bluuebunny/crossref_metadata_2025 huggingface dataset:
An investigation of the effect of randomness in trunk size and location on the attenuation of sound through forests generated by Lindenmayer systems
Authors: Sahoo Divyamaan
Published: July 2024

This presentation provides an overview of what is known about the influence of randomness in trunk size and location on the attenuation of sound through forests. It also provides an explanation of the Lindenmayer system (L-system) approach to modeling plants and trees in addition to the multiset L-system approach to modeling the dynamics of forest ecosystems, investigating its applicability to the problem of optimizing sound attenuation through forests. This project was completed in partial fulfillment of the requirements for the Diploma in Acoustics and Noise Control, Institute of Acoustics, Milton Keynes, UK, under the mentorship of Dr. Keith Attenborough.
Hypercompression in a tapered, viscous, nonlinear cochlear model
Authors: Agarwal Vipin , Cai Wen , Grosh Karl
Published: July 2024

In our current research, we focus on predicting the stationary nonlinear response of a cochlear model that extends from the base to the apex when subjected to harmonic input, taking into account the tapering of the cochlear scalae along with electromechanical coupling of outer hair cells and the microstructures of the organ of Corti. We are interested in explaining the paradoxical phenomenon of hypercompression in the motion of the reticular lamina (RL), whereby the response decreases with an increase in the acoustic excitation at the stapes. We derive frequency–response curves for both the basilar membrane and RL at various locations, exploring a wide range of excitation frequencies and amplitudes. In accordance with experimental data, we find that the RL exhibits hypercompression at the base of the cochlea. This behavior is found to arise from the interaction of saturating nonlinearity arising from the active process and the linear response identified as the passive response to acoustic stimulus. These two responses are not synchronized in phase. As the excitation level increases, the two effects tend to partially cancel, giving rise to lower responses with increasing sound pressure levels. [Work supported by NIH-NIDCD R01 04084.]
Does hearing impairment affect gait under realistic conditions?
Authors: Wesson Troy , Haddad Jeffrey , Ambike Satyajit , Foley Abigail , Patel Radha , Burgin Sarah , Francis Alexander L.
Published: July 2024

Hearing loss is associated with increased risk of falling, but little is known about how hearing impairment might affect mobility. Here, we report preliminary results from an ongoing experimental study investigating dynamic properties of gait while walking with and without impaired hearing. Young- and middle-aged adult participants with self-reported normal hearing completed four walking tasks, 2 indoors, 2 outdoors, each conducted with and without a simulated hearing loss. In the impaired condition, participants wore an insert earplug in one ear combined with binaural, circumaural noise-damping headphones. We quantified gait parameters using data from inertial measurement units (IMUs) affixed to participants’ ankles and waist, allowing participants to walk farther than in typical gait assessments. Parameters included standard gait metrics, such as limb acceleration as well as a novel spatiotemporal index quantifying variability in step patterns. Data have been collected from 13 of 30 projected participants. Analysis will focus on how temporal-spatial gait parameters and variability differ between indoor and outdoor walking with and without hearing impairment. We expect gait parameters to reflect a more variable and potentially less mobile walking strategy in the outdoor condition and with impaired hearing due to increased cognitive load and/or reduced spatial awareness in those conditions.
Do individual differences correlate across speech perception tasks?
Authors: Zhai Weiyi , Clayards Meghan
Published: July 2024

Speech perception requires listeners to take into account acoustic cues as well as lexical context and phonetic (coarticulatory) context. Individuals have been shown to vary in how they integrate these factors. To better understand the sources of these differences, we conducted three phoneme categorization tasks on speech continua with 82 native Canadian English speakers. Task 1 (lexical + coartic) embedded a /s-ʃ/ continuum in lexically biasing contexts (e.g., a(s)ume, a(ʃ)ure) followed by different coarticulatory contexts (rounded or unrounded vowels). Task 2 (lexical) had only lexical context cues for /ɛ/-/ɪ/ vowel continua (e.g., v(ɛ)st, k(ɪ)t). In task 3 (coartic), a /d/-/g/ stop continuum in nonsense syllables followed different coarticulatory contexts (/ar/ or /al/). We found those who used lexical context more used coarticulatory context less in task 1, consistent with prior research. However, this correlation disappears when examined across tasks 2 and 3. We also found no correlation between individual use of lexical and coarticulatory context across tasks, suggesting task dependency. Participants’ use of acoustic continua was positively correlated across tasks, indicating an individual trait for utilizing acoustic cues.
Novel auditory alerts that foster efficient detection and discrimination in complex auditory environments: Dual-task conditions
Authors: Cummins Mabel L. , Schutz Michael , Bernstein Leslie R. , Shive Joshua , Schlesinger Joseph J.
Published: July 2024

In many complex auditory environments, auditory alerts must be perceived and distinguished accurately while distractions are mitigated. Previously, we measured detection and discrimination performance using four novel alerts: one pair of narrowband and one pair of broadband alerts. Within each pair, one alert was perceived as consonant (labeled “friendly”) and one as dissonant (labeled “enemy”). The alerts were presented along with maskers consisting of “truck noise” or “truck noise” combined with “speech babble.” Separately, for the pairs of alerts, detectability of the alerts and discriminability between “friendly” and “enemy” pairs of alerts were measured as a function of signal-to-noise ratio (S/N). Results indicated that the alerts allow for robust detection and discrimination, even though the “friendly”-“enemy” pairs of alerts occupied similar spectral loci. In the study reported here, we measured the detectability and discriminability of the alerts under single and dual-task “N-back conditions” to simulate more attentionally demanding environments (e.g., in military conflict). The goal was to assess the impact of the N-back task on the robustness of the alerts to convey crucial information. Results will be presented and discussed in terms of the influence of the relevant variables on the form of the dual-task ROC. [Work funded by ONR N000142212184.]
Predictive factors for broad binaural pitch fusion in adults with hearing aids and cochlear implants
Authors: Reiss Lina A. , Sanders Holden D. , Dean Nicole
Published: July 2024

Many adults with hearing loss who use hearing aids (HAs) and/or cochlear implants (CIs) have broad binaural pitch fusion, such that sounds with large pitch differences are fused across ears, leading to difficulties separating voices in multi-talker environments. However, there is individual variation. The goal of the current study was to investigate whether auditory experience factors explain this variation in broad binaural fusion. Binaural fusion was measured in adults with various hearing device combinations: 26 bilateral HA, 25 bimodal CI, and 27 bilateral CI users. Fusion ranges were measured by simultaneous, dichotic presentation of reference and comparison stimuli in opposite ears, and varying the comparison stimulus to find the range that fused with the reference stimulus. Factors examined included age at testing, degree of HL, amount of amplification, age of onset of HL, duration of HL without CI, and duration of HA, bimodal CI, and bilateral CI use. Preliminary analyses suggest that broad fusion is correlated with long durations of HA use; further analyses using multivariable models will be described. The findings will indicate how hearing device experience may influence binaural pitch fusion. [Work supported by NIH grant R01 DC013307.]
Improving speech understanding for face-to-face communication in noise when wearing hearing protectors
Authors: Brammer Anthony J. , Soleymanpour Rahim , Golzari Kia , Heiney Erin , Marquis Hillary , Kim Insoo
Published: July 2024

Active hearing protection devices (eHPDs) are commercially available that enhance communication at low noise levels by amplifying both speech and noise. A study has been performed to develop algorithms for improving face-to-face speech communication in both low- and high-level noises when wearing eHPDs. The (mixed) environmental noise and speech at frequencies from 200 to 6000 Hz are divided into 16 or 24 contiguous subbands corresponding to 1.5 and 1 times the bandwidth of auditory filters, respectively. Signals are processed in the time domain with the overall group delay intended to maintain synchronization between speech and lip movements. The temporal modulation in each subband is band-limited to 2–16 Hz using zero time delay, moving detrend, high-pass, and moving average low-pass filters. This modulation is compared to the normalized unfiltered modulation in the subband to provide an estimate of the speech signal-to-noise ratio (eSNR). A threshold is selected for the eSNR above which time signals in subbands are combined to produce the instantaneous output and below which subband signals are attenuated. Subjects “wearing” a simulated HPD and listening to speech in industrial noise experience, on average, a 12% increase in intelligibility when using the algorithm. [Work supported by NIOSH and the Alpha Foundation.]
Eric Ungar: Austria to Acentech—Highlights, contributions, reflections, and stories
Authors: Zapfe Jeffrey A.
Published: July 2024

This paper will describe Eric Ungar's journey from boyhood in 1920's Austria to his stature as one of the most significant contributors to the field of acoustics and vibration. I will go over some of Eric's highlights, major awards, and contributions. I will also describe how Eric has influenced me both before and after our time together at Acentech. And what discussion of Eric's career would be complete without some of the stories and anecdotes that show off the not so technical side of Eric as well.
Eric Ungar and vibration damping
Authors: McDaniel James G.
Published: July 2024

This presentation honors Eric Ungar’s contributions to vibration damping. The classic paper titled “Loss Factors of Viscoelastic Systems in Terms of Energy Concepts,” which he co-authored with Edward Kerwin in 1962, takes center stage. Two significant contributions of the paper deserve detailed discussion. The first is the time dependence of total energy for viscoelastic systems with added mass, which results in ambiguous loss factor definitions off resonance. The second and most important contribution is an equation for the loss factor of a network of ideal viscoelastic springs. This equation reveals that the system loss factor is a weighted average of the loss factors of the viscoelastic springs, where the weights are the total energies in the viscoelastic springs. Next, the value of this work is illustrated by some recent work inspired by the 1962 paper. This recent work, which was published 60 years after the 1962 paper, involves the calculation of loss factors from finite element models of complex structures with viscoelastic elements. The presentation concludes with recollections that celebrate Eric Ungar’s talent in educating and inspiring generations of engineers and researchers. [Work supported by ONR under Grant N00014-22-1-2785.]
How will climate change affect the underwater sound environment? Modelling the potential effects of climate change on the oceanic speed of sound
Authors: Farkas Abigail
Published: July 2024

The purpose of this study is to create a year-by-year prediction model for surface-level oceanic speed of sound in different future climate scenarios and discuss how these results may be relevant for current and future work in underwater acoustics. It is already understood that sound propagation and speed in the oceans will be affected by future changes of oceanic pressure, temperature, salinity, and acidity. Certain trends of future sound speed have been analyzed in precedent studies; however, there are some information gaps and the availability of future oceanic sound speed models for use by the acoustical industry is lacking. This study intends to fill in some of these gaps and initiate an interdisciplinary discussion between both research and industry professionals in the field of acoustics.
SONAR target classification with complex-valued neural networks
Authors: Linhardt Timothy , Sen Gupta Ananya , Bays Matthew , Kirsteins Ivars P.
Published: July 2024

Popular acoustic signal processing techniques analyze acoustic color, which is a magnitude representation of a frequency spectrum. This paradigm allows for easy visualization of results and simpler models at the cost of throwing out phase information. Analysis and modeling of complex-valued data does have inherent difficulty from the nature of complex numbers. Optimization on the complex field requires alternate partial derivative definitions to circumvent consequences of the Cauchy–Riemann equations regarding holomorphic functions. To make use of phase information, we demonstrate classifier model optimization with complex-valued parameters on data with both magnitude and phase components and show how complex neural networks yield marked improvements over similarly shaped real-valued networks in both classification accuracy and generalization ability. We apply these techniques to small target SONAR with simulated Lamb wave resonance signals for hollow spheres, differentiating between different material classes. [Research funded by the DoD Navy (NEEC) Grant No. N001742010016.]
Sound transmission loss through steel-framed building partitions
Authors: Raley Michael , Shafer Benjamin M.
Published: July 2024

Load-bearing steel-stud walls pose a unique acoustical challenge because both the spacing and the mil thickness of the studs can significantly affect the acoustical performance of these walls. A single wall type in a load-bearing steel building may actually include many acoustically distinct walls as the stud spacing and mil thickness change from floor to floor in the building due to the different loads. This can be especially challenging on delegated design projects where the final selection of the steel studs is left to the framing contractor and may not be included in the project documents supplied to the acoustical consultant, if there even is one on the project. In prior ASA presentations, Ben Shafer has clearly documented the effects of steel stud spacing and mil thickness on acoustical performance. This presentation adds to Ben’s prior work by investigating the effects of resilient sound isolation clips on wall performance. Clips are a unique application because the spacing of both the clips, and the hat channels does not change with changing stud spacing and because the low stiffness of the rubber isolators should reduce the effect of stud stiffness. This presentation will present the results of recent lab tests of walls with resilient sound isolation clips and compare the results to the data previously presented by Mr. Shafer.
Ultrasound imaging of mammalian cells using drug-induced acoustic reporter genes
Authors: Kim John , Welch Phoebe J. , Howells Alessandro R. , Lian Xiaojun L. , Shi Chengzhi
Published: July 2024

The emerging area of genetically engineered ultrasound contrast agents, like gas vesicles, has the potential to broaden the applications of medical ultrasound imaging by enabling targeted and deep tissue imaging at the cellular level. Nevertheless, the existing gene construct/encoding relies on significant cell processing to ensure sufficient gas vesicles are formed within the cell to produce ultrasound contrast. Here, we describe a drug-inducible and drug-selectable acoustic reporter gene construct that can enable gas vesicle expression in mammalian cell lines, which we demonstrate in wild type HEK293T cells. Fluorescence microscopy was employed to validate the integration of the plasmid, while the creation of single-cell clones was achieved through the utilization of flow cytometry. The expression for gas vesicle was optically and ultrasonically verified, achieving 80% improved signal to noise ratio in cells expressing gas vesicles compared to negative controls. This technology introduces a novel paradigm for reporter genes, utilizing ultrasound to visually identify particular cell types both in vitro and in vivo, serving diverse purposes such as cellular reporting and applications in cell therapies.
A study of rocket launch directivity and the potential correlation of various parameters
Authors: Yancey Matthew G. , Jackson Ian R. , Hart Grant W. , Mathews Logan T. , Gee Kent L.
Published: July 2024

A rocket launch produces sound that is highly directional. This radiation is presumed to be due to Mach wave radiation, but the production mechanism is poorly understood. Previous research has shown that launch vehicle sound directivity peaks at an angle near 65° relative to the plume for most large rockets during early stages of the launch, yet limited attention has been given to the physical parameters involved in this sound production. This study explores the influence of some additional parameters that are not usually considered, such as ambient atmospheric pressure effects and the launch vehicle's Mach number on a rocket's directivity. Drawing parallels with similar studies on military jets, we aim to compare results and further our understanding of rocket acoustics, contributing to improved modeling precision.
Generalized least squares beamforming of infrasound data
Authors: Bishop Jordan W. , Blom Philip S. , Webster Jeremy
Published: July 2024

Generalized least squares (GLS) beamforming is a method for determining the direction of arrival and trace velocity of transient infrasound signals that may be otherwise obscured by persistent, correlated background noise, such as microbaroms. This method complements the adaptive F-detector by using an estimate of the noise background to form a generalized power ratio, which is used to estimate plane wave parameters (trace velocity and back-azimuth). Using a suite of fully synthetic signals, we first investigate the resolving power of the GLS estimator as a function of signal to noise ratio compared with a conventional, non-adaptive estimator. Fstatistic and plane wave parameter estimates will then be compared between GLS beamforming, conventional Bartlett beamforming, Capon beamforming, and the MUSIC algorithm. Recorded infrasonic signals from the Forensics Surface Experiment, where a persistent signal was observed from the south, will then be used to evaluate the GLS method. Initial analyses suggest that GLS beamforming results in a lower F-statistic during noise regions and a higher F-statistic value for transient signals compared to the Bartlett beam. Algorithmically determining an optimal window to characterize the background noise presents a significant challenge and different approaches will be discussed.
What else can we do with auxiliary parameters in ray tracing? How about back projection for localization?
Authors: Blom Philip S.
Published: July 2024

Auxiliary parameters describing variations in ray path geometry with respect to initial launch angles have been leveraged in computing the Jacobian determinant needed to solve the transport equation as well as to build a Levenberg–Marquardt algorithm for identification of source-receiver paths (termed eigenrays) in a 3D inhomogeneous moving atmosphere. Building on these applications, recent investigations have demonstrated that these parameters can be computed along back projected ray paths from infrasonic detections to improve Bayesian localization capabilities. An overview of the auxiliary parameters as introduced in previous work will be provided along with a summary of current localization development. Example applications of the method will be presented and compared with existing Bayesian infrasonic localization methods using more general propagation models.
Transforming discrete movements of a sonar array into smooth vibration of a plate: Constant force input versus constant displacement input
Authors: Anderson Brian E. , Hughes W. Jack , Hambric Stephen
Published: July 2024

Numerical modeling of systems is an efficient way to explore new ideas, but an understanding of modeling assumptions as they relate to the actual physics of the system is critical. As a graduate student, I inherited a potentially revolutionary project involving sonar arrays. The idea was that grating lobes in the sound radiation from arrays could be eliminated by placing a plate between the array and the water. Grating lobes result from unintended, aliased superposition of traveling waves when discrete transducers attempt to create a single traveling wave in the array’s plane. The model initially assumed the transducers provided a constant displacement input to the plate’s vibration and the plate, thus, transformed the discrete traveling wave into a smooth continuous wave, resulting in elimination of grating lobes. The problem was that the plate’s impedance was higher than the transducers’ impedance and, thus, a constant force input was a better approximation to use in the model. When accurate assumptions were used, the grating lobes returned. A new idea was conceived to make something useful out of the original idea, though it required yet undeveloped materials. Although unexpected and disappointing at the time, “negative discoveries” can sometimes lead to positive ones.
The phonetically balanced kindergarten assessment: An alternative evaluation method utilizing machine learning technology
Authors: Cauchi Ayden M. , Negandhi Jaina , Cornacchia Micheal , Gordon Karen A.
Published: July 2024

This study investigates the use case of machine learning (ML) technologies to score the Phonetically Balanced Kindergarten (PBK) word test. The PBK, developed to test pediatric speech perception, is also used to monitor outcomes of cochlear implantation (CI) in children. Many factors contribute to high PBK score variability in CI users, but effects of scoring by human listeners are unclear. In this study, an alternate ML scoring method was developed. The Ursa classifier (Speechmatics, 2023) was used to recognize 100 PBK words spoken by 12 adults with normal hearing, speech, and language. The spoken words (n = 1200) were manipulated in six conditions (first and last phoneme deletion; high frequency filtering at 1, 2, 4, and 6 kHz), yielding 8400 stimuli. The classifier correctly recognized most of the unaltered words (true positive rate of 0.90; peak distribution density at 96%) and provided a null or incorrect response to altered stimuli as expected; most frequently for the first phoneme deletion and 1 kHz filtered conditions. Inter-class correlations showed comparable results between Ursa and 7 normal hearing human scorers (Cohens Kappa: 0.39). Thus, ML tools show promise in the scoring of the PBK; increasing the accuracy, efficiency, and reliability of this clinical assessment.
Real-time word recognition in heritage speakers: Evidence from the visual world paradigm
Authors: Kutlu Ethan , Boudreau Jacob , Chiu Samantha , Klein-Packard Jamie , McMurray Bob
Published: July 2024

Up to 20% of children in the US come from multilingual households (Census, 2019). However, current language development models are still mostly based on monolingual input. Heritage speaker (HS) children pose a challenge for current understandings of monolingual language development because their first language (the heritage language, HL) becomes their less dominant language as they age (often when schooling begins), and their second language (the majority language, ML) eventually becomes dominant, despite a delayed start. Here, lexical processing—a key hub in the language system—is targeted as a model system in which to ask how HSs become more automatic language processors in the ML. To do so, the proposed study focuses on spoken word recognition, which is a critical bottleneck in early language development, as the ability to efficiently recognize continuous speech paves the way for higher level language abilities. Successful recognition requires listeners to suppress competitors, such that only the target word remains. We present eye-tracking data from school-aged HSs who have become more dominant in English (n = 30). Our findings show that HSs, despite a delayed start in English, reach target words efficiently. We discuss the strategies employed by HS children to resolve competition in their ML.
Heritage speaker children’s speech categorization patterns
Authors: Kutlu Ethan , Theuerkauf Nick , Chiu Samantha , Klein-Packard Jamie , McMurray Bob
Published: July 2024

Bi-/multilingual experiences vary widely due to differing degrees of input frequency, dominancy in languages, and societal preferences for the language(s) used. Heritage bilingualism has received notable attention over the last decade. Heritage bilinguals are those who often use or are exposed to one language at home (the heritage language) that differs from that used outside of the home (the majority language). For a long time, bilingual children’s unbalanced input was targeted as a source of deficiency in categorizing speech. However, prior studies used tasks that were not equipped to address variability in speech categorization across listeners (see Apfelbaum et al., 2022). We use the visual analogue scaling (VAS) task which can capture variability in speech categorization across listeners with more sensitivity and which can reveal new dimensions of differences such as the trial-by-trial variability in responding (Kapnoula et al., 2017; Kutlu et al., 2022). In an ongoing study, we are testing school-aged bilingual children with various heritage language backgrounds (n = 30, 6–11 years). Preliminary findings suggest that heritage speaker children are not deficient in speech categorization, and in fact, show more gradient categorization patterns, suggesting a functional adaptation to increased phonetic variation in their language environment (Kutlu et al., 2022).
Listening effort mitigates rollover effects on speech-in-noise perception
Authors: Huang Chengjie G. , Field Natalie A. , Latorre Marie-Elise , Farrar Rebecca M. , Anderson Samira , Goupell Matthew J.
Published: July 2024

Increasing the sound intensity may lead to worse speech understanding, especially in noise. This is known as the “Rollover” phenomenon. There is mounting evidence that listening effort plays an important role in challenging listening conditions and can be directly quantified with objective measures such as pupil dilation. However, there is limited understanding of how listening effort relates to rollover in speech understanding. We hypothesized that listening effort plays an essential role in mitigating rollover effects to differential extents across age and hearing status. We recruited across the adult lifespan (N = 50, 20–83 years) with different hearing statuses in acoustic listeners and cochlear implant users to perform a speech discrimination task. Minimal word pairs were presented both in quiet and in 0 dB SNR babble noise, ranging from 35–85 dB SPL. Pupil area was tracked simultaneously with behavioral responses during the task. We found that normal-hearing listeners are fully able to utilize effort contributions to minimize rollover effects between in quiet and in noise conditions, with diminishing benefit as a function of age and increased hearing loss. The results of this project could broadly influence how to design future hearing devices and interventions that maximize hearing abilities for those affected by hearing loss.
Exploring auditory recognition: Native Japanese listeners' perception of frequent and infrequent English words in noise with varied phonological neighborhood densities
Authors: Nozawa Takeshi , Wayland Ratree
Published: July 2024

Native Japanese listeners identified English words embedded in noise (SNR + 6dB) produced by native American English and Japanese speakers. The identical word list used in our previous study (Nozawa and Wayland 2023) was employed. Listeners responded by typing the words they heard. The findings revealed no discernible impacts of phonological neighborhood density, possibly due to the small vocabulary size limiting competition from other words. However, frequent words exhibited greater precision in recognition compared to infrequent words. Despite the introduction of noise, the overall accuracy of responses did not experience a substantial impact. Nevertheless, specific words like “class” and “club,” initially nearly perfect in accuracy without noise, dropped to 0% accuracy. Most errors included phoneme substitution rather than phoneme addition or deletion, and in / CVC/ words, the onset consonant was identified substantially better than the vowel and the coda consonant. Initial consonant clusters, such as /br/, / kl/, and /pl/, were frequently perceived as /l/ or /r/, leading to the initial stops or fricatives being missed. Errors encompassed the substitution of phonemes with others that are not typically perceptually assimilated into the same native language (L1) categories, which would not be revealed through predefined multiple-choice identification or discrimination tasks.
The effect of talker intelligibility on adaptation to an unfamiliar accent
Authors: Matthews Kaitlyn L. , Van Engen Kristin J.
Published: July 2024

Listening effort, as indexed by task-evoked pupil response, increases as the intelligibility of second language (L2) speech decreases, and this processing cost is mitigated by experience with talkers of the same accent (Porretta and Tucker, 2019). Prior research has shown that listeners can adapt to unfamiliar L2-accented speech and generalize the adaptation to different L2-accented talkers with the same L1 (Bradlow and Bent, 2008). The current study investigates what type of L2-accented speech exposure (high or low intelligibility) best reduces later listening effort to a novel talker of the same accent. We hypothesize that exposure to highly intelligible L2 speech supports adaptation to the features of an accent because it allows listeners to map accented tokens to the talker’s intended lexical representations. Alternatively, adaptation to a less intelligible talker may require exposure to other low-intelligibility speech, which is characterized by features that deviate more significantly from L1 norms. To test this, participants were exposed to simple sentences spoken in English by either L1 talkers, highly intelligible L2 talkers, or less intelligible L2 talkers. At test, all participants listened to different sentences spoken by a previously unheard L2 talker while listening effort was measured via pupillometry. Data collection is ongoing.
Effects of time compressed speech and background noise on memory recall
Authors: Lee Min Young , Lim Sung-Joo
Published: July 2024

Listeners demonstrate remarkable flexibility in processing speech under varying conditions, rapidly adapting to acoustically distorted signals like time-compressed speech or speech in noise. Existing research has explored the immediate effects of time-compression and background noise on speech intelligibility, but the impact of these distortions on long-term memory recall for speech content is less well understood. With the growing use of online education platforms where listeners often prefer to use accelerated playback, understanding the effects of time-compression and noise on long-term memory from speech is crucial. Here, we investigated how speech distortions from time-compression influence explicit memory, and how this interacts with background noise. Participants (N = 35) each listened to six recorded lectures at varying time-compression rates (1.0x, 1.5x, 2.0x) and background noise levels (quiet versus babble noise at + 10 dB SNR). After each lecture, participants answered 10 multiple-choice questions probing explicit content recall. Listeners exhibited a significant decline in content recall with increased time compression, while background noise during the lecture did not impact listeners’ content recall. Our findings demonstrate that time-compressed speech negatively affects long-term memory recall, which may arise from the additional cognitive demands necessary to process time-compressed speech, even after perceptual adaptation.
Investigating the efficacy of autoencoders and other machine learning methods for studying dynamic oceanic processes in long-range acoustic propagation environments
Authors: Sen Gupta Ananya , Christensen Andrew J. , Linhardt Timothy , Kirsteins Ivars , Gemba Kay L.
Published: July 2024

Long-range acoustic propagation is a topic of great interest in applications like acoustic thermometry and underwater navigation. These applications utilize the measured arrival times and structure obtained from the transmitted probe pulses. However, they are highly dependent on the stability and repeatability of the ocean channel. In real oceans, random medium effects like internal waves [1] can induce considerable fluctuations and distortions to the received probe pulses. Our objective here is to investigate the use of machine learning (ML) methods such as autoencoders and other deep learning architectures to see if they can unravel and give insight into the dynamics of the ocean processes generating the fluctuations. In particular, we will investigate geometric concepts from braid, loops, and knot theory that can capture the changing shapes of smoothly deforming features that represent these processes. For the analysis, we will use transmitted frequency maximum length sequence (MLS) signal probe pulses from the 75 Hz Kauai Beacon source received at the International Monitoring Station near Wake Island at a nominal distance of 3500 km. We show that ML analysis can provide some useful insights. J. Xu, “Effects of internal waves on low frequency, long range, acoustic propagation in the deep ocean,” MIT Ph. D. dissertation (2007).
Feature extraction of small underwater sonar targets using neighborhood discriminant analysis
Authors: Christensen Andrew J. , Sen Gupta Ananya , Kirsteins Ivars
Published: July 2024

Linear discriminant analysis (LDA) stands as a widely used supervised feature extraction technique that maps data onto a low-dimensional subspace such that the between-class scatter is maximized while within-class scatter is minimized. Despite its utility, LDA faces many challenges, particularly when the within-class scatter matrix becomes singular due to small sample size. Other dimensionality reduction techniques, such as Neighbourhood Component Analysis (NCA), have been proposed as alternatives to LDA. NCA learns a linear transformation that maximizes the likelihood that datapoints of the same class are clustered together in the lower-dimensional space. However, the optimization of NCA relies on a non-convex cost function, making it prone to local minima. To address the challenges faced by both LDA and NCA, we propose a novel dimensionality reduction method named neighborhood discriminant analysis (NDA). Like NCA, NDA learns a linear transformation that aims to cluster datapoints based on class label. However, NDA is framed as an eigendecomposition problem, eliminating the need for non-convex optimization. We demonstrate the new approach on real small target sonar data. [Work funded by the ONR grant numbers N000142112420 and N000142312503, and DoD Navy (NEEC) Grant No. N001742010016.]
Automatic detection and 2D localization using a network of unsynchronized passive acoustic sensors in a dispersive waveguide
Authors: Goldwater Mark , Zitterbart Daniel P. , Bonnel Julien
Published: July 2024

The time-frequency positions of modal dispersion curves in shallow-water low-frequency impulsive signals are strongly dependent on source-receiver range, making them suitable for range-based localization. Here, we apply a temporal convolutional network (TCN) to spectrograms estimated from individual sensors in an array of unsynchronized hydrophones to simultaneously detect dispersive signals and produce source-range estimates. The TCN is trained on simulated signals generated over a spatial grid and various environmental parameters using the adiabatic approximation for normal modes. Assuming that the number of unique sources is unknown, range measurements from the same source across different sensors are simultaneously associated and used in localization. To accomplish this automatically, the proposed method considers all unique combinations of range measurements from every collection of k sensors. For every range measurement combination, if the location estimate generated using each subcombination of k-1 measurements is within a certain threshold of the remaining measurement, the whole collection is labeled as group-k consistent. All such groups of measurements are represented as neighboring nodes in a graph, and strongly connected components are used to calculate the final source location estimates. The whole detection/localization method is validated using both simulated and experimental marine data. [Work supported by NDSEG and ONR].
Establishing a definition for maximum hourly sound level
Authors: Rawlings Samantha , Lo Verde John , Dong Wayland
Published: July 2024

The use of a maximum hourly sound level is common within modern standards and design requirements, referenced in ANSI and ASHRAE standards, CHPS and LEED design guidelines, and is included in proposed language for draft ANSI standards, to name a few. Due to its wide adoption, its use is proscribed for education, hospitality, residential, retail, office, affecting nearly every design sector. The problem with this is that the metric lacks formal definition, leaving designers and evaluators without an agreed-upon approach to determine if an assessment is sufficient. Since 2014, the authors have presented multiple works assessing a proposed definition for maximum hourly sound level for vehicular sources. In this work, the authors will be presenting a continuation of the dataset. Long-term rooftop monitoring of the noise from an interstate was used to create an hourly max calculation noise model. The calculations are based on the statistical spread of the data and the learnings can be used to better design the structures for assessing environmental and traffic noise.
Recent sound absorption testing for baffles and blades
Authors: Good Kenneth W. , Bock Zachary A.
Published: July 2024

It’s easy for laboratories to make sound absorption measurement for baffles and blades treatment options. The difficult part is to make measurements and report results in a method that can be useable across a wide variety of users. As example, the architectural acoustic community, the specifying community, and marketing teams, all need to communicate effectively to a diverse clientele each with different needs and understanding. This paper will explore the methodology of one company’s latest investment to provide applicable baffles and blades data.
Acoustic tomography in a deep stratified lake; toward reliable estimates of large-scale currents by differential travel time
Authors: Wells John C. , Taniguchi Naokazu
Published: July 2024

Lakes worldwide are under increasing stress. For example climate change tends to strengthen stratification, promoting hypoxia and other nefarious effects. Targeting scientific investigation of such changes, and operational monitoring, we have been developing the lacustrine application of Coastal Acoustic Tomography (“CAT”) at Lake Biwa. The largest lake in Japan, Biwa supplies water to 14 million, and has a mean depth in its main basin of 41 m. Since November 2016, we have used M-sequences (5 kHz carrier) to execute the first successful tests of CAT at multikilometer ranges in any lake, with multiweek deployment of 3 (5) stations in Nov 2017 (2018). In 2021, we confirmed reciprocal transmission in spring through early summer between two stations at 7 km. In autumn, we often observed two distinct groups of arrivals. Ray-tracing simulations, based on sound-speed distributions from realistic hydrodynamic simulations, revealed a warm “surface channel” and cold “deep channel” whose predicted arrival times corresponded well with the observations. In continuing work to be reported, we are targeting reliable measurement of raypath-averaged currents by differential travel time, as already achieved by the second author’s team for the stronger tidal currents in the Seto Inland Sea at ranges of 3 km. Preliminary processing has revealed continuous evolution of phase difference between reciprocal receptions. By exploiting such phase differences we hope to achieve a useful level of accuracy in differential travel time despite the weak currents.
Acoustic measurements of photosynthetically formed gas bubble distributions in Posidonia oceanica seagrass meadows
Authors: Lyons Anthony P. , Xenaxi Angeliki , Pailhas Yan
Published: July 2024

The formation of gas bubbles on seagrass leaves commonly occurs during periods of high productivity in many aquatic ecosystems. Gas bubbles produced by seagrass meadows and subsequently released to the atmosphere can be a major component of primary production that is not quantified by measurements of dissolved oxygen. Acoustic propagation measurements have been explored as a possible tool for estimating photosynthetically relevant properties in situ. However, a lack of knowledge of bubble densities and size distributions has hindered acoustic attenuation as a quantitative measurement technique. Here we explore using measurements of acoustic backscattering from gas bubbles in and above Posidonia oceanica seagrass meadows as a more direct method for estimating bubble densities and size distributions. Acoustic backscattering data for this study was collected in September 2023 in Biodola Bay, Elba Island, Italy, using 1i–16 kHz linear frequency modulated pulses. Inversions of the broadband acoustic data yielded log-normal bubble size distributions with mean sizes between 0.4 and 0.5 mm in agreement with previously reported values. Estimated bubble densities yielded predicted attenuations in line with those found by propagation experiments. Results demonstrate the possible use of direct acoustic backscattering measurements as a supplement to dissolved oxygen measurements to better estimate rates of gas production which in turn can be used to better understand seagrass ecosystem properties, such as productivity and health.
Real-time 3D ultrasound imaging with a clip-on device attached to common 1D array transducers
Authors: Dong Zhijie , Li Shuangliang , Huang Chengwu , Lowerison Matthew R. , Yan Dongliang , Wang Yike , Chen Shigao , Zou Jun , Song Pengfei
Published: July 2024

Performing 3D ultrasound imaging at a real-time volume rate (e.g., >20 Hz) is a challenging task. While 2D array transducers remain the most practical approach for real-time 3D imaging, the large number of transducer elements (e.g., several thousand) that are necessary to cover an effective 3D field-of-view impose a fundamental constraint on imaging speed. Although solutions such as multiplexing and specialized transducers, including sparse arrays and row-column-addressing arrays, have been developed to address this limitation, they inevitably compromise imaging quality (e.g., SNR, resolution) in favor of speed. Coupled with the high equipment cost of 2D arrays, these compromises hinder the widespread adoption of 3D ultrasound imaging technologies in clinical settings. In this presentation, we introduce an innovative transducer clip-on device comprising a water-immersible, fast-tilting electromechanical acoustic reflector and a redirecting reflector to enable real-time 3D ultrasound imaging using common 1D array transducers. We will first introduce the principles underlying our novel technique, followed by validation studies incorporating simulation and experimental data. We will also demonstrate the feasibility of using the clip-on device to achieve a high 3D imaging volume rate that is suitable for advanced imaging modes such as shear wave elastography, blood flow imaging, and super-resolution ultrasound localization microscopy.
Graduate-level research communication course: Reflections on the first offering
Authors: Simon Julianna C.
Published: July 2024

In Fall 2022, I offered a new course on research communication within the Graduate Program in Acoustics at Penn State. With 8 students from Acoustics and Biomedical Engineering, the course was split into three modules of grant writing, paper writing, and outreach. I found the students were reasonably adept at critically reading single papers. They also generally understood the mechanics of writing sentences, although combining sentences into paragraphs, connecting ideas, reducing fluff, and adding emphasis to key points were areas that significantly improved throughout the semester. In the middle of the course, two mock grant review panels were held, which highlighted a need to help students develop strategies to cope with critical feedback. Thus, for the final project, students had the option to either write a journal paper draft or revise their proposal; they also presented orally on their final project. Additionally, communicating with the public and developing outreach activities were included in the course, which culminated in an Acoustics Day at a science center. Overall, I found identifying gaps in the literature using multiple sources and developing a cogent argument on how their proposed work would fill the gap needed the most improvement. [Partially supported by the NSF CAREER:1943937].
Risks and benefits of air taxis as perceived in Germany
Authors: Eißfeldt Hinnerk
Published: July 2024

In a telephone survey on the acceptance of civilian drones in Germany conducted at the end of 2022 a certain part of questions was dedicated to air taxis explicitly. While the attitude towards civilian drones tended to be slightly more positive than in the preceding study, in the again nationwide representative study attitudes concerning air taxis were revealed to be relatively balanced, with a slight negative tendency. Inferential statistical analyses showed factors such as age, gender, active experience with drones, and interest in environmental protection to be significantly associated with the attitudes towards both, civilian drones and air taxis. According to prior findings on the relevance of noise concerns for drone acceptance in general, the current study asked for some noise related information, revealing subjective noise sensitivity and general noise annoyance being associated with the acceptance of civilian drones as well as air taxis. Evaluating information from answers to free format questions about air taxis, this contribution reveals reasons why in Germany—although the contributing factors are the same—the general attitude towards air taxis is less positive compared to civil drones, and what role noise considerations might play.
Very short time Fourier transform for turbulent analysis
Authors: Oba Roger , Ramamurti Ravi
Published: July 2024

Nominally stationary acoustic sensors, e.g., buoy suspended or bottom anchored, are subject to various environmental flows. Direct numerical simulation of hydrodynamic flow around the acoustic sensor housing show turbulence along the housing surface. Time-space analysis of pressure variation along the surface of the housing shows significant acoustic-like surface interaction at low frequencies, even at very low mean flow velocity. In this case turbulence manifests as intermittently generated, irregular, coherent vortex structures that separate from the surface of the housing and advect downstream. These structures drive rapid surface pressure transients that propagate within the housing to the acoustic sensors as noise. Fractional Fourier Analysis provides joint time-frequency methods to extract transient features of turbulence. Use of Hermite-Gauss functions and phase relations are of particular interest. [This research is supported by 6.2 NRL base program sponsored by the Office of Naval Research. Distribution Statement A: Approved for public release. Distribution unlimited.]