bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of diluent fluid viscosity on acoustic droplet vaporization-mediated dissolved oxygen scavenging
Authors: Mercado Karla P. , Kalaikadal Deepak S. , Lorenz John N. , Manglik Raj M. , Holland Christy , Redington Andrew N. , Haworth Kevin J.
Published: June 2017

Acoustic droplet vaporization (ADV) can be used to scavenge dissolved oxygen and reduce the partial pressure of oxygen (pO2) in a fluid containing perfluoropentane droplets. The impact of the diluent fluid’s viscosity on ADV-mediated pO2 reduction was investigated. Polyvinylpyrrolidone (PVP) was dissolved in saline to modify the solution’s viscosity. The diluent fluid viscosity (η) and surface tension (γ) were measured. Droplets were manufactured using amalgamation and differential centrifugation to yield diameters between 1-6 μm. Droplets were diluted to 6.5x106 droplets/mL in saline (γ = 68 mN/m, η = 0.7 cP), 3 mg/mL PVP solution (γ = 65 mN/m, η = 1.2 cP), or 15 mg/mL PVP solution (γ = 65 mN/m, η = 4 cP). The viscosities of the 3 mg/mL and 15 mg/mL PVP solutions mimicked those of plasma and whole blood, respectively. Droplet solutions were exposed to ultrasound (5 MHz, 4.25 MPa peak negative pressure in situ, 10 cycles) in a 37°C in vitro flow system. The initial pO2 in the fluids was 113±2 mmHg, similar to human arterial pO2. After ultrasound exposure, the pO2 in saline, 3 mg/mL PVP, and 15 mg/mL PVP solutions were reduced by 39.9±0.8 mmHg, 31.9±0.7 mmHg, and 16.0±0.4 mmHg, respectively. These studies indicated that ADV-mediated pO2 reduction increased with decreasing viscosity.
The H-scan analysis and results in tissues
Authors: Parker Kevin J.
Published: June 2017

The H-scan is based on a simplified framework for characterizing scattering behavior, and visualizing the results as color coding of the B-scan image. The methodology begins with a standard convolution model of pulse-echo formation from typical situations, and then matches those results to the mathematics of Gaussian Weighted Hermite Functions. The nth successive differentiation of the Gaussian pulse G = exp(-t2) generates the nth order Hermite polynomial (Poularikas 2010). The function H4(t)G resembles a broadband pulse. Assuming a pulse-echo system has a round trip impulse response of A0H4(t)G, then we expect that a reflection from a step function of acoustic impedance will produce a corresponding received echo proportional to GH4(t). However, a thin layer of higher impedance, or a small scatterer or incoherent cloud of small scatterers would produce higher order Hermite functions as echoes. In this framework, the identification task is simply to classify echoes by similarity to either GH4(t), or GH5(t), or GH6(t). The resulting B-scan image is examined and echoes can be classified and colored according to their class. Results from tissue scans also demonstrate groups of echoes separated by Hermite order Hn. A theoretical framework is introduced where reflections are characterized by their similarity to nth order Hermite polynomials.
Optical imaging of propagating ultrasonic wave fronts resulting from ultrasonic pulses incident on heel bones using refracto-vibrometry
Authors: Huber Thomas M. , Huber Matthew T. , Hoffmeister Brent
Published: June 2017

Ultrasonic measurements of the heel bone (calcaneus) are used commonly for osteoporosis screening. Pulses emitted by an ultrasound transducer are incident on the calcaneus, and the transmitted wave fronts are detected with a separate transducer. In the current in-vitro study, full field videos of propagating ultrasonic wave fronts incident on a calcaneus sample, along with transmitted and backscattered waves were obtained using refracto-vibrometry. Pulses were emitted by a 500 kHz Panametrics V303 transducer. To optically detect ultrasonic wave fronts, the measurement beam from a Polytec PSV-400 scanning laser Doppler vibrometer laser was directed through a water tank towards a stationary retroreflective surface. Acoustic wave fronts (density variations) which pass through the measurement laser cause variations in the integrated optical path length between the vibrometer and retroreflector. The time-varying signals detected by the vibrometer at numerous scan points were used to determine the time evolution of ultrasonic wave fronts. The resulting videos enable visualization of the propagating wave fronts incident on the calcaneus and the backscattered and transmitted wave fronts. These videos enable direct investigation of wave front distortion due to reflection, refraction and diffraction effects for pulses transmitted through the calcaneus during ultrasonic heel scanning.
The native frequency of B-lines artifacts may provide a quantitative measure of the state of the lung
Authors: Demi Libertario , van Hoeve Wim , van Sloun Ruud J. , Demi Marcello , Soldati Gino
Published: June 2017

B-lines are ultrasound-imaging artifacts, which correlate with several lung-pathologies. However, their understanding and characterization is still largely incomplete. To further study B-lines, ten lung-phantoms were designed. A layer of microbubbles was trapped in tissue-mimicking gel. To simulate the alveolar size reduction typical of various pathologies, 166 and 80-micrometer bubbles were used for phantom-type 1 and phantom-type 2, respectively. A normal alveolar diameter is around 280 micrometer. A LA332 linear-array connected to the ULA-OP platform was used for imaging. Standard ultrasound imaging at 4.5 MHz was performed. Next, a multi-frequency approach was used: images were sequentially generated using orthogonal sub-bands centered at different frequencies (3, 4, 5, and 6MHz). Results show that B-lines appear predominantly with the phantom-type 2, suggesting a link between increased artifact formation and the reduction of the alveolar size. Moreover, the multifrequency approach revealed that the B-lines have a native frequency: B-lines appeared with significantly stronger amplitude in one of the 4 images, and spectral-analysis confirmed B-lines to be centered at specific frequencies. These results can find relevant clinical application since, if confirmed by in-vivo studies, the native frequency of B-lines could serve as a quantitative-measure of the state of the lung.
In-vivodemonstration of a self-contained ultrasound-based battery charging approach for medical implants
Authors: Makin Inder Raj S. , Radziemski Leon , Jabs Harry , Mast T. Doug
Published: June 2017

Ultrasound electrical recharging (USERTM) has been developed to demonstrate application-specific charging of a 200 mA-h Li-ion battery currently used in a clinical device for lower esophageal sphincter stimulation. In refining earlier developments [JASA 134(5) 4121, 2013], the receiver transducer and charging chip circuitry was miniaturized by an order of magnitude to a volume of 1.1 cc, the transducer attached directly to the 0.4 mm thick titanium device casing. Transmitter was a 1 MHz, 25 mm diameter piezo-composite disk, while the 1 MHz frequency matched receiver was either a 15 mm diameter disk or a 15 mm square tile. During a series of acute in vivo porcine experiments, the titanium prototype was implanted 10—15 mm deep in the subcutaneous tissue, the battery being successfully charged at a current of up to 75 mA, whereby the nominal transmitted RF power was 2 W. Maximum tissue temperature increase during the 4-hour charging cycle was 2.5 0C, directly in front of the receiver face, with no histologic thermal changes noted in the tissue post-mortem. The ultrasound approach with 10-15% system efficiency is a potentially favorable option for charging a Li-ion battery for next generation gastric stimulation implants. [Work supported by the NIH/NIBIB R43EB019225.]
Model-based ultrasound attenuation estimation
Authors: Ilyina Natalia , D'hooge Jan
Published: June 2017

Ultrasound attenuation coefficient (α) has shown a potential to provide quantitative information on the pathological state of the tissue. However, the main difficulty in the estimation of α consists in the need for diffraction correction that is currently done by means of a reference measurement. Previously, we proposed an alternative attenuation reconstruction technique, wherein the attenuation coefficient was estimated by iteratively solving the forward wave propagation problem and matching the simulated signals to measured ones. The simulation procedure involved modeling of the diffraction effects and allowed to avoid several assumptions made by conventional methods. The proposed method showed promising results when applied to the data recorded using a single-element transducer. In the present study, this methodology was extended to a case of a clinically used phased array transducer. The proposed approach was validated on simulated in Field II data and data recorded in a tissue mimicking phantom with varying focal position. For the simulated data, exact α estimates were obtained regardless of the focal position, while the relative error of α for the phantom data remained below 10 %. Currently, the performance of the proposed method is validated on the phantom an in-vivo liver data.
A thermoacoustic test rig for low-onset temperature gradients and the validation of numerical models
Authors: Avent Andrew W. , Bowen Christopher R.
Published: June 2017

The design and implementation of a low-onset temperature gradient standing wave thermoacoustic test rig is presented together with results which validate previous numerical models. A summary of on-going development of innovative fabrication methods and the novel use of composite materials for thermoacoustic stacks and regenerators is provided. An evaluation of the performance of these components is presented and further research and development work proposed. A novel design for selective laser sintered (SLS) aluminium thermoacoustic heat exchangers to deliver heat to from the thermoacoustic core is also presented together with numerical models and experimental validation from a purpose-built test rig.
Model based prediction of sound pressure at the ear drum for an open earpiece equipped with two receivers and three microphones
Authors: Vogl Steffen , Blau Matthias , Sankowsky-Rothe Tobias
Published: June 2017

In future hearing systems, one or more microphones and one or more receivers located within the earmold or ear canal are feasible. In order to predict the sound pressure at the ear drum in such a scenario, a one-dimensional electro-acoustic model of a prototype open earpiece with two integrated receivers and three integrated microphones was developed. The transducers were experimentally characterized by their (frequency-dependent) sensitivity (microphones) and Norton equivalents (receivers). The remaining acoustical system was modeled by 12 frequency-independent parameters which were fitted using a training set-up with well-defined loads at both sides of the ear piece. Put on an individual subject, the model could then be used to determine the acoustic impedance at the medial end of the earpiece, based on measured transfer functions between the integrated components. Subsequently, a model of the ear canal and its termination was estimated from the measured ear canal impedance, which could eventually be used to predict the drum pressure in the individual subject. Comparison to probe tube measurements of the drum pressure in 12 human subjects showed good agreement (less than ±3 dB up to 3 kHz, less than ±5 dB up to 6…8 kHz).
Coherence resonance and stochastic bifurcation in standing-wave thermoacoustic systems
Authors: Li Xinyan , Zhao Dan
Published: June 2017

In this work, we develop a noisy nonlinear model and conduct experimental tests to study the stochastic bifurcation and coherence resonance in standing-wave thermoacoustic systems. When white Gaussian noise is added and its intensity is varied, it is found that the stochastic behaviors of the system can be better described by stochastic P bifurcation. When the noise intensity is chosen as a bifurcation parameter, it is shown to the bi-modal region and reduce the bi-modal area are shifted. In addition, the noise-induced coherence motions are examined and confirmed. Resonance-like behaviors of signal to noise ratio (SNR) are clearly observed. When the system approaches the critical bifurcation point, SNR is found to become larger and the optimal noise intensity is decreased to a smaller value. This property can be used as a precursor to the Hopf bifurcation in standing-wave thermoacoustic systems. Experiments are then conducted on a Rijke-type thermoacoustic system with 3 loudspeakers implemented. Transition to instability is found to be subcritical. Comparison is then made between the present theoretical and experimental results. Good qualitative agreements are obtained in terms of (1) SNR, (2) the peak height of power spectrum, and (3) the width of the frequency.
Actively passive control of thermoacoustic instability
Authors: Zhao Dan , Tarique Ashique Akram , Li Shen
Published: June 2017

In this work, experimental studies of actively passive control of perforated liner with a bias flow on mitigating thermoacoustic instability are performed. For this, a well-designed Rijke-type thermoacoustic combustor with a perforated liner implemented is designed. A premixed propane-fueled flame is confined in the bottom half. A mean cooling flow (known as bias flow) generated from a centrifugal pump is forced to pass through the lined section. To maximize the damping capacity of the perforated liners, the bias flow rate is optimized by implementing a real-time tuning algorithm. The algorithm determines the optimum actuation signal to drive the centrifugal pump. On implementing the tuning algorithm, the unstable thermoacoustic combustor is successfully stabilized by reducing sound pressure level over 64 dB. To evaluate the off-design performance of the developed control approach, an extension tube is added/removed to the Rijke-type thermoacoustic combustor to give rise to the dominant unstable mode frequency being changed by approximately 17\%. It is found that the present actively passive control approach is able to mitigate the new limit cycle. And sound pressure level is reduced by about 41 dB. This confirms that the developed actively passive control scheme is sufficiently robust for use in real combustion systems.
Numerical simulations of the turbulent flow and the sound field of the Turkish ney end-blown flute
Authors: Fischer Jost L. , Bader Rolf
Published: June 2017

The Turkish ney is an end-blown flute in which sound generation is generated as an interplay between the air jet caused by the player and the sound pressure inside the flute tube. Sound pressure measurements inside the instrument show considerable nonlinear behavior which are crucial for the operation of the instrument and its sound character. Numerical simulations of the dynamics of both the turbulent flow field and the sound field solving the compressible Navier-Stokes equations, are performed. The transient process as well as the quasi steady-state operation mode of the instrument are performed. Varying the initial conditions of the blowing velocity as well as its attack time and shape result in the normal and in the overblown tones, which are characteristic of the instrument. Active elements like the turbulent air jet and rotating vortices around the labium as well as passive elements like the role of the mouthpiece and the resonator are discussed. A high agreement between numerical simulations and measurements is achieved.
Non-planar vibrations of an ideal string against a smooth unilateral obstacle
Authors: Kartofelev Dmitri
Published: June 2017

In this paper, we study the various possible motions of a string free to vibrate in two mutually perpendicular planes in the presence of a finite unilateral curved obstacle. We consider an obstacle which is curved only along the direction of the string at rest, and which is located at one of the ends of the string. The nonlinear problem of a non-planar string vibration against an obstacle is investigated using a kinematic numerical model under a number of simplifying assumptions. The complex interaction of the string with rigid obstacle is studied without the interfering effects of wave dissipation and dispersion. Also, it is assumed that no energy is lost due to friction and collision of the string with the obstacle. In this paper, we are especially interested in strings that are excited primarily parallel to surface of the obstacle. The modeling results show that presents of the obstacle changes dynamics of the string motion qualitatively. The conclusions of this studied idealized scenario are relevant to string vibrations in Indian stringed musical instruments like sitar or in Japanese shamisen. These lutes are equipped with finite curved bridges, and their strings are primarily excited parallel to those bridges.
High-resolution directivities of played musical instruments
Authors: Leishman Timothy W. , Strong William
Published: June 2017

Recent experimental developments have enabled directivity measurements of played musical instruments with high angular resolution. The measurement system assesses directional radiation characteristics while including diffraction and absorption effects of seated musicians. The results are advantageous to better understand and visualize sound produced by the instruments, provide benchmarks for physical modeling, predict and auralize sound in rehearsal and performance venues, and improve microphone placement techniques. This paper explores steady-state directivities, contrasting key differences between brass, tone-hole, and string instruments. As expected, brass instruments generate relatively predictable results because of their single radiation elements. Tone-hole instruments produce notable interference patterns due to radiation from multiple instrument openings. String instruments produce even more complex directivities because of radiation from distributed vibrating structures and instrument openings. To illustrate the effects, the paper focuses on an instrument from each category, within the same pitch range.
Improvement of method for tone wood properties examination using the very near field sound pressure scanning for mode visualization
Authors: Pantelic Filip , Mijic Miomir , Sumarac Pavlovic Dragana
Published: June 2017

The ability to predict behavior of some type of wood as part of a musical instrument is of a great importance. One of the important characteristics of a material, from the standpoint of musical acoustics, is Young's modulus of elasticity, which can be determined by observing the wooden beam test sample response to an excitation. In this process visualization of modes helps pairing frequencies from response spectra with vibration modes of sample, which allows us to numerically calculate Young’s modulus for different frequencies. In this paper, visualization of vibration modes are achieved using the microphone Very Near Field scanning of excited samples.
Deep convolutional neural networks for semi-supervised learning from synthetic aperture sonar (SAS) images
Authors: Chen Johnny L. , Summers Jason E.
Published: June 2017

Advancements in deep neural networks for computer-vision tasks have the potential to improve automatic target recognition (ATR) in synthetic aperture sonar (SAS) imagery. Many of the recent improvements in computer vision have been made possible by densely labeled datasets such as ImageNet. In contrast, SAS datasets typically contain far fewer labeled samples than unlabeled samples—often by several orders of magnitude. Yet unlabeled SAS data contain information useful for both generative and discriminative tasks. Here results are shown from semi-supervised ladder networks for learning to classify and localize in SAS images from very few labels. We perform end-to-end training concurrently with unlabeled and labeled samples and find that the unsupervised-learning task improves classification accuracy. Ladder networks are employed to adapt fully convolutional networks used for pixelwise prediction based on supervised training to semi-supervised semantic segmentation and target localization by pixel-level classification of whole SAS images. Using this approach, we find improved segmentation and better generalization in new SAS environments compared to purely supervised learning. We hypothesize that utilizing large unsupervised data in conjunction with the supervised classification task helps the network generalize by learning more invariant hierarchical features. [Work supported by the Office of Naval Research.]
Talker age estimation using machine learning
Authors: Berardi Mark , Hunter Eric J. , Ferguson Sarah H.
Published: June 2017

As a person ages, the acoustic characteristics of their voice change. Understanding how the sound of a voice changes with age may give insight into physiological changes related to vocal function. Previous work has shown changes in acoustical parameters with chronological age as well as differences between perceived age and chronological age. However, much of this previous work was done using cross-sectional speech samples, which will show changes with age but may average out important individual variability with regard to aging differences. The current study used a longitudinal recording sample gathered from a corpus of speeches from an individual spanning about 50 years (48 to 97 years of age). This study investigates how the voice changes with age using both chronological age and perceived age as independent variables; perceived age data were obtained in a previous direct age estimation study. Using the longitudinal recordings, a range of voice and speech acoustic parameters were extracted. These acoustic parameters were fitted to a supervised learning model to predict chronological age and perceived age. Differences between the chronological age and perceived age models as well as the usefulness of the various acoustic parameters will be discussed.
Automated assessment of bird vocalization activity
Authors: Kendrick Paul , Wood Mike , Barçante Luciana
Published: June 2017

This paper presents a method for the automated acoustic assessment of bird vocalization activity using a machine learning approach. Acoustic biodiversity assessment methods use statistics from vocalizations of various species to infer information about the biodiversity. Manual annotations are accurate but time-consuming and therefore expensive, so automated assessment is desirable. Acoustic Diversity indices are sometimes used. These are computed directly from the audio and comparison between environments can provide insight about the ecologies. However, the abstract nature of the indices means that solid conclusions are difficult to reach and methods suffers from sensitivity to confounding factors such as noise. Machine learning based methods are potentially are more powerful because they can be trained to detect and identify species directly from audio. However, these algorithms require large quantities accurately labeled training data, which is, as already mentioned, non-trivial to acquire. In this work, a database of soundscapes with known levels of vocalization activity was synthesized to allow training of the algorithm. Comparisons show good agreement between manually annotated and automatic estimates of vocalization activity in simulations and data from a field survey.
Improved feature extraction for environmental acoustic classification
Authors: Blevins Matthew G. , Bunkley Steven L. , Nykaza Edward T. , Netchaev Anton , Ochi Gordon
Published: June 2017

Modern automated acoustic classifiers have been shown to perform remarkably well with human speech recognition and music genre classification. These problems are well defined; there is a deep understanding of the source signal, and the required robustness of the model can be decreased without significantly sacrificing accuracy. Unfortunately, this simplification creates models that are insufficient when tasked with classifying environmental noise, which is inherently more variable and difficult to constrain. To further close the gap between human and computer recognition, we must find feature extraction techniques that address the additional set of complexities involved with environmental noise. In this paper, we will explore sophisticated feature extraction techniques (e.g., convolutional auto-encoders and scattering networks), and discuss their effect when applied to acoustic classification.
Deep learning for unsupervised separation of environmental noise sources
Authors: Wilkinson Bryan , Ellison Charlotte , Nykaza Edward T. , Boedihardjo Arnold P. , Netchaev Anton , Wang Zhiguang , Bunkley Steven L. , Oates Tim , Blevins Matthew G.
Published: June 2017

With the advent of reliable and continuously operating noise monitoring systems, we are now faced with an unprecedented amount of noise monitor data. In the context of environmental noise monitoring, there is a need to automatically detect, separate, and classify all environmental noise sources. This is a complex task because sources can overlap, vary by location, and have an unbounded number of noise sources that a monitor device may record. In this study, we synthetically generate datasets that contain Gaussian noise and overlaps for several pre-labeled environmental noise monitoring datasets to examine how well deep learning methods (e.g., autoencoders) can separate environmental noise sources. In addition to examining performance, we also focus on understanding which signal features and separation metrics are useful to this problem.
A likelihood-based method for point and interval estimation on auditory filter data
Authors: Christian Andrew
Published: June 2017

A novel method for determining auditory filter (AF) shapes given a set of n-alternative forced choice (nAFC) responses from a single human subject or set of subjects is discussed. The method works by developing a function which maps individual nAFC responses into likelihood values—either supporting or conflicting with a proposed model AF shape. The aggregate of these likelihoods is then used as an objective function for optimization schemes for point estimation, or as the basis function for Metropolis Hastings-like algorithms for interval estimation, both of either parameters of the AF model or of the entire AF shape. The method is demonstrated on simulated up-down staircase data. The consistency of the method is discussed in the context of canonical methods for AF data analysis, some of which are shown to produce systematic errors. Other possible benefits of this approach are discussed including the ability of the method to: combine data from heterogeneous nAFC tasks (e.g., notched-noise maskers with tone masking) into single AF models; combine data from different frequency-AFs into single analyses—shedding light on effects due to off-frequency listening; produce consistent results between different AF basis models.
Structural equation modeling of partial and total annoyances due to urban road traffic and aircraft noises
Authors: Gille Laure-Anne , Marquis-Favre Catherine , Lam Kin-Che
Published: June 2017

Data of a French in situ socio-acoustic survey were used to model partial annoyance due to urban road traffic noise, partial annoyance due to aircraft noise and total annoyance due to these combined noises. Structural equation modeling carried out on the in situ data showed that long-term noise annoyance depends on noise exposure but also on noise disturbance, dwelling satisfaction, visibility of a main road from the dwelling and noise sensitivity. Both noise exposure and noise sensitivity were introduced as independent variables in structural equation modeling of partial and total noise annoyances. Their contributions to the models highlight the necessity to consider these two variables in annoyance model prediction. Finally, in total noise annoyance models, whereas partial annoyance due to aircraft noise contributes to total noise annoyance as much as partial road traffic annoyance, aircraft noise exposure contributes to total noise annoyance much more than road traffic noise. Several reasons may explain this difference, such as the event character of aircraft noise or the fact that aircraft noise exposure reflects also the city exposure to aircraft noise. These hypothesis need to be confirmed on wider samples.
The quest for good, quiet spaces: Evaluating the relationship between office noise annoyance, distraction, and performance
Authors: Lawless Martin S. , Vigeant Michelle C. , Dittberner Andrew
Published: June 2017

To facilitate office work performance, acousticians must design spaces that minimize annoyance from background noise, primarily from HVAC equipment, and reduce worker distraction caused by intermittent sounds, e.g., ringing telephones. Increasing background noise can mask intermittent sounds and mitigate distraction, but negatively affects annoyance. Additionally, some disrupting sounds, such as alarms, contain informational content necessary for workplaces. Balancing worker annoyance and distraction can be difficult since the definition of what constitutes a good, quiet space is yet unclear. The goal of the present work was to perform a literature review to inform ideal office noise conditions and develop an experimental procedure to test such environments. The review included papers about indoor environmental quality and the effects of acoustics on environmental satisfaction, job performance, and noise annoyance, as well as cognitive, neurobehavioral, and physiological measures that can quantify work performance. The results of the literature survey will be used to form the basis of a future subjective study. In particular, an experimental design will be discussed that aims to evaluate the effects of various simulated environments, reproduced using higher-order Ambisonics, on work performance, annoyance, and distraction. The data from these future studies will be used to investigate ideal office acoustic environments.
Computer design, 3D printing, testing, and commercialization of a revolutionary machine gun suppressor (silencer) design*
Authors: Moss William , Anderson Andrew
Published: June 2017

Since their invention over 100 years ago, firearm suppressors have achieved acoustic suppression using baffles and chambers to trap and delay propellant gases from exiting the muzzle of a weapon. A modern suppressor is functionally identical to the original 1908 design, with most of the improvements made by lawyers trying to circumvent extant patents. We have produced a flow-through suppressor that functions completely differently from all previous suppressors. We used a few rapid design cycles of high performance computing, 3D printing of titanium prototypes, testing, and analysis to create our suppressor, which has been patented and licensed for commercialization. Ours is the only design to simultaneously limit blowback, flash, noise, and temperature. It will last the lifetime of the barrel on single shot and fully automatic weapons, requires minimal maintenance, and therefore, is the first practical suppressor for battlefield use. If adopted for general use, the main benefit would be the reduction of debilitating long-term hearing loss, one of the most prevalent injuries in the military. [This work was performed under the auspices of the U.S. Department of Energy by Lawrence Livermore National Laboratory under Contract DE-AC52-07NA27344.]
Speech intelligibility and spatial release from masking in maskers with different spectro-temporal modulations
Authors: Schubotz Wiebke , Brand Thomas , Ewert Stephan D.
Published: June 2017

Speech-reception thresholds (SRTs) decrease as target and maskers are spatially separated (spatial release from masking, SRM) even if two maskers are symmetrically placed around the listener's head. In this case, speech intelligibility (SI) cannot be explained by an improved long-term signal-to-noise-ratio (SNR) caused by the head shadow at one better ear alone, but could be facilitated by short-term spectro-temporal segments (“glimpses”) in each ear that provide favorable SNRs. Here it was systematically assessed how SRT and SRM depend on the spectro-temporal masker properties and on the availability of specific binaural cues for a frontal target in a symmetric masker setup. Maskers ranged from stationary noise to single, interfering talkers. Maskers were modified by head-related transfer functions providing different binaural cues (interaural level and time differences; ILD, ITD, both), by presenting only glimpses derived with a fast-switching better-ear mechanism, and an “infinite ILD,” removing crosstalk of the maskers between the ears. Results were compared to model predictions showing that spectral cues contribute to SRM for all maskers, while IPD and ILD cues were more important for modulated maskers. The “infinite ILD” condition suggests binaural processing limitations, resulting in a maximal SRM of 12 dB for low or absent informational masking.
The speech-based envelope power spectrum model (sEPSM) family: Development, achievements, and current challenges
Authors: Relaño-Iborra Helia , Chabot-Leclerc Alexandre , Scheidiger Christoph , Zaar Johannes , Dau Torsten
Published: June 2017

Intelligibility models provide insights regarding the effects of target speech characteristics, transmission channels and/or auditory processing on the speech perception performance of listeners. In 2011, Jørgensen and Dau proposed the speech-based envelope power spectrum model [sEPSM, Jørgensen and Dau (2011). J. Acoust. Soc. Am. 130(3), 1475-1487]. It uses the signal-to-noise ratio in the modulation domain (SNRenv) as a decision metric and was shown to accurately predict the intelligibility of processed noisy speech. The sEPSM concept has since been applied in various subsequent models, which have extended the predictive power of the original model to a broad range of conditions. This contribution presents the most recent developments within the sEPSM “family:” (i) A binaural extension, the B-sEPSM [Chabot-Leclerc et al. (2016). J. Acoust. Soc. Am. 140(1), 192-205] which combines better-ear and binaural unmasking processes and accounts for a large variety of spatial phenomena in speech perception; (ii) a correlation-based version [Relaño-Iborra et al. (2016). J. Acoust. Soc. Am. 140(4), 2670-2679] which extends the predictions of the early model to non-linear distortions, such as phase jitter and binary mask-processing; and (iii) a recent physiologically inspired extension, which allows to functionally account for effects of individual hearing impairment on speech perception.
A model predicting the effect of audibility on speech reception thresholds and spatial release from masking
Authors: Lavandier Mathieu , Buchholz Jorg M. , Rana Baljeet
Published: June 2017

A binaural model is proposed to predict the effect of audibility on speech reception thresholds (SRTs) measured in the presence of two (unintelligible) vocoded-speech maskers which were either (artificially) spatially separated or co-located with the frontal speech target. Comparing these two configurations allowed to evaluate a spatial release from masking (SRM) which was based here primarily on better-ear glimpsing. Audibility was varied by testing four sound levels for the combined maskers (while the target level was varied relative to these reference levels to measure the SRTs). The proposed model is based on a short-term binaural speech intelligibility model described by Collin & Lavandier [J. Acoust. Soc. Am. 134, 1146-1159 (2013)] and takes the calibrated target and masker signals (independently) at each ear as inputs along with the listener hearing thresholds in order to calculate a binaural “effective” signal-to-noise ratio. Differences in ratio across conditions can be directly compared to differences in SRT. This model allows a good prediction of the decrease of SRT as well as the increase of SRM with increasing audibility/levels. The averaged absolute error between data and prediction was about 1dB across tested conditions.
The interaction between reverberation and digital noise reduction in hearing aids: Acoustic and behavioral effects
Authors: Reinhart Paul , Zahorik Pavel , Souza Pamela
Published: June 2017

Digital noise reduction (DNR) is widely implemented in hearing aids to improve the signal-to-noise ratio (SNR) of speech-in-noise. To accomplish this, the DNR processor must be able to accurately discriminate the speech versus noise signals. Acoustically, reverberation causes the blending of the speech and noise signals. The purpose of the present experiment was to examine whether reverberation impacts the benefits of DNR processing. Speech stimuli were combined with white noise at multiple SNRs. Speech-in-noise signals were processed using virtual auditory space techniques to simulate reverberation times and with a DNR simulation that mimicked hearing aid processing based on spectral subtraction. Signals were acoustically analyzed to quantify changes in SNR as a result of DNR processing. As reverberant degradation increased, the improvement in SNR decreased. Behaviorally, hearing-impaired individuals listened to low-context sentences in noise with varying reverberation either with or without DNR processing. Without reverberation, DNR had no or minimal impact on speech intelligibility, consistent with previous work. However, as reverberant degradation increased, the effects of DNR on speech intelligibility were variable. These results suggest that the benefit of DNR processing in hearing aids in noisy environments may depend on the amount of reverberation in the environment. [Work supported by NIH.]
The effect of room acoustics on speech intelligibility and spatial release from masking
Authors: Biberger Thomas , Ewert Stephan D.
Published: June 2017

In daily life, verbal communication often takes place in indoor situations with interfering sounds, where speech intelligibility (SI) is affected by (i) masking and (ii) reverberation. Both introduce spectral and temporal changes to the signal. A critical spatial configuration to assess (binaural) SI is a frontal target speaker and two interfering sources symmetrically placed to either side (± 60°). Here a spatial release from masking (SRM) is observed in comparison to co-located frontal target and interferers, showing that the auditory system can make use of temporally fluctuating interaural differences. Room reverberation affects the temporal representation of the target and maskers and, moreover, the interaural differences depending on the spatial configuration and room acoustical properties. Here the effect of room acoustical properties (room size, T60, frequency dependency of T60), temporal structure of the interferers, and direct to reverberation ratio (DRR) on speech reception thresholds (SRT) and SRM were systematically assessed in a simulated room using headphone-based virtual acoustics. For constant T60 and DRR a different room size resulted in, e.g., significantly different SRTs but similar SRMs, implying the temporal structure of reverberation is less relevant for exploiting binaural cues. Data are discussed and compared to predictions of a binaural SI model.
Localization of speech in noise with and without self-directed room exploration
Authors: Clapp Samuel W. , Seeber Bernhard U.
Published: June 2017

The auditory system adapts to reflections by suppressing them in favor of the direct sound, aiding greatly in sound localization and speech understanding in reverberation. Previous studies have shown an increase in the echo threshold over repeated exposure to a reflection pattern, and the breakdown of this effect when the pattern changes suddenly. These studies examined this effect as a low-level cognitive process. The current study gives listeners a more active role in adapting to reflections, to see if this can engage higher-level processes more durable over time. In a psychoacoustic test, listeners localized a short speech signal in the presence of either point-like or diffuse noise in a simulated room, using the Simulated Open Field Environment for simulation and playback over a loudspeaker array. In the “Block” condition, subjects heard stimuli from the same room in all trials. In the “Interrupt” condition, trials in the target room were interspersed with dummy trials from different spaces. In the “Learning” condition, listeners used a GUI to position a virtual source in a room and hear the resulting auralizations before each block of trials, conducted as in the “Interrupt” condition. Localization was most accurate in the “Learning” condition, particularly in the presence of the point-like noise source, and for target locations the furthest angular distance from the noise. These results suggest it is possible for actively engaged listeners to learn the acoustics of a room in a temporally durable manner.
Binaural detection-based estimates of precision of coding of interaural temporal disparities across center frequency
Authors: Bernstein Leslie R. , Trahiotis Constantine
Published: June 2017

This presentation reports binaural detection data obtained using three masking configurations previously shown, in toto, to reveal: (1) whether listeners can compensate internally for external interaural temporal disparities (ITDs) and (2) the precision with which such compensation of ITD changes as a function of reference ITD [L. R. Bernstein and C. Trahiotis, J. Acoust. Soc. Am. 138, EL474-EL479 (2015)]. The current experiment extended that paradigm in order to obtain binaural detection data at center frequencies that ranged in octave steps from 250 Hz to 8 kHz. Results indicate that, at all center frequencies, listeners are able to compensate internally for externally imposed ITDs. Such compensation allows them to outperform what would be expected were they only utilizing changes in interaural correlation produced by the addition of the tonal signal to the noise masker. The data will be discussed in terms of quantitative analyses that show how detection performance is constrained by two distinct sources of interaurally and mutually uncorrelated internal noise: one having a power that increases with the power of the external masker and the other having a power that grows with the magnitude of the ITD.
A model of the ongoing precedence effect
Authors: Zurek Patrick , Freyman Richard L.
Published: June 2017

The precedence effect is the observation that, when a sound is repeated after a delay of a few milliseconds, the interaural cues of the leading sound have greater influence on localization and lateralization judgments of the composite sound than do the cues of the lagging sound. Classic studies of the precedence effect with transient stimuli have suggested auditory mechanisms that emphasize earlier-arriving cues. More recent studies have shown, however, that non-transient sounds—steady, long-duration noises with slow onset—also give rise to the precedence effect. This ongoing precedence effect is difficult to explain with the abrupt-onset mechanisms invoked to explain the effect with transients. The present model of the ongoing effect assumes that listeners use interaural cues within auditory bands to form band-specific lateral positions, which are then integrated across bands. Within-band positions are estimated from the interaural level difference and the interaural phase during the rising side of bandpass envelope modulations. Cross-band integration includes a mechanism that favors cues that are consistent across wider frequency regions. Predictions from the model will be compared to published data. [Work supported by NIDCD R01 01625.]
Specificity of adaptation to non-individualized head-related transfer functions
Authors: Romigh Griffin D. , Simpson Brian , Wang Michelle
Published: June 2017

Initial accuracy is poor when listeners are asked to localize virtual sound sources that have been low-pass filtered or rendered with non-individualized head-related transfer functions (HRTFs). However, Majdak et al. (2013) showed that, with training, localization performance improved when virtual sounds were rendered using HRTFs that were low-pass filtered at 8.5 kHz. This result suggests that previous outcomes showing training-induced improvement in localization performance with non-individualized HRTFs may merely be the result of listeners attending to low-frequency spectral information that is consistent with their own HRTFs, and not attending to the information from high frequencies, where HRTFs differ widely across individuals. That being the case, one would expect training to generalize to all non-individualized HRTFs, not just the non-individualized HRTF used during training. The current study investigated this hypothesis by performing tests of localization accuracy before and after three weeks of auditory localization training with a single non-individualized HRTF. For all subjects, localization performance with non-individualized HRTFs improved to a level at or near their performance with individualized HRTFs, and no generalization to the other non-trained HRTFs was found, suggesting subjects do learn to utilize an alternative set of high-frequency spectral information.
Non-conforming finite element method for efficiently computing multilayered microperforated plate absorbers
Authors: Kaltenbacher Manfred , Floss Sebastian , Metzger Jochen
Published: June 2017

The concept of acoustic impedance is a very useful model approach to efficiently compute configurations for sound absorption. Thereby, the measurements are performed by an impedance tube and the obtained data is used for the first order Robin (impedance) boundary condition within the numerical simulation. However, this approach is only valid for sound incidence perpendicular to the boundary. A second approach is to resolve the volume of the absorber and use a rigid or even elastic frame model. Especially for multilayered silencers based on microperforated plates (MPPs) the volume resolving approach is beneficial. Here, a main challenge is to cope with the quite different mesh sizes needed for accurately resolving the waves in the MPPs and the surrounding air regions. To efficiently simulate such designs, we apply a Nitsche-type mortaring within the Finite Element Method to allow for non-conforming meshes and thereby directly connect the different mesh sizes in the MPPs and surrounding air regions. We will discuss in detail our absorber design, the performed measurements and numerical simulations and plan to publish the complete setup and results as a benchmark case for computational acoustics.
Quantification of numerical damping in the acoustic boundary element method for the example of a traveling wave in a duct
Authors: Baydoun Suhaib K. , Marburg Steffen
Published: June 2017

The boundary element method (BEM) is a popular numerical method for solving linear time-harmonic acoustic problems. Using the BEM, modeling is limited to the boundary of the fluid domain, which is particularly advantageous for exterior problems with unbounded domains. A widely unknown drawback of the acoustic BEM is numerical damping. This work is concerned with numerical damping encountered in the benchmark problem of an air-filled duct with rigid walls. A traveling wave, induced by a particle velocity at the inlet, is fully absorbed at the outlet of the duct by imposing an impedance boundary condition. The exact solution gives a constant pressure amplitude over the entire frequency range. However, the numerical solution exhibits decay of the pressure amplitude, which is clearly an indication for numerical damping. This phenomenon is studied for different frequencies and elements-per-wavelength ratios. The extent of numerical damping is quantified by relating the decay to the pressure distribution obtained for a fluid model considering damping. The gained knowledge enables more accurate estimations of real damping phenomena in the future.