bluuebunny/crossref_metadata_2025 huggingface dataset:
Evolution of miniature microphones in hearing instruments
Authors: LoPresti Janice L.
Published: December 2020

The invention of the transistor enabled early hearing instruments to be miniaturized: from bulky body worn device to a device that was more visually appealing and could fit in the ear canal. This change also brought many technical challenges related to the miniaturization of the microphone components and creation of low power amplifiers to maintain battery life. In the early 1950s, magnetic microphones were considered a good choice with their smaller size and lower output impedance compatible with the transistor amplifiers of that time period. However a susceptibility to shock and vibration drove further innovation leading to first piezoelectric and then electret microphone designs. The next large step in the design evolution occurred in the early 2000s with the introduction of the MEMS (Micro-Electro-Mechanical Systems) microphone. The adoption of MEMS microphones in hearing industry is driven by multiple factors including environmental stability requirements for beam forming applications and reflow compatibility for manufacturing. Modern technology drivers such as sound quality, miniaturization, reliability, and advanced audio controls have pushed the limits of current technology to create best performing microphones today. This presentation chronicles evolution of miniature microphones in hearing instruments from first magnetic microphones to MEMS microphones in present day.
Acoustics of the French Horn
Authors: West Natalie , Ramsey Gordon P.
Published: December 2020

This study explored the French horn and its various properties that contribute to its unique timbre. With an emphasis on the role of the right hand, the variables studied were: type of material, manufacturer, right hand position with respect to the bell, the range of notes, and player. The project mainly included experiments at Loyola University Chicago and was supported using Northwestern University's anechoic chamber. Horns are cylindrical-conical hybrids and are able to produce many harmonics, creating a rich sound. The right hand in the bell can be used for quick intonation adjustments and its placement has large effects on the timbre. Our analysis determined the variable that most contributed to the unique sound was the right hand position, while the type of horn (a combination of material and manufacturer) and the range of notes played also had moderate effects on timbre. We will present the theory surrounding the horn's shape and various components, experimental procedure, and analysis of our results. The significance of the results to the horn player will be discussed.
Richard H. Lyon: Sound propagation, acoustic modeling, and the MIT A&V Lab
Authors: Donavan Paul
Published: December 2020

In the mid 1970s, the Acoustics and Vibration Laboratory at the Massachusetts Institute of Technology was an exciting place for any student interested in acoustics and noise control. As one of the directors of the lab, Professor Lyon had a profound influence on the activities, students, and researchers working and studying there. Among the other things that Dick is known for such as Statistical Energy Analysis, Machinery Noise Diagnosis, and Product Sound Quality, he was also overseeing research on urban sound propagation and applications of acoustical scale modeling. This presentation will discuss some of this research, the culture of the A&V Lab, and personnel interactions with Professor Lyon from one of his student's and researcher's point of view.
Application of event-based beamforming to the study of acoustic shocks in installed supersonic engine noise
Authors: Grow Jordan , Vaughn Aaron B. , Leete Kevin M. , Gee Kent L. , Wall Alan T.
Published: December 2020

Crackle is a dominant component of the noise from supersonic military-style jet engines. Because the study of crackle is tied to the characterization of acoustic shocks, examination of the locations and propagation direction of the largest-derivative waveform events in the near field can provide insights regarding source radiation and propagation characteristics. This paper describes initial results of applying an event-based beamforming algorithm to a ground-based microphone array located in the near-field of a T-7A Red Hawk aircraft with its afterburner-capable F-404 engine. The work builds on recent work by Vaughn et al. [AIAA Paper 2019-2664]; the algorithm identifies large-derivative events at pairs of adjacent microphones and determines the propagation direction by applying a cross correlation to short time segments around the events. Statistical analysis of these events, and associated beamforming, for the T-7A noise radiation leads to an improved understanding of the noise radiation of temporal events tied to acoustic shocks and, therefore, to crackle. [Work supported by AFRL.]
Dynamic human strategies for localizing tones in rooms
Authors: Hartmann William , Macaulay Eric
Published: December 2020

Reflections and standing waves in a room cause distorted binaural information, making it difficult for listeners to localize ongoing sound sources. Possibly head motion allows a listener to sample a distorted sound field, gaining dynamic information that enables more accurate localization. The present research tries to discover listener strategies for localizing pure tones using a variable acoustics environment. Over the course of 9-s trials, a head tracker recorded head position and orientation while probe microphones in the ear canals recorded the signals and interaural differences. Therefore, the experiment provided complete information. Two model strategies are considered. In the “nulling” strategy, the listener discovers temporally and spatially discrete head pointing directions where the dominant interaural difference is zero. Candidate directions are weighted by head-pointing perseverance in the vicinity of the interaural zero. In the “inferred source” strategy the listener registers temporally and spatially continuous inferred-source candidates based on instantaneous interaural differences and head pointing directions. Each candidate is weighted by the instantaneous ratio of head-angle change to inferred-source change evaluated at the candidate location. Model strategies are evaluated by comparing weighted candidate source locations with listener response locations.
How does temporal diffusion affect the ongoing precedence effect?
Authors: Pastore M. Torben , Braasch Jonas
Published: December 2020

In room acoustic scenarios, listeners' localization is often dominated by the sound propagating directly from the its source despite numerous reflections that present different spatial cues only milliseconds later. This is called the precedence effect (PE). Most studies have simulated the PE by presenting one sound (the lead) followed by a copy of the lead that is delayed and presented with different interaural cues (the lag). These simulations assume that reflective surfaces are flat, yet interior surfaces are often far more complex and variable, resulting in spatially and temporally diffuse reflections. The effect of the temporal aspect of this diffusion on listeners' localization of lead/lag, 200-ms duration, noise stimuli filtered to 100–900 Hz and presented over headphones is investigated. Lag stimuli are convolved with a Hanning-windowed 2-ms noise burst to simulate temporal effects of uneven reflective surfaces. Results show that listeners' localization is dominated by the interaural cues of the lead, even when gating onsets/offsets are windowed out. Modeling analyses based on those in Pastore and Braasch (2019) suggest that interaural time differences in the ongoing stimulus portion can be extracted from rising slopes of the envelopes of neural output, even when lead and lag envelopes are decorrelated.
Reverberation detection threshold estimates in normal-hearing listeners: Effect of direct-path level
Authors: Zahorik Pavel , Neal Matthew T.
Published: December 2020

Previous work [Zahorik, P. and Shehorn, J. J. Acoust. Soc. Am. 145, 1875 (2019)] has used virtual auditory space techniques to measure reverberation detection thresholds in normal-hearing listeners for three sound field conditions: an office-sized room (broadband T60 = 0.5 s), a concert hall (broadband T60 = 1.5 s), and a reference condition with a single echo at 40 deg to the right of midline. Although thresholds were found to vary by as much as 18 dB across conditions, they were well predicted by a monaural temporal window model that has been used to effectively predict various temporal masking phenomena. Here, the effect of direct-path level on reverberation detection was examined for the same three listening conditions. The source signal was a 220 Hz complex tone, 250 ms in duration. In general, masking of reverberation by direct-path sound was found to increase as a function of direct path level, but only in listening conditions with shorter reverberation times. Longer reverberation times resulted in little masking at any direct path level. These results are consistent with predictions from the temporal window model, and thus further validate its use for predicting absolute sensitivity to reverberation.
Numerical investigation of fully acoustic nondestructive testing method of delamination in glass fibre reinforced plastics
Authors: Hanuman Nudurupati S. , Bose Tanmoy
Published: December 2020

The contact based NDE methods are not suitable to inspect the structures at high elevated temperatures and large aerospace structures. Therefore, in recent days, acoustic based NDE methods are address to all these challenges. A local defect resonance (LDR) technique is one of the nonlinear techniques to get effective imaging of the defect. In this paper, the numerical work is presented for glass fibre reinforced plastic (GFRP) plate having single delamination. The abaqus software package is used to model the GFRP plate having single delamination. Explicit dynamic analysis is performed to get sound pressure levels (SPL) at defect location. Non-contact based acoustic incident wave is injected into the material through an impinging region for exciting the delamination. Impinging region is created over the one side surface of the GFRP plate, which is circular plane. Incident wave is created by using spherical co-ordinates about the center of the impinging region. Finally, optimize the spherical co-ordinates for effective excitation of delamination to get high sensitivity sound pressure levels at the delamination location. Further, the numerical work is validated with experimental work. The experiments are carried out on the GFRP plate. Corresponding sound pressure images are obtained. Finally, good judgment is achieved between the numerical and experimental work.
Measuring plate vibration using deflectometry: The advantages and limitations of add-on reflective material
Authors: Rhoades Gary , Shepherd Micah , Harris Jeff
Published: December 2020

Deflectometry is a full field optical method which utilizes the slope fields on the surface of a planar object to track deformations. The use of a high speed camera gives the ability to measure all points on the surface instantaneously and with high spatial resolution, providing more knowledge on how structures react to transient excitations. Here, the reflected grid method will be used, as it involves geometry that allows for simple calculation of curvature distributions. This method of deflectometry relies on specular reflections to create amplifications of the measured deformations in the test plate. Therefore, this requires the plate under test to have a reflective surface. An experiment was constructed to test the vibration of a flat plate excited by an automatic force hammer. A collection of adhesive tapes, films, and spray were applied to the test object in order to increase the reflectivity of the plate. The advantages and limitations of each type of add-on reflective material will be discussed. Finally, the add-on material was used on a more complex structure to showcase the potential of deflectometry for measuring plate vibration.
Non-contact air-coupled resonant ultrasound spectroscopy for characterization of soft structures
Authors: Khandelwal Aakash , Chakrapani Sunil Kishore
Published: December 2020

Characterization of soft, deformable materials using conventional contact-based methods can be challenging due to errors associated with the applied pressure. Therefore, techniques which use non-contact methods to characterize these soft structures are highly desirable. The present work explores the development of a non-contact air-coupled resonant ultrasound spectroscopy (AC-RUS) technique for the characterization of soft materials. A sample of polydimethylsiloxane (PDMS) was mounted as a circular membrane clamped along its periphery and was excited first in the low frequency acoustic range (i<100 Hz) to induce the lower-order resonant vibration modes. The sample was then excited at ultrasonic frequencies (>50 kHz) to excite the higher order ultrasonic modes. A preliminary assumption of linear elasticity was used to analytically model the vibration of a circular membrane, followed by validations using numerical FEM simulations. The analytical and numerical models were then modified to include the effect of viscoelasticity using Kelvin-Voigt and Maxwell models. The resonant models were further used to obtain the elastic constants and viscoelastic parameter from the experimental frequency response curve.
Resonant ultrasound spectroscopy: Sensitivity analysis for anisotropic materials with hexagonal symmetry
Authors: Sevigney Christopher L. , Farzbod Farhad
Published: December 2020

Resonance ultrasound spectroscopy (RUS) is an experimental method for measuring the elastic properties of a material. A sample is excited to vibrate, and its resonant frequencies are measured. From the resonant frequencies and mode shapes, a complete set of elastic constants can be extracted. This result is not always reliable, however. In some cases, the resonant frequencies are insensitive to changes in certain elastic constants or their linear combinations. Previous work has been done to characterize these sensitivity issues in materials with isotropic and cubic symmetry. This work examines the sensitivity of elastic constant measurements by the RUS method for materials with hexagonal symmetry, such as titanium-diboride. We investigate the reliability of RUS data and explore supplemental measurements to obtain an accurate and complete set of elastic constants.
Single-sensor remote damage localization in vibrating structures using compressive sensing
Authors: Flynn Tyler J. , Dowling David R.
Published: December 2020

Many engineering structures are subject to broadband excitation while in operation, and it is common for such structures to radiate measurable sound as a result. When a structure is damaged the character of its frequency response changes and these changes can be exploited for health monitoring. In particular, shifts in resonant frequencies are appealing indicators of damage because they are relatively easy to obtain even with a single, non-contacting sensor. Previous methods have been developed to localize structural damage by minimizing an objective function of the differences between measured modal frequencies and those computed from an analytical model (e.g., a finite element model). However, these approaches suffer significantly when measured resonances are not known exactly, or when only few modes are obtainable. This talk details a compressive sensing (CS) approach to the damage localization problem that yields improved localization accuracy when only a small set of modal frequencies is available, even with imprecision. Simulated results are presented for both 1D (beam) and 2D (plate) systems using analytical finite element models, comparing the conventional and CS approaches. Effects of sensor noise and model inaccuracy are addressed. [Sponsored by NAVSEA through the NEEC and by the US DoD through an NDSEG Fellowship.]
Measuring the subjective cost of listening effort using a discounting task
Authors: McLaughlin Drew J. , Braver Todd , Peelle Jonathan
Published: December 2020

Objective measures of listening effort have been gaining prominence, as they provide metrics to quantify the difficulty of understanding speech. A key challenge has been to develop paradigms that enable the complementary measurement of subjective listening effort in a quantitatively precise manner. In the present study, we used a novel decision-making paradigm to examine age-related and individual differences in subjective effort during listening for spoken sentences in speech-shaped noise. On each trial subjects were offered the choice between completing an easier trial (presented at a +20 dB signal-to-noise ratio; SNR) for a smaller monetary reward, or a harder trial (presented at either +4, 0, −4, −8, or −12 dB SNR) for a greater monetary reward. By varying the amount of the reward offered for the easier option, the subjective value of performing effortful listening trials at each SNR could be assessed. Older adults discounted the value of effortful listening to a greater degree than young adults, opting to accept less money in order to avoid more difficult SNRs. Additionally, older adults with poorer hearing and smaller working memory capacities were more likely to choose easier trials; however, in younger adults, no relationship with hearing or working memory was found.
A Bayesian dynamic time warping approach for nonuniform Doppler paths in acoustic multipath channel
Authors: Yang Sijung , Choi Jae Won , Singer Andrew C.
Published: December 2020

The estimation of Doppler effects is indispensable to many acoustic signal processing applications including reliable underwater acoustic communication. Computationally expensive ambiguity-function based methods have good performance in estimating slowly varying uniform Doppler shifts. However, for highly reverberant environments such as in shallow water acoustics, there has not been a robust approach to estimate non-uniformly time varying Doppler along each of the different propagation paths. In this paper, we propose a multidimensional dynamic time warping (DTW) algorithm to precisely estimate non-uniform time varying Doppler in acoustic multipath channels. The proposed algorithm exploits a Bayesian approach to map multidimensional temporal scaling, while guaranteeing polynomial time complexity in the length of the signal. Both simulation and experimental results for signals received from moving sources in confined reverberant spaces will be provided to show the capability of the proposed approach.
Doppler tracking and compensation for underwater acoustic channels using shift-orthogonal pilot sequences
Authors: Bassam Ali , Seto Mae
Published: December 2020

A novel time-domain Doppler tracker and compensator, using shift-orthogonal OFDM pilot sequences, was developed, and assessed in simulations and early in-water trials. The objective is to address large Mach numbers like those experienced in communicating autonomous underwater vehicles (AUV). To start, the Doppler estimator extracts pilots in the received signal to produce Mach number estimates for every received signal sample. Given the large number of estimator samples, a Doppler tracker is proposed to reduce the number and simultaneously track Mach number variations. This novel tracker reduces the computational load on the compensator and makes it possible to implement on AUVs. Then, the Doppler compensator's first stage resamples the received signal using the Mach number estimates from the tracker's output. Since the estimates vary with time, the resampling performed is time-varying. Then, the final stage estimates the residual Doppler shift from the estimator and eliminates it with a phase rotation. The proposed tracker and compensator receiver outperform most existing compensators as measured through the mean-squared error, is more computationally efficient and addresses higher Mach numbers. Each element of the proposed receiver is tested in simulations and the results are shown to agree with theory. Comprehensive at-sea trials are next.
Inverse laplace transform and image-based room impulse response for information on source location
Authors: bhatta ambika , Gandhi Pratik , Denis Max
Published: December 2020

The source localization in indoor and outdoor using sensors based empirical model has been traditionally useful. The presented work is investigating the possibility of extending an exact “image-based” analytical room impulse response (RIR) to classify the source region. The approach is governed by its validity for a spherical acoustic simple source. One technique for a given spatial grid the time responses being mapped to an intensity profile for grayscale images will be processed by a tailored neural network, and other, the conventional processing of raw impulse responses to obtain a likelihood function. The study for both the outcomes will be of particular interest to determine location-driven features.
Bubble curtain effectiveness during impact pile driving for monopile installation at the Coastal Virginia Offshore Wind project
Authors: Amaral Jennifer L. , Frankel Adam S. , Miller James H. , Potty Gopu , Lin Ying T. , Newhall Arthur E. , Mason Tim
Published: December 2020

The Coastal Virginia Offshore Wind (CVOW) pilot project consists of two 6-megawatt wind turbines located 27 miles off the coast of Virginia Beach, VA. Monopile foundations with a diameter of 7.8 m at the seafloor were installed via impact pile driving on two separate days during May 2020. A double bubble curtain was used during the installation of one of the monopiles and no sound mitigation system was used during the installation of the second. The resulting acoustic field was measured during the impact pile driving using a suite of stationary and towed sensors to characterize the effectiveness of the bubble curtain in attenuating the sound levels at various ranges and azimuths. Metrics including the peak sound pressure level, sound exposure level, and kurtosis of each pile strike were determined and analyzed as a function of distance from the foundation. The frequency and azimuthal dependence of the bubble curtain effectiveness was also investigated. [Work supported by Bureau of Ocean Energy Management (BOEM.)]
Vertical sonar beam width of wild belugas (Delphinapterus leucas) in West Greenland
Authors: Zahn Marie J. , Koblitz Jens , Laidre Kristin L.
Published: December 2020

Echolocation signals of wild beluga whales (Delphinapterus leucas) were recorded in 2013 using a vertical, linear 16-hydrophone array at two locations in the pack ice of Baffin Bay, West Greenland. Individual whales were localized for 12 min of 1:04 h of recordings where on-axis clicks were isolated to calculate sonar parameters. We report the first sonar beam estimate of in situ recordings of belugas with an average 3-dB asymmetrical vertical beam width of 5.7 deg, showing a wider ventral axis. This is narrower than the commonly used estimates in the literature obtained from captive whales, suggesting beluga sonar beam width may be different in captive and wild contexts and is not necessarily symmetrical. Apparent source levels ranged from 210 to 220 dB pp re 1 μPa and whales were shown to vertically scan the array from 120 m distance. Our findings support the hypothesis that highly directional sonar beams and high source levels are an evolutionary adaptation for Arctic odontocetes to reduce noise and surface echoes from ice. These results provide the first baseline beluga sonar metrics from free-ranging animals using a hydrophone array and are important for acoustic programs throughout the Arctic, particularly for acoustic classification between belugas and narwhals (Monodon monoceros).
Nonlinear Lucassen waves with interface viscosity
Authors: Simon Blake E. , Hamilton Mark F.
Published: December 2020

Linear theory for quasi-longitudinal surface wave propagation along an elastic interface coupled to a viscous, incompressible liquid was first developed by Lucassen [Trans. Faraday Soc. 1968]. Lucassen waves are modeled with a fractional diffusion-wave equation in which the order of the fractional time derivative is 3/2. Nonlinearity in the elastic interface was taken into account recently by Kappler et al. [Phys. Rev. Fluids 2017]. Nonlinear Lucassen interface waves exhibit certain features associated with the mechanical disturbance that accompanies the electric action potential in the biological membranes of nerve axons, such as the “all-or-none” principle in which wave speed and pulse shape change dramatically above some amplitude threshold. While Lucassen waves are highly damped, for nonlinear propagation the attenuation described by the fractional time derivative provides insufficient energy loss near regions where shocks form in the waveform, resulting in the failure of conventional numerical algorithms such as Runge-Kutta schemes. Presented here is a modified model equation for nonlinear Lucassen waves that includes viscoelastic effects in the interface. The inclusion of viscosity in the interface results in greater losses at shock fronts and increased stability for numerical calculations. [B.E.S. is supported by the ARL:UT Chester M. McKinney Graduate Fellowship in Acoustics.]
Microphone length and its effect on vibration interference
Authors: Walsh Jonathan D. , Miles Ronald
Published: December 2020

One-dimensional duct models yield the deflection of an idealized massless diaphragm subjected to pressure and vibration. Additional ANSYS models also include the effects of a flexible, thin diaphragm. These models are used to predict the acceleration pressure that a microphone senses when subject to vibration, in terms of the effective mass added by the loading due to nearby air. This effective mass value is shown to be proportional to one half the length of the back volume, plus the length of the front volume. Previously measured data from electrets and MEMS microphones are shown to support this conclusion. The length-dependence of the vibration pressure suggests that a shorter microphone with a diaphragm placed as close as possible to the outside air will have minimal vibration pressure sensitivity. [This work is supported by a grant from the NIH National Institute on Deafness and other Communication Disorders (1R01DC017720-01).]
Determining low frequencies of the Japanese koto using a finite element model based on a computed tomography scan
Authors: Coaldrake A. K. , Dunlop Eric H.
Published: December 2020

The Japanese koto is a plucked 13-string wooden instrument noted for its complex resonances. A finite element model of the koto has been developed based on a computed tomography (CT) scan and has previously been reported. It can now be used as a tool to undertake more detailed examination of the koto's sound behavior. While first iterations of the model were found to predict frequencies above 100 Hz, predictions for frequencies below 100 Hz were less accurate. Physical studies however have shown many low frequencies do occur and are not artefacts. This paper discusses initial simulations with the model and more recent studies with Gaussian pulses that reveal low frequencies not previously obtained from the CT model. Their contribution to the characteristics of the sound behavior of the koto are discussed.
Richard H. Lyon and his contributions to noise control
Authors: Maling George C.
Published: December 2020

Richard H. Lyon and I had similar interest in modal structure, his in rooms and in structures, and mine in reverberation rooms designed for the determination of sound power. He always had a keen interest in the sound radiated by complex structures. This led to the idea that the sound quality of a product could be influenced by careful adjustment of the many sound sources in a complex machine. He made many contributions to sound quality over the years. He had many other technical interests too numerous to list here, but he was the “father” of statistical energy analysis for which he was greatly admired. Later in life, we were both elected to the National Academy of Engineering (NAE). One tradition of the NAE is to prepare memorial tributes to deceased members for publication in an annual volume. I was pleased to collaborate with a colleague, David Bowen of Acentech Incorporated, on the tribute which detailed many of his contributions to acoustics and noise control.
Assessments of vehicle interior noises—Considerations about the quality of listening test data
Authors: Fiebig Andre
Published: December 2020

As there is a need for quantifying the perceived quality of automotive sounds in order to define target sound characteristics and set-up development targets, diverse jury test designs are frequently applied to collect data. The jury tests range from structured listening tests performed under controlled laboratory conditions to qualitative methods collecting free verbalization data during test drives. Though, jury tests are usually only used as an intermediate step to derive sound quality metrics based on psychoacoustic quantities allowing the quantitative prediction of subjective responses to product sounds. However, the quality of jury test data and the way of processing the data drive the validity of the developed sound quality metric. Therefore, a detailed quality check of listening test data is of particular importance but often neglected in practice. The quality check involves the investigation of intra- and inter-rater reliability issues, the detection of outliers and clusters, the use of adjusted error measures, the (cross) validation of sound quality metrics for estimating the generalizability of results. The different aspects in investigating listening test data in the context of vehicle noises and their suitability for sound quality metric development will be discussed on the basis of selected examples.
From 2D to 3D: Exploring the validity of two-point correlation functions to study polycrystalline media
Authors: Alam Tanni , Arguelles Andrea P.
Published: December 2020

Acoustic attenuation in polycrystalline materials is generally formulated using statistical representations of the material in the form of two-point correlation functions. Common formulations assume decoupling of the spatial and tensorial components and invoke analytical forms to represent the spatial correlation function (W(r)). A widely accepted form of W(r) is an exponential expression that employs a single characteristic length to describe the microstructure. More recently, this approach has been extended to include grain size distributions and different grain morphologies. To date, experimental approaches have relied on using 2D micrographs of material sections to obtain the necessary material parameters. However, the accuracy of the 2D statistical parameters relative to the three-dimensional two-point statistics is yet to be validated. In this presentation, we propose an analytical approach to transform 2D image data to 3D volumetric data by extending Wicksell's corpuscle problem to the case of equiaxed grains in polycrystalline materials. We use digital microstructures to evaluate the validity of the transformation for a broad scope of grain size distribution parameters and evaluate the accuracy of the analytical forms of the spatial correlation function. Finally, we study the impact of these approximations on acoustic attenuation.
Investigation of a temporal modulation based method on the intelligibility of speech in speech-spectrum shaped noise
Authors: Soleymanpour Rahim , J. Brammer Anthony , Kim Insoo
Published: December 2020

Temporal modulation is frequently considered for improving the intelligibility of noisy speech owing to the importance of the envelope in conveying speech information. In this study, we evaluated the performance of temporal modulation on word scores obtained with speech-in-noise. We firstly divided the noisy speech into 16 contiguous subbands from 200 Hz to 6 kHz with bandwidth approximately 1.5xERB of an auditory filter. The temporal envelope was constructed for each subband from the absolute value. It was then low-pass filtered at 16 Hz and used as the instantaneous gain for the subband's noisy speech. We established a psychophysical test (Modified Rhyme Test) with different speech-in-noise SNR values of 0, −3, −6, and −9 dB. Eleven native speakers (age 29 ± 8) with normal hearing were recruited for the study. The signals were processed by MATLAB and presented over earphones to participants seated in an audiometric room. Comparing the processed and unprocessed noisy speech, the mean differences in word scores for SNRs of 0, −3, −6, and −9 dB were −0.4%, 1.5%, −10.5%, and 1.4%, respectively. Using ANOVA, we concluded that the temporal modulation-based algorithm does not produce a statistically significant improvement in speech intelligibility. [Work supported by NIOSH.]
Cochlear-implant simulation mimics increased age in temporal processing of syllables by young and middle-aged normal-hearing listeners
Authors: Rogers Catherine L. , Vallario Jenna , O'Malia Morgan , Donaldson Gail S.
Published: December 2020

To better understand the effects of signal quality on phonetic processing, the present study examines the effects of signal degradation on processing of two-syllable sequences by young and middle-aged normal-hearing listeners. Temporal processing of American-English vowel sequences was compared across unprocessed and cochlear-implant (CI) simulation conditions for young listeners and for a CI simulation condition for the middle-aged listeners. Using the method of Fogerty, Humes and Kewley-Port [J. Acoust. Soc. Am. 127, 2509–2520 (2010)], listeners heard 70-ms, resynthesized versions of four syllables (“pit, pet, put, pot”) in a two-syllable temporal-order processing task. Task difficulty was adjusted by increasing or decreasing syllable-onset asynchrony (SOA), i.e., the duration between syllable onsets. SOA thresholds for accuracy of syllable-sequence identification was estimated using the method of constant stimuli on each of four 72-trial blocks. Results for the CI simulation conditions for both listener groups were comparable to those for the next older age-group in Fogerty et al. (2010), i.e., young adult listeners performed similarly to the middle-aged listeners Fogerty et al., and middle-aged listeners performed similarly to the older listeners Fogerty et al. Results will be discussed with regard to implications for phonetic processing of speech in demanding listening conditions and practical implications for CI users.
Bilinguals' comprehension of foreign-accented speech
Authors: Carraturo Sita , Van Engen Kristin J.
Published: December 2020

Foreign accents represent a common challenge to successful speech recognition, and while much has been uncovered about the factors contributing to the intelligibility of foreign-accented speech, most of it is based on data from monolingual listeners. There is good reason to believe, however, that bilinguals have a different experience when decoding foreign-accented speech: their two phonologies offer both greater flexibility in phonological-lexical mapping (Samuel and Larraza, 2015) but also greater lexical competition (Marian & Spivey, 2003). Further, they are known to perform more poorly than monolinguals in speech-in-noise comprehension (Rogers et al., 2006). Given these consequences of bilingualism, the current study compares the intelligibility of foreign-accented speech for monolingual listeners and simultaneous bilingual listeners. Spanish-English bilinguals and English monolinguals performed a sentence repetition task for sentences produced by speakers of Mandarin-Chinese accented English with varying levels of proficiency in English. The competing hypotheses for this study are as follows: (1) because bilinguals' have a more flexible phonological-lexical mapping system, they will outperform the monolinguals in foreign-accented speech recognition; or, (2) greater lexical competition due to the bilinguals' L2 lexicon will result in poorer performance. The results will provide insight on bilingual sentence processing, and shed light on the shared and unique experiences of bilinguals and monolinguals.
Processing mixed talkers in noise suggests two mechanisms for perceptual adaption to speech
Authors: Saupe Maya , Best Virginia , Lim Sung-Joo , Choi Ja Young , Perrachione Tyler K.
Published: December 2020

Perceptual adaptation to a talker allows listeners to efficiently resolve inherent ambiguities present in the speech signal introduced by the lack of a one-to-one mapping between acoustic signals and intended phonemic categories across talkers. In ideal listening environments, preceding speech context enhances perceptual adaptation to a talker. However, little is known regarding how perceptual adaptation to speech occurs in more realistic listening environments with background noise. Here, we explored how talker variability and preceding speech context affect identification of phonetically confusable words in adverse listening conditions. With dependent variables of response time and threshold signal to noise ratio (SNR), our results showed that listeners were less accurate and slower in identifying mixed-talker speech compared to single-talker speech when target words were presented in multi-talker babble, and that preceding speech context enhanced word identification performance under noise both in single- and mixed-talker conditions. These results extend previous findings of perceptual adaptation to speech in quiet environments and suggest that two distinct mechanisms underlie perceptual adaptation to speech: rapid successful feedforward allocation of attention to salient talker-specific stimuli via auditory streaming, and an additional mechanism that preallocates cognitive resources to support processing of talker variability over longer time scales.
Automatic classification of kinematic flap variants using ultrasound and optical flow
Authors: Faytak Matthew , Mayer Connor , Kuo Jennifer , Teixeira G. , Zhou Z. L.
Published: December 2020

Study of the four covert kinematic variants of North American English flaps can provide insight into questions in speech motor control and articulation-acoustics relations [Derrick and Gick, Can. J. Linguist. 56(3), 307–319 (2011)]. These variants are typically labeled by human annotators from ultrasound video, which is time-consuming and labor-intensive. In this study, we present an automatic classification method, taking as basis data optical flow fields over a series of ultrasound frames flanking the flap; fields are calculated using a method tailored to ultrasound video [Moisik et al., JIPA 44(1), 21–58 (2014)]. Two classifiers are compared: support vector machines, which learn an optimal linear separator between labeled class instances; and simple recurrent neural networks [Elman, Cognit. Sci. 14(2), 179–211], which operate recursively over sequences of data, and whose state at any timepoint is calculated based only on the state at the preceding timepoint and the current input field. We train these classifiers on human-labeled flap tokens and test classification performance against a held-out subset of the labeled tokens. We go on to discuss the general applicability of this method for disambiguating covertly different lingual articulatory events.
Ejective production mechanisms in English
Authors: Price Lavinia , Pouplier Marianne , Hoole Philip
Published: December 2020

Ejective realization of stops seems to be increasing in many varieties of English, which has triggered a new debate on the basic mechanism of ejective production. While traditionally the glottalic initiation mechanism of ejectives is thought to require a closed glottis and active larynx raising, this has been questioned for the emergent English ejectives, and a pulmonic airstream mechanism has been suggested. In order to shed light on the possible production mechanism of English ejectives, we present rtMRI data acquired at 50fps from a corpus of 27 native British English and 18 native American English speakers. Auditory analysis of 20 BE-speakers from the corpus confirms that 26% of all word-final stops in target words are realized as ejectives. Our results so far agree with previous observations about the conditioning factors favoring ejective production. Ejectives occur predominantly in stressed position as allophones of word-final velar, less frequently of labial and coronal stops. Surprisingly, also phonemically voiced stops may be realized as ejectives. Importantly, the rtMRI data will allow us to measure larynx height and to observe the extent of active larynx movement involved in the initiation process in English ejectives. [Work supported by ERC Horizon 2020 grant Interaccent to J. Harrington.]
Childhood language exposure: Does early experience affect sound perception and production in speakers with reduced language proficiency?
Authors: Hanini Rawan , Alkhudidi Anwar , Rafat Yasman
Published: December 2020

Early language exposure is crucial for acquiring native mastery of the phonology of a language (Fromkin 1974, Flege 1987, Antoniou et al. 2015). It is not clear, however, whether early language exposure has lasting benefits when the quantity and quality of speaking drop dramatically after childhood (Oh et al. 2003). In this study, we investigate the production of Arabic speech sounds in 20 native speakers (who heard and spoke Arabic during childhood regularly), 20 childhood speakers (who heard Arabic regularly during childhood but did not speak it regularly afterwards), and 20 novice speakers (did not have any exposure to Arabic during childhood but are currently enrolled in Arabic language classes). The experiment includes a language proficiency test and a phoneme production and perception task. The target sounds are geminate consonants. The acoustic analysis consists of manual alignment of each consonant, and extracting their duration and acoustic information (voicing, aspiration, and formant values). The findings shed more light on whether early language experience has measurable long-term benefits for an individual's phonetic and phonological skill even if the language experience diminishes over time. In addition, we gain further insight into the role played by universal markedness factors in L2 acquisition (Davidson, 2011).
Phonetic interaction between late Japanese-English bilinguals' L1 and L2 vowels—A longitudinal study
Authors: Takahashi Chikako
Published: December 2020

We investigated late bilinguals' (N = 31) change in their production and perception of the L1 Japanese vowel /i/ and its phonetically associated L2 vowel /i/ and /I/ over one year. The perceptual boundaries between both English /i/-/I/ and Japanese /i/-/e/ were examined through identification tasks. Participants' production of the high front vowels was also analyzed. More than half of the participants exhibited evidence of separating English and Japanese high front vowels from each other in production. Participants who seemed to use similar F1-F2 values for these vowels still tended to separate them. The perception task results showed a perceptual boundary shift in bilinguals' L1 Japanese after one year. That is, they identified more tokens as /i/ than /e/ in /i/-/e/ continuum. This phenomenon was also observed when comparing bilinguals to Japanese monolingual controls. We argue that this change in L1 perception stems from learning a new L2 phonetic category /I/ and associating it with a similar L1 /i/. Individual differences in bilinguals' relative language use and exposure were also examined as potential factors driving changes in perception and production of Japanese/English high front vowels.
Voice onset time in monolingual and code switching modes in Lebanese Arabic and English
Authors: Kelly Niamh , El Houry Mia , Ghamloush Farah
Published: December 2020

The effect of speaking mode and language in speakers of Lebanese Arabic and English was examined. Voice onset time (VOT) was measured for voiced and voiceless stops in both languages, when speakers were in monolingual mode or code-switching mode. Nine speakers who were highly proficient in English, and used to code-switching, produced 534 tokens in an image-labelling experiment (Olson, 2013; Mayr et al., 2018). Research has found that speakers may produce the same sound differently depending on whether they are in monolingual or bilingual mode (e.g., Olson, 2013; Simonet, 2014; Amengual, 2018). Linear regressions found no significant difference in VOT between conditions (monolingual versus code-switching) or between languages (Arabic versus English). The only significant predictor of VOT was voicing of the stops. Average VOT for voiced and voiceless stops in monolingual Arabic: -59msec & 44msec, monolingual English: -55 ms and 48 ms, code-switched Arabic: -63 ms and 45 ms, and code-switched English: -56 ms and 47 ms. These results indicate voicing during the closure for voiced stops and some aspiration for voiceless stops, and show that speakers do not have separate stop categories for the two languages. Current research is examining monolingual Arabic speakers to determine whether the bilingual speakers have L2 transfer from English aspiration.
Tone differentiation as a means for assessing non-native imitation of Thai tones by Mandarin speakers
Authors: Chen Juqiang , Best Catherine T. , Antoniou Mark
Published: December 2020

Non-native tone production and imitation have been found to be phonetically deviant from native production for some discrete measures. However, it remains unresolved whether non-native imitation differs from native production in terms of the differentiation of tones in acoustic tone space. 32 native Mandarin speakers who had no experience with Thai imitated five Thai tones, and each participant produced 160 tokens in total under differing memory load and stimulus variability conditions to determine effects of cognitive demands. We calculated two tone differentiation indices (i.e., Index 1: tone differentiation within the tonal space; Index 2: differentiation among tones, both as in Barry & Blamey, 2004) based on F0 onset and F0 offset for Thai tones and the non-native imitations of these Thai tones by Mandarin imitators. There was a significant memory load by vowel variability interaction for Index 1 and a main effect of talker variability and a three-way interaction (memory load “talker variability” vowel variability) for Index 2, suggesting that tone differentiation is affected by cognitive factors. Nonetheless, non-native tone imitations were not significantly different from native productions on either index, indicating that non-native imitation resembles native production in terms of tone differentiation in an onset-offset F0 space.
Alternative representations and object classification of circular synthetic aperture in-air acoustic data
Authors: Park J. Daniel , Blanford Thomas E. , Brown Daniel C. , Plotnick Daniel
Published: December 2020

Synthetic aperture sonar imagery is typically generated using data collected with unmanned underwater vehicles. The prohibitive cost of collecting underwater data and the need for well-controlled factors such as collection geometry and object configuration has provided the motivation for devising a benchtop in-air circular acoustic data collection framework. This set-up makes it practically feasible to explore a multitude of parameters that are not as feasible with underwater measurement scenarios, including waveform type, object shapes and material. It is also practically feasible to explore various representations of the collected acoustic data that help better emphasize different aspects of the information embedded in the acoustic signal, which various machine learning algorithms can utilize. Signal processing and feature organization are critical to improving performance of machine learning algorithms. For example, geometric scattering response of objects is well-represented in spatial imagery with sharp contrast of pixel intensity between the object and surrounding environment, while spatial spectrum of the complex SAS image better represents the aspect-dependent spectral response of the object that help discriminate objects of the same shape, but with different material. We will discuss the relationship between the choice of representation and discriminatory information with illustrative classification problems.