Authors: Abhayapala Thushara D. , Samarasinghe Prasanga N. , Chen Hanchi
Published: April 2018

Three dimensional soundfield decomposition based on spherical harmonic analysis is becoming an integral part of 3D audio for virtual and augmented reality systems. Spherical harmonic analysis reveals the underlying characteristics of the soundfield, thus allowing high accuracy manipulation and analysis of the soundfield. This requires a microphone array with 3D pick up capability to detect the soundfield. A well-studied type of such array configuration is the spherical array. Since its geometry coincides with the spherical harmonics, the sound signal captured by a spherical microphone array is well-suited for the spherical harmonic transform. The placement of microphones on a spherical array has to follow a strict rule of orthogonality of the spherical harmonics which limits the flexibility of the array configuration. The spherical shape of the array also pose difficulties on implementation as well as practical usage. Recently, the authors presented the theory and design of a compact hybrid microphone array distributed on a 2D plane for three dimensional soundfield analysis. The proposed array uses a combination of omnidirectional microphones and differential microphones placed along multiple circles on the X-Y plane. In this paper, we present the practical implementation of a prototype array based on the above design.
A raytracing method to create inhomogeneous wave fields for collaborative virtual reality systems
Authors: Braasch Jonas
Published: April 2018

A new rapid, raytracing method was developed and implemented for collaborative virtual reality systems. The goal of the method was to auralize historic acoustic venues in Rensselaer’s Collaborative-Research Augmented Immersive Virtual Environment Laboratory (CRAIVE-Lab). The CRAIVE-Lab provides a physical/digital environment for collaborative tasks using seamless video from multiple projectors and a 128-channel wave-field system with 6 additional ceiling loudspeakers. For the raytracing method, scanned floor plans are marked up by the user to create a two-dimensional geometric room model to calculate the temporal and spatial early-reflection pattern for the horizontal array of 128 loudspeakers resulting in an individual impulse response for each of the 128 loudspeakers. For this purpose, the loudspeaker array is virtually placed within the floor plan to calculate the impulse responses at each loudspeaker position considering the angles of incidence. The user can define wall materials and a directivity pattern for each sound source. The late reverberation tail is generated using a stochastic model based on the volume and surface characteristics of the space. An individual exponentially decaying reverberation tail is computed for each of the 134 loudspeakers with frequency-specific decay times. [Work supported by NSF Grant Nos. #1229391 and CISL.]
A computational framework for objective assessment of spatial audio wavefields
Authors: Bocko Mark F. , Crawford Steven
Published: April 2018

Fusion of binaural acoustic stimuli by humans to form spatial perceptions of virtual sound sources may be described in terms of the cross-coherence properties of a binaurally sampled acoustic wavefield. In this paper, a computational framework for predicting listener perception of acoustic wavefields is presented. A plane-wave expansion of the wavefield, with component amplitudes and phases described by random variables, provides a complete description of the wavefield’s spatio-temporal coherence properties. The acoustic source locations and the signals driving the sources, the listening space acoustic response, the location, head orientation, and physiological characteristics of the listener’s pinnae, head, and shoulders, and an explicit model of the human auditory system, including the peripheral auditory system and central binaural fusion process, comprise the full model. The source signals serve as inputs to the model and listener percepts indicating the locations and extent in space of the perceived virtual acoustic sources, are the outputs. The framework, which can accommodate constituent sub-models with varying levels of detail and sophistication, provides a useful framework to support the ongoing development of spatial sound rendering systems for applications such as cinema, teleconferencing, and virtual and augmented reality.
How physical versus panned sources in dry or reverberant conditions affect accuracy of localization in sound field synthesis systems
Authors: Catton Anna C. , Wang Lily M. , Bosen Adam K. , Vallier Timothy J. , Keefe Douglas H.
Published: April 2018

Sound field synthesis systems vary in number and arrangement of loudspeakers and methods used to generate virtual sound environments to study human hearing perception. While previous work has evaluated the accuracy with which these systems physically reproduce room acoustic conditions, less is known on assessing subjective perception of those conditions, such as how well such systems preserve source localization. This work quantifies the accuracy and precision of perceived localization from a multi-channel sound field synthesis system at Boys Town National Research Hospital, which used 24 physical loudspeakers and vector-based amplitude panning to generate sound fields. Short bursts of broadband speech-shaped noise were presented from source locations (either coinciding with a physical loudspeaker location, or panned between loudspeakers) under free-field and modeled reverberant-room conditions. Listeners used a HTC Vive remote laser tracking system to point to the perceived source location. Results on how physical versus panned sources in dry or reverberant conditions impact accuracy and precision of localization are presented. Similar validation tests are recommended for sound field synthesis systems at other labs that are being used to create virtual sound environments for subjective testing. [Work supported by NIH GM109023.]
Evaluation of near field distance perception in virtual environments
Authors: Robinson Philip W.
Published: April 2018

Binaural synthesis of spatial audio for virtual and augmented environments is typically performed by convolving an anechoic signal with impulse responses measured at the listener's ears using a source at a fixed distance. This method is accurate for sounds presented at the measured impulse response's distance, and for more distant sources, decreasing intensity provides a reasonable approximation. However, sources nearer than 1 m provide additional distance cues. The head-shadow effect becomes markedly stronger and exponential sound intensity falloff comes into play, exaggerating inter-aural level differences. Also, the angle to each ear diverges, changing the directionally dependent spectral filtering of the pinnae. Changes in inter-aural level differences can be approximated across listeners, but changes in pinna cues with changes in distance are highly individual, and thus less easily approximated. Reproducing some or all of these cues for each individual may be necessary to create a convincing percept of very near objects in virtual and augmented reality environments. The present work aims to determine the importance of each cue for static and dynamic sources. Listening tests have been conducted in a virtual environment using generic and individual head related impulse responses, with and without near-field compensation.
Improving the perception of a sound source's polar angle in mixed reality
Authors: Gamper Hannes , Tashev Ivan J.
Published: April 2018

Mixed reality applications blend real and virtual scenes. To render virtual objects in a scene, the rendering system needs to accurately control their perceived location. In the acoustic domain, the location of a sound source is often given in interaural coordinates, i.e., as the lateral and polar angle and distance relative to the midpoint of the listener's interaural axis. This description is useful as it allows separating the effect of various perceptual cues on each interaural spatial dimension. Prior research has shown that the human perception of a sound source's polar angle, i.e., the angle of rotation about the interaural axis, tends to be less accurate than the perception of its lateral angle, i.e., the angle off the median plane. When rendering virtual sound sources, listeners often confuse locations above and below the horizontal plane or in the front and in the back. Here, we review cues that affect the perception of polar angle as well as approaches to improve the accuracy of rendering the polar angle of a virtual sound source in mixed reality applications.
Localization and externalization in binaural reproduction with sparse HRTF measurement grids
Authors: Ben-Hur Zamir , Alon David L. , Rafaely Boaz , Mehra Ravish
Published: April 2018

High-quality spatial sound reproduction is important for many applications of virtual and augmented reality. A key component in spatial audio reproduction is the head-related transfer function (HRTF). To achieve a realistic spatial audio experience, in terms of sound localization and externalization, high resolution individualized HRTFs are necessary. However, these are typically unavailable as they require specialized equipment and a large number of measurements, which motivates the use of sparsely measured HRTFs. Reducing the number of measurement points requires spatial interpolation, and may lead to spatial aliasing error. Previous studies suggested the use of spherical-harmonics (SH) decomposition for the interpolation. With a sparse grid, the SH representation is order-limited, leading to a constrained spatial resolution due to the truncation error. In this study, the effect of sparse measurement grids on the reproduced binaural signal is perceptually evaluated, through localization and externalization tests under varying conditions of aliasing and truncation. Preliminary results indicate relatively large effect of SH order on these attributes, while smaller effect is observed due to aliasing error.
Characteristics of the under-ice soundscape in the southern Beaufort Sea during Ice Exercise 2016
Authors: Joseph John E. , Reeder D. Benjamin , Margolina Tetyana
Published: April 2018

Ice Camp SARGO was the remote hub of operations for the multi-national naval operation Ice Exercise 2016 (ICEX-16), held in March of that year. Over a three-day period in early March, continuous recordings of the under-ice soundscape were collected with receivers deployed at various depths through first-year ice in the vicinity of the ice camp as it drifted westward across the Beaufort Sea approximately 175 nm north of Prudhoe Bay, Alaska. A significant reduction in the strength of easterly winds resulted in deceleration of the ice sheet during the period of observation, inducing notable ice cracking and ridging events near the camp. Ice sheet movement slowed from about 0.5 knots early in the test to virtually coming to a halt near the end of the recording period. Sounds from naturally occurring and anthropogenic sources in the 10-Hz to 10-kHz band detected in the recordings were analyzed in connection to the origins of the sound and correlated to varying environment conditions including wind speed and ice motion. Results show a wide variety of persistent and transient sound sources contribute to the total soundscape at this remote location. Comparisons to other soundscape observations in the region are discussed.
Underwater noise measurements during a year long Shallow Water Canada Basin Acoustic Propagation experiment from 2016 to 2017
Authors: Badiey Mohsen , Lin Ying-Tsong , Pecknold Sean , Turgut Altan , Ballard Megan S. , Sagers Jason D. , Whitt Christopher
Published: April 2018

A year-long, multi-institution, acoustical oceanographic measurement on the Chukchi Continental Shelf region of the Canada Basin started on October 2016 is reported. Ten vertical receiver line arrays and a horizontal receiver line array were deployed together with oceanographic sensors to measure the sound speed profiles, ice formation, and currents over an area of approximately 30 km by 50 km. Various aspects of the noise including spectral fluctuations, directionality, seasonal dependence, and intensity fluctuations are studied over time. In this paper, we present preliminary measurement results depicting spatial and temporal distribution of underwater background noise at the experiment site. [Work supported by ONR321 OA.]
Studying melting icebergs with ambient noise oceanography
Authors: Glowacki Oskar , Deane Grant B. , Moskalik Mateusz
Published: April 2018

Marine-terminating glaciers are retreating at an unprecedented pace, largely as a result of enhanced submarine melting. However, studying ice-ocean interactions is complicated due to both harsh conditions prevailing in glacial bays and lack of scientific methods enabling continuous measurements. Recent studies have shown that high underwater noise levels measured in the Arctic are related to glacier melt, but quantitative research requires proper separation of individual noise sources, including icebergs and glacier fronts. Therefore, we show results of field experiments carried out in 2013, 2015, and 2016 in Hornsund fjord, Svalbard, to present directionality and statistics of the noise produced by melting icebergs. Measurements of noise directionality were conducted with 3-hydrophone acoustic array. Calculated angles of arrivals for the noise at the frequency range of 1–10 kHz correspond well to locations of individual, grounded icebergs. The amplitude of sound emitted by these sources has a symmetric α-stable distribution, with parameters depending on separation between iceberg and receiver. These findings demonstrate that ambient noise oceanography is an efficient tool to detect and track icebergs using natural noise they produce during melting. [Work funded by the Polish National Science Centre, Grant No. 2013/11/N/ST10/01729 and by ONR, Grant No. N00014-17-1-2633.]
Histotripsy—From nucleation of bubbles to tissue disruption
Authors: Fowlkes Jeffrey B.
Published: April 2018

Histotripsy utilizes focused ultrasound to nucleate gas bubbles deep within tissue and the associated bubble activity to induce biological effects at the level of tissue disruption/emulsification. Ultrasonic fields for inducing such effects include both pulses with as little as a single predominant negative half cycle to pulses with thousands of cycles and more. Different ensembles of bubbles leads to differing proposed mechanisms for tissue damage. Control of the damaged area is due to the threshold phenomena associated with the nucleation and the large focal gain in the main lobe, where the tissue disruption can be confined to fractions of the lateral beam width with little collateral damage. In addition, manipulation of the residual nuclei from a given ultrasound pulse can control cavitation events in subsequent pulses to allow greater and more uniform effects. The degree of tissue damage is equally impressive where disruption can be down to the subcellular level. At the same time, ultrasound parameters can be adjusted to selectively spare larger vessels from damage or retain the extracellular matrix. This variety of histotripsy properties opens up numerous potential medical applications, which will be discussed along with the conditions needs to achieve controlled and effective tissue disruption.
The effect of hypobaric pressure on the kidney stone twinkling artifact
Authors: Simon Julianna C. , Sapozhnikov Oleg A. , Kreider Wayne , Breshock Michael , Williams James C. , Bailey Michael
Published: April 2018

Recently, our group proposed the color Doppler ultrasound twinkling artifact originates from stable crevice bubbles on the kidney stone surface because overpressure suppressed twinkling on ex vivo stones (Lu et al., Ultrasound Med. Biol. 2013). However, the hypothesis is not fully accepted because the bubbles were not directly observed. Here, the submicron-sized bubbles predicted by the crevice-bubble hypothesis are enlarged in ex vivo kidney stones by exposure to a pre-focal, off-axis lithotripter pulse (p +  = 1.5 MPa, p-=2.5 MPa) or hypobaric static pressures (0.021 MPa, absolute) to simultaneously capture their appearance by high-speed photography and ultrasound imaging. In rough stones that twinkle, consecutive lithotripter pulses caused more than 50% of bubbles to grow reproducibly from specific locations on the stone surface, suggesting the bubbles were pre-existing. Conversely, on smooth stones that did not twinkle, repeated lithotripter pulses initiated bubbles from varying locations on the stone surface. Similarly, upon exposure to hypobaric static pressures, the simple expectation that twinkling would increase by enlarging bubbles largely held for rough-surfaced stones but was inadequate for smoother stones. These results suggest a correlation between kidney stone surface topography or stable surface crevice bubbles and twinkling. [Work supported by NSBRI through NASA NCC 9-58 and NIH DK043881.]
Application of acoustic fabrics to improve the insertion loss of a partial enclosure
Authors: Liu Weiyun , Herrin David
Published: April 2018

The sound absorption coefficient, transmission loss, and transfer impedance of different sound absorbing fabrics are measured. It is shown that the properties are similar to microperforated panels. Using microperforated panel equations, effective hole diameter and porosity are determined via a least squares curve fit. The effective parameters can then be used to predict the sound absorptive performance for different cavity depths. It is demonstrated that the fabrics have good sound absorptive performance over a wide range of frequencies. The fabric was then positioned in a partial enclosure. Single and double layers of the fabric were placed over an opening and the insertion loss as a result of fabric application was measured. The insertion loss is increased by approximately 5 dB if two layers of the fabric is used.
Structural vibration damping of a vacuum-assisted toilet
Authors: Rose Michael T. , Kilts Joshua F. , Pielstick B. Dagan , Sommerfeldt Scott D. , Gee Kent L. , Thomson Scott L.
Published: April 2018

Vacuum-assisted toilet noise can be unsettling and even uncomfortable. One common way to reduce noise levels is to damp structural vibrations that radiate sound. We investigated whether constrained layer damping (CLD) treatments could reduce the radiated noise level on a vacuum-assisted toilet. To find the structural anti-node locations of a toilet, we excited an airplane toilet with a shaker and scanned the velocity response of the inside of the bowl with a 3-dimensional scanning laser Doppler vibrometer (3D SLDV). We also scanned the bowl with an accelerometer during a repeated flush cycle. A microphone placed one meter above the bowl measured the radiated sound level. We applied 3M 4014, 3M 2552, Pyrotek Decidamp CLD, and Velcro individually to the bowl and determined the reduction in structural vibrations and sound radiation. The bowl’s rim on the front vibrated the most. Structural vibrational energy concentrated around 100-400 Hz while radiated sound concentrated around 400 Hz–2 kHz. Applying damping materials reduced structural vibrations, sometimes by 20 dB. We conclude that CLD treatments are able to reduce structural vibrations. Further results of the investigation will be shown and discussed.
Real-time source localization using phase and amplitude gradient estimator for acoustic intensity
Authors: Young Jacey G. , Lawrence Joseph S. , Gee Kent L.
Published: April 2018

Recent demonstrations of real-time source localization systems have used a variety of direction-finding methods. One such system uses a multi-microphone probe to estimate acoustic intensity [Inoue et al., Proc. Meetings Acoust. 29, 025001 (2016)], although the bandwidth of intensity estimates are traditionally limited by the microphone spacing. The Phase and Amplitude Gradient Estimator (PAGE) method [Thomas et al., J. Acoust. Soc. Am. 137, 3366–3376 (2015)] greatly extends the bandwidth of active intensity, providing an accurate method of source localization for a wide bandwidth. An initial system implemented in LabVIEW estimates active intensity using the PAGE method to identify the direction of the source location in real time. Using a paired webcam, the program then overlays the direction as a three-dimensional arrow onto a webcam image. A demonstration will be provided. Initial quantitative results show the accuracy of the system and limitations of the device. [Sponsored by the NSF REU program.]
Focusing airborne acoustic vortex beams by means of active diffraction gratings
Authors: Muelas-Hurtado Ruben D. , Ealo Joao L. , Pazos-Ospina Jhon F. , Volke-Sepulveda Karen
Published: April 2018

New technologies to generate acoustic vortex beams (VB) have received increasing attention due to their interesting capabilities to transfer angular momentum to matter and to attract particles to the propagation axis. One of the recently reported methods to generate acoustic VB is the fabrication of active diffraction gratings, i.e., acoustic sources whose geometry and vibration mode emulate the field obtained after passing a plane wave through a given passive grating [1]. In this work, we extend this approach by using an active spiral zone plate to produce a beam with a helical and focused structure. This is obtained by gluing an electroactive ferroelectret film on top of a lower electrode structured on a printed circuit board, which in turn results easy and reliable. The broadband feature of the film makes it possible to generate VB throughout the range of interest for most applications in air. Experimental results are compared with numerical simulations. The proposed ultrasonic source opens up new possibilities for particle manipulation and rotation control of small objects. [1] R. D. Muelas H, J. F. Pazos-Ospina, and J. L. Ealo, “Active diffracting gratings for vortex beam generation in air,” J. Acoust. Soc. Am. 141(5), 3569–3569, 2017.
Calibration and acoustic performance of sparse transducer array
Authors: Miller Lane P. , Thompson Stephen C. , Dittberner Andrew
Published: April 2018

In a previous research, a theoretical approach has been taken to predict, form, and optimize acoustic beam patterns from a sparse array of transducers. The current objective is to compare these theoretical results to experimentally obtained results from an assembled sparse array of transmit transducers. The array design and calibration procedure is presented. Acoustic beam patterns of the assembly are measured and compared to the theoretical model. Limitations of the array’s performance are evaluated.
Study on multi-channel speakers for controlling sound field of flat panel display
Authors: Lee SungTae , Park Hyung Woo , Bae Myungjin , Park Kwanho
Published: April 2018

Technological improvement in display industry, the organic light-emitting diode(OLED) panel manufacturer changed the quality of picture of the screen from simple high definition to an augmented-reality or a virtual-reality. And OLED that construct with the actual pixels and emit light by themselves are becoming widespread, making the thickness thinner and the bezel thinner or even disappearing. However, these technology improvement made it difficult to achieve sufficient performance in a small space for sound quality, which is less important component of TV. In this study, we have developed a multi-channel flat panel speaker which is able to synthesis the sound field. That flat panel speaker consist with direct drive actuator and diaphragm as an outer—glass layer of OLED. This direct drive actuator speakers are used to realize multi-channel sound from a single OLED panel. Multi-channel speakers can control the sound field to set the direction of sound. This feature is a good condition for implementing mixed-reality. In previous research, it was confirmed that the viewer 's sense of reality increased when the position of sight and hearing were matched. Furthermore, we have been developed to increase the mixed reality, to improve the acoustic performance by adding directionality to the sound generated by the panel.