bluuebunny/crossref_metadata_2025 huggingface dataset:
Measurement verification of reverberation time in seating areas close to the lateral wall in the auditorium
Authors: Xia Changqing , Tang Shiu Keung
Published: November 2021

This study started with building a database for existed Jockey Club Auditorium. Using impulse response method to measure numerous seats which cover the whole auditorium. Reverberation time distributed in four range parameters simply look up the hall acoustic performance visualizing by chromatogram. With the hall geometric data and the first lateral reflection path difference calculated from the point of source and receiver in seating areas close to lateral wall, The measurement shows in bass and middle frequency, Early delay time and T20 and T30 exists large fluctuation which reveals seat-dip effect in the front seat area. While in mid and high frequency, T30 presents stable values indicating the direct sound and lateral reflection are predominant in early sound attenuation. Computer simulation is using Pachyderm Acoustical Simulation plug-in based on parametric architectural design software Rhino. Sound attenuation can be calculated with surface shapes defined surface transform efficiency by diffracting energy. However, Reverberation time differences still exist by comparing actual measurement and computer simulation results.
Effect of interactive threats on foraging effort by endangered killer whales (Orcinus orca)
Authors: Holt Marla M. , Tennessen Jennifer , Hanson M. Bradley , Emmons Candice , Giles Deborah , Hogan Jeffrey , Ward Eric J. , Ford Michael J. , Thornton Sheila J.
Published: November 2021

Killer whales face many anthropogenic threats including vessel traffic, noise, and reduced prey availability. Here, we use high-resolution suction-cup Dtags to study the behavior of endangered Southern Resident killer whales that rely on biosonar to hunt salmon, and investigate how proximate vessels affect foraging behavior. From tag data, we identified subsurface behavior, including foraging and prey capture events. We then tested several vessel and associated sound, demographic, and environmental variables on behavioral state occurrence, time spent within each state, and foraging effort involving prey capture. Whales made fewer prey capture dives and spent less time in these dives when vessels had an average distance <400 yard (366 m). Lower prey abundance and higher vessel speed reduced prey capture probability, documenting the interplay between these effects. Finally, whales dove to depth more slowly while increasing dive duration to capture prey in the presence of vessel-emitted sonar, but descended more quickly with higher noise levels and closer vessels. Current efforts investigating foraging behavior and noise exposure over the diel cycle aim to better quantify foraging rates and activity budgets in this endangered population. These findings advance awareness of vessels and noise consequences on killer whales to inform conservation and management actions.
Soundscapes experienced by southern resident killer whales (Orcinus orca) in the Salish Sea and implications of vessel noise
Authors: Burnham Rianna , Vagle Svein , Thupaki Pramod , O'Neill Caitlin
Published: November 2021

Endangered southern resident killer whales (SRKW, Orcinus orca) use waters of the Salish Sea around southern British Columbia and northern Washington State to forage, following the in- migration of prey. However, these waterways experience heavy vessel traffic, with important SRKW foraging areas near to international shipping lanes. Here we use recordings from six passive acoustic moorings to characterize the soundscape, and determine spatiotemporal patterns on daily to annual scales. Particular focus is given to the noise levels in frequencies used by SRKW to communicate and echolocate. A numerical model is used to map the sound field between the mooring locations and examine the anthropogenic inputs in SRKW critical habitat in the Salish Sea by commercial traffic. Extrapolation of the model output to higher frequencies allows an examination of the impact this may have on the range of effective acoustics use, with implications for SRKW navigation, foraging success and group cohesion. Finer scale analysis of vessel signatures as they transit the study area aids in identifying areas of increased risk of disturbance for whales. In addition, differences in modelled to observed data will help the characterisation of other sonic inputs to the soundscape, including smaller recreational vessels.
Decades of progress in active fisheries acoustics: Many challenges remain
Authors: Stanton Timothy K.
Published: November 2021

The technology in this field has evolved significantly over the past 6 + decades to one involving operational scientific echosounders (single-, split-, and multi-beam; broadband) used across a multitude of types of platforms. In parallel with hardware development has been significant development of various methods of data processing, scattering models, and associated interpretation, yielding new understanding of behavior of fish and zooplankton. Overall, this field is very active and productive as well as one that is fertile for new growth and understanding. Trends of development, key milestones, and current challenges are briefly reviewed.
The use of deep learning techniques in acoustic tag signal identification
Authors: Steig Tracey W. , Nealson Patrick A. , Quirion Jean , Smith Frank
Published: November 2021

Acoustic telemetry is routinely employed to detect and quantify behaviors of aquatic animals. The ability to acoustically tag and release fish and aquatic animals allows researchers to monitor their presence/absence and fine-scale spatial movement. One method of identifying individual tags employs a period signal encoding (HTI signal) method, which utilizes the full transmitted acoustic energy for both detection and identification, providing significantly greater detection ranges than encoded BPSK signal types. However, identifying period encoded tags typically requires manual data review by trained human analysts, resulting in increased processing cost and a lack of scalability. Innovasea conducted a project to determine the efficacy of applying modern Deep Learning techniques to automatically identify animals implanted with period encoded acoustic tags. Specifically, the well-known U-Net model was adapted and trained to detect period encoded acoustic tag signals from raw acoustic telemetry data. The raw data is first transformed into a spectrogram-like image representation and the U-Net learns an image mask that separates actual tag detections from noise and interference. The trained U-Net achieved greater than 95% acoustic tag identification accuracy when compared to tag identification performed by human analysts on the same test data. Numerous examples of the U-Net performance will be presented.
The effect of resonator shape on air flow in flue organ pipes
Authors: Buzbee Elizabeth , Moore Thomas R.
Published: November 2021

The effect of changing the shape of the resonator of a flue organ pipe was studied in a replica of an organ pipe that was printed using transparent media. The pipe was designed so that the rectangular resonator could be replaced by a cylindrical one of the same cross sectional area, and the airflow in the pipe was imaged using high-speed transmission electronic speckle pattern interferometry. The resulting images reveal that there is a fundamental difference in the flow when the resonator shape is altered. Differences in the airflow and sound attributable to the shape of the resonator will be discussed.
Comparing two statistical models for low boom dose-response relationships with correlated responses
Authors: Vaughn Aaron B. , Doebler William , Ballard Kathryn , Rathsam Jonathan
Published: November 2021

This study compares two statistical modeling approaches to construct summary dose-response curves for low boom community noise surveys. NASA field survey data were used that consist of multiple responses from a survey participant. These data require an approach that accounts for the correlation among repeated annoyance observations from the same participant. A multilevel model accounts for the correlation by allowing estimated parameters to vary with each survey participant. On the other hand, a population average model utilizes generalized estimating equations and accounts for the correlation via a user-specified within-subject correlation structure. A visual comparison of the dose-response curves for these two methods reveals similar results. When comparing specific points along the summary curves, the multilevel model yields more precise confidence bounds than the population average model. The similarity between the summary curves derived from each model lends validity to both approaches for approximating a population representative summary curve.
Infant ear canal modeling
Authors: Kurdila Hannah , Vignola Joseph , Maruvada Subha
Published: November 2021

MRI scanners produce significant sound pressure levels (SPLs), sometimes in excess of 120 dBA. While the effect of these SPLs has been explored for adult patients, it is currently unknown how they affect neonatal patients. One approach to understanding the effect of this noise exposure on infants is to use existing experimental data that associates a given sound level to a hearing outcome in adults and re-interpret it with respect to infants. This approach requires modeling the differences between the adult and infant hearing apparatus—outer (pinna + concha + ear canal), middle and inner ear. In this presentation, we will present simulations of the infant ear canal SPL gain and compare it to the expected acoustic exposure of the neonate during MRI brain procedures to better understand the effects of MRI noise on the neonatal ear.
Detection thresholds with active hearing protection devices in quiet and noise
Authors: Thompson Eric R. , Pryor Nina J.
Published: November 2021

The Real-Ear Attenuation at Threshold (REAT) method is a standardized (ANSI/ASA S12.6) method of assessing the passive attenuation of hearing protection devices (HPD), including electronic HPDs with the electronics turned off. Characterizing HPDs with electronics turned on is performed with the Microphone in Real Ear (MIRE) method, or with an acoustic test fixture, as described in ANSI/ASA S12.42. It has been proposed that the REAT method may be useful for characterizing the effect of HPDs on detection of ambient sounds for HPDs with hear-through microphones, including the effects of electronics noise from the HPD. In this study, detection thresholds were measured with three electronic earmuff-type HPDs using a method similar to the REAT method. Detection thresholds were obtained for warble tones presented from a loudspeaker placed in front of the subject in a sound treated room. Thresholds were measured with unoccluded ears and with three HPDs with hear-through capability in quiet and in noise. In addition, thresholds were measured with one of the HPDs with foam earplugs inserted into both ears to simulate a conductive hearing loss. The results point to the interaction between device noise, ambient noise, and human absolute thresholds as limitations on the detectability of ambient sounds.
Effects of mandibular motion on earplug function
Authors: Eddins David A. , Armstrong Steve , Budinsky Robert , Higgins Nathan , Juneau Roger , Formby Craig
Published: November 2021

Earplug attenuation depends on developing and maintaining an acoustic seal between the earplug and the walls of the ear canal. During use, earplugs may lose some of their effectiveness by loosening over time and with activity, thereby compromising the acoustic seal in the ear canal and allowing more sound to enter the ear canal from the sound field. Mandibular motion (as during talking, singing, chewing, breathing) provides a nearly continuous source of ear canal movement resulting from forces applied by the condyle of the mandible. Individual differences in ear canal geometry also may impact the effects of mandibular motion on earplug function. The effects of mandibular motion on earplug function were investigated using an acoustic test fixture that combined (1) individualized pinna and ear canal structure beyond the second bend, inclusive of the mandibular bump, with (2) a standardized ear simulator (GRAS RA0045) with ear canal extension and microphone (GRAS 40AH ¼”). Mandibular motion was simulated using a computer-controlled drive-pin with a silicone tip that articulated with the mandibular bump of the individualized silicon ear canal. Results indicated that as few as 50 cycles of mandibular motion could significantly compromise earplug function and that this compromise was highly subject-dependent.
David Blackstock and the University of Rochester
Authors: Dalecki Diane
Published: November 2021

This presentation provides perspective on David T. Blackstock at the University of Rochester in Rochester, NY. Dr. Blackstock’s interactions with the University of Rochester began early in his academic career. In the early 1960’s, he was an associate professor in the Department of Electrical Engineering at the University of Rochester for several years. Beginning in 1970, he moved to The University of Texas at Austin for his long academic career with the Applied Research Laboratories and the Department of Mechanical Engineering. Then in 1987, enabled through collaborations with Dr. Edwin L. Carstensen, Dr. Blackstock returned to Rochester to spend his summers at the University of Rochester. That same year, he became a visiting member of the Rochester Center for Biomedical Ultrasound directed by Dr. Carstensen. Dr. Blackstock continued to spend summers at the University of Rochester for several decades. Summers with Dr. Blackstock were filled with engaging scientific discussions, research collaborations, and interactions with many students and faculty in various departments. During the summer, he regularly taught his Fundamentals of Acoustics course and, during select years, his Nonlinear Acoustics course. Dr. Blackstock’s longtime engagement with the University of Rochester greatly impacted the careers and lives of many students and faculty.
David Blackstock and the parametric array in air
Authors: Pompei Joseph , Hamilton Mark F.
Published: November 2021

In 1971, Peter Westervelt and David Blackstock attended a Navy meeting on underwater acoustics. Following mention of rumored failed attempts to create a parametric array in air, Peter offered reasons why it should not be expected to work in air. David, sitting nearby, said to him “Bulls**t Peter!,” to which Peter responded “Alright, you prove it!” After returning to Austin, David proceeded to do just that with master’s student Mary Beth Bennett, whose experiments resulted in their 1975 JASA paper “Parametric array in air.” In the 1990s, Joseph Pompei, then a master’s student and former engineer at Bose, became interested in the airborne parametric array. His vision was to use the technology to make a practical directional loudspeaker and enable new ways to control sound. Facing similar skepticism from the audio industry, Pompei responded with a doctoral dissertation demonstrating the first high performance, low distortion parametric loudspeaker. The main technical challenges he overcame were developing correct signal processing methods and creating an ultrasonic transducer and amplifier with adequate bandwidth, power and fidelity. Now commercially successful as the “Audio Spotlight” by Holosonics, it has proved successful in digital signage, museums, retail displays, libraries, and similar places that require well localized sound.
A general-purpose pipeline to interface the Tympan hardware with an external computer
Authors: Benesch Danielle , Raj Kocherla Nithin , Voix Jeremie
Published: November 2021

Some audio applications require resource-heavy algorithms which cannot be run on real-time digital signal processors such as the Tympan hardware directly due to memory and processing constraints. These algorithms can, however, run on an external computer (PC), and their outcomes can be relayed back to the Tympan where necessary adjustments can be made in the audio processing elements of the underlying hardware. The proposed pipeline includes a 4-channel audio input, the algorithm running on the PC, the Tympan hardware, and a 4-channel output. The Tympan hardware acquires input audio from the Tympan’s onboard and/or external microphones with the I2S protocol. The input data are transmitted to the algorithm running on the external PC via USB/Serial communication by exposing the Tympan as a soundcard interface. The output of the PC algorithm is then sent back to the Tympan, through serial communication, so that the underlying audio processing parameters are adjusted. A possible use case is also discussed: a detection algorithm runs on the PC and serial commands are sent back to Tympan to alter the audio outputs, e.g., by playing masking noise. The pipeline is built such that each element can be used independently, abstracting the interconnection of the pipeline elements.
Meta-mass systems: A harbinger for a vibration energy harvesting technology
Authors: McCoy John J.
Published: November 2021

For a limited class of fuzzy structures, the “master” structure is a grounded Newtonian mass wherein are distributed the “fuzzies,” modeled as nearly equal magnitude sprung masses with resonances densely filling a frequency band. We term said system a grounded “meta-mass,” showing a properly designed “meta-mass vibration energy harvester” addresses a fundamental technical obstacle to the harvesting of waste mechanical energy, the so-called “aperture” problem; thereby making meta-mass systems a harbinger for a vibration energy harvesting technology. Shown is the oscillators’ resonances provide ``bounds’’ on the system resonances, the relative “locations” of the system resonances within its bound-pair being determinative of system behavior. Then shown is the value of a certain combination of non-dimensional system measures—twice the fractional bandwidth squared divided by the product of the mass ratio and the natural log of N—is determinative of said relative locations. Finally shown is virtually all the energy inputted impulsively to the Newtonian mass is transferred to the encapsulated oscillators in a time of the order of the inverse of the band of oscillators, for a narrow system subclass wherefore each system resonance is near the center of its bound-pair.
Impaired phonetic learning in Parkinson's Disease
Authors: Heffner Christopher C. , Myers Emily B.
Published: November 2021

Parkinson’s Disease (PD) is a neurodegenerative disorder primarily associated with its motor consequences. However, it also has important implications for cognitive functions, including speech production (Altmann and Troche, 2011), the perception of speech intensity (Richardson and Sussman, 2019), and visual category learning (Shohamy et al., 2008), due to its effects on subcortical brain systems that rely on dopamine. In the present study, we examine the effects of PD on a variety of speech perception tasks: one of non-native phonetic learning, one of learning to comprehend rate-compressed speech, one of learning to understand accented speech, and one probing the use of speech rate to guide word segmentation. People with PD showed significant deficits when compared to young and age-matched controls in their ability to adapt to rate-compressed speech. Within the population of those with PD, participants using levodopa-based medication performed significantly worse on the non-native learning task than a small subset of participants without such medication. This suggests that changes in dopamine regulation and use associated with PD have significant effects on higher-level aspects of speech perception such as speech learning.
Speech characteristics of children with ASD to a humanoid robot and peers
Authors: Kinney Kaelin , Kondaurova Maria V. , Welch Karla C. , Kuravackel Grace M. , Pennington Robert , Barnes Gregory , Popa Dan
Published: November 2021

Children with Autism Spectrum Disorder (ASD) display difficulty engaging in social interactions. How does the use of a NAO robot during a social skills intervention affect the speech of children with ASD? This study examined whether the use of a NAO robot during a social skills intervention affects the quantity and quality of speech in six children (mean age = 11.6y, range = 10.5–12.6) with ASD who attended four weekly social skills intervention sessions. The speech quantity was defined as an overall speech rate (utterances per minute), initiation rate (utterances per minute addressing the robot/other child), and response rate (utterances per minute responding to the robot/other child). Mean length of utterance (MLU) in overall speech, initiations, and responses was also measured. A marginally higher initiation rate and overall MLU in speech to the robot were identified in later sessions. Children produced a significantly higher MLU in responses to the robot in later sessions. MLU for initiations were marginally higher than MLU for responses to the robot. No changes in speech rate or MLU in speech directed to other children were identified. This suggests children may require repeated exposure to the NAO robot before changes in speech characteristics are revealed.
Multitalker speech perception in neurodiverse populations
Authors: Emmons Katherine A. , Estes Annette , Dager Stephen , Maddox Ross K. , Astley Hemingway Susan , Thorne John C. , Lee Adrian KC , Lau Bonnie K.
Published: November 2021

The ability to selectively attend to one talker in the presence of competing talkers is a crucial skill employed in everyday life. In this study, multitalker speech perception thresholds were measured in three groups; Autism Spectrum Disorder (ASD), Fetal Alcohol Syndrome Disorder (FASD), and an age- and sex-matched typically functioning (TF) group. Participants listened to three simultaneous sentences from the Coordinate Response Measure corpus: the target stream to be attended (0° azimuth) and two spatially separated (±45° azimuth) masker streams. Participants were asked to identify the color and number associated with the callsign “Charlie.” Target-to-masker ratios (TMRs) were estimated based on the average of four runs in which the target was fixed at 40 dB SPL and maskers were adaptively varied using a one-up-one-down procedure to estimate 50% correct. The target speaker was always male; the two maskers were either male/male or female/female. Overall, TMR thresholds were higher in both ASD and FASD groups than the TF group. Additionally, a negative correlation between intellectual ability and TMR thresholds was observed. These preliminary results suggest intellectual ability may impact how well listener’s perceive speech in multitalker situations, especially in neurodiverse populations.
Lexical effects on the perceived clarity of noise-vocoded speech in younger and older listeners
Authors: Tamati Terrin N. , Moberly Aaron C.
Published: November 2021

When listening to degraded speech, such as speech delivered by a cochlear implant (CI), listeners can make use of linguistic knowledge to facilitate speech recognition. Yet, the extent to which linguistic knowledge can be effectively used to compensate for degraded input may depend of the level of degradation and the age of the listener. The current study investigated lexical effects in the compensation for degraded speech via noise-vocoder simulations of CI hearing in younger and older normal-hearing listeners. In two online experiments, listeners rated the clarity of noise-vocoded sentences, with varying filter slopes. Lexical information was provided in the form of text priming and/or the lexical content (i.e., lexical frequency and neighborhood density) of key words in the sentences. Results showed that lexical information from both text priming and the lexical content enhanced the perceived clarity of degraded speech, but the strength of the lexical effects depended on the level of degradation. Additionally, lexical information appeared to equally benefit younger and older listeners. Findings demonstrate that lexical knowledge may be used in cognitive compensation during the recognition of noise-vocoded speech in both younger and older listeners. However, these mechanisms may not be as effective when the signal is highly degraded.
Talker variability in Mandarin spoken word recognition: Comparing normal and hearing-impaired listeners
Authors: Zhang Yu , Wang Suju , Shang Yingying
Published: November 2021

During natural speech processing, talker variability is inherently embedded in the acoustic signal and may affect the speed and accuracy of processing. Prior studies have demonstrated the effect of talker variability in normal listeners. What is the effect of inter-talker variability in people with hearing impairment? This study aims to explore the question in native speakers of Mandarin Chinese, trying to compare listeners with and without hearing impairment. Participants are instructed to listen to pre-recorded pairs of Mandarin Chinese words (e.g., in English translation, happy – sad) and make responses only to the second item of each pair (e.g., sad) in a short-term priming paradigm. The speed and accuracy of the responses are measured. The fact that the pairs were spoken by either the same or different native speakers of Mandarin Chinese allows us to examine the effect of speaker voice change on the reaction time and accuracy. The hearing-impaired listeners appear to be facilitated more by the same speaker voice condition in terms of response accuracy, compared with the listeners without hearing impairment. There also seems to be more semantic facilitation to the hearing-impaired in terms of reaction time, although this effect does not seem to be modulated by talker variability.
Variability in the production of /s/ by adults who do and do not stutter
Authors: Lei Xiaofan , Munson Benjamin
Published: November 2021

Speech produced by typically developing adults at their habitual speech rate is characterized by low utterance-to-utterance variability (Smith et al., 1995). Wiltshire et al. (2021) posited that higher production variability underlies or causes stutter-like disfluencies among individuals who stutter. Given the hypothesis, higher variability should be observed among individuals who stutter at all linguistic units. Adults who stutter (AWS) have been found to have higher articulatory variability when producing sentences and nonwords relative to controls (Kleinow and Smith, 2000; Smith et al., 2010). However, no studies have investigated the group difference at thelevel of individual phones, such as when producing the fricative /s/ (Munson, 2001; 2004). The current study attempts to fill this gap of knowledge by comparing the acoustic variability of /s/ between matched pairs of AWS and fluent controls (n = 18). We found that /s/ was produced with greater variability in nonwords relative to words (β = 0.48, F[1,617] = 23, p < 0.001, 95% CI [0.29, 0.68]). However, AWS did not show higher acoustic variability relative to their fluent peers (β = 0.16, F[1,16] = 0.09, p > 0.1, 95% CI [-0.94, 1.26]). Results are not consistent the hypothesis that the speech of AWS is always more variable than fluent controls. Implications will be further discussed.
Individual differences in speech and language within 72 h after a concussion
Authors: Grabowski Caryn , Ahn Jisook , Testa Anthony J. , Ackermann Sarah , Hernandez Emily , Terlecky Austin , Grunstein Matthew , Silverman Ava , Scher Gabe , Craig Deja , Kelly Kaitlin , Cunningham Mercedes , Patel Sona
Published: November 2021

Traumatic brain injuries (TBIs) cause immediate and temporary alterations in brain function that result in a variety of symptoms including impacts on speech. Symptoms of dysarthria and aphasia occur but are primarily reported in moderate-severe injuries. Less is known about the specific changes in acoustic parameters and language that occur in mild concussions. A total of 250 athletes at Seton Hall University participated in baseline speech testing. Speech testing was performed again in a subset of 15 athletes who sustained a mild concussion (within 72 h). Speech and language analysis was performed on samples derived from a picture description task. Statistical comparisons showed minimal changes in acoustic parameters on average, but individual differences were present. Specifically, some individuals presented with a flatter prosody and/or lower mean pitch than others post-injury compared to baseline. Although no clear pattern of acoustic changes emerged, language analysis (semantics, syntax, morphology) showed an overall reduction of complexity of productions. Language outcomes indicate that changes in speech symptoms occur even in mild cases. Further investigations should examine the speech and language changes after a head injury against the individual’s complete symptom profile and neurological regions of damage in a larger sample size.
How hospital noise impacts intelligibility of medically related sentences
Authors: Bent Tessa , Baese-Berk Melissa M. , Ryherd Erica E. , Perry Sydney
Published: November 2021

Hospital noise often exceeds recommended sound levels set by health organizations, potentially leading to communication breakdowns between healthcare providers and patients. Despite this concern, there is a paucity of literature on speech perception in healthcare settings. Studies investigating how hospital noise impacts intelligibility have been limited by not using medical terminology or not including objective intelligibility measures with listeners. To address these gaps, a new corpus of medically related sentences, with quantified word frequency, word familiarity, and sentence predictability, was employed in an intelligibility task. Recordings of the medically related sentences with different familiarity/frequency types (low/low, high/low, high/high) plus standard speech perception sentences were presented to listeners in quiet, speech-shaped noise, or hospital noise for transcription. Results showed main effects of sentence type and listening condition and an interaction between the two. Although listeners were less accurate on all sentence types in noise, sentences with low frequency and/or familiarity words showed the greatest intelligibility decreases in noise. Thus, miscommunications between healthcare providers and patients could occur when medical information is provided in noisy settings like hospitals, situations that could lead to lack of comprehension of discharge instructions or other medical communication errors. [Work supported by the IU Institute for Advanced Study.]
Communication channel, emotion category and attentional requirement modulate emotional speech perception: Electrophysiological and behavioral evidence
Authors: Lin Yi , Fan Xinran , Zhang Hao , Chen Fei , Zhang Hui , Ding Hongwei , Zhang Yang
Published: November 2021

The present study employed electrophysiological and behavioral measures to examine the modulatory effects of communication channels, emotion categories and attentional requirements on the timing, neural oscillatory activities and behavioral performances of emotional speech processing. Thirty participants (15 women and 15 men) were presented with spoken words denoting happiness, neutrality and sadness in prosody or semantics. They were asked to attend to the emotional content (explicit tasks) and speakers’ gender (implicit tasks) of the stimuli. Auditory event-related potential (AERP), inter-trial phase coherence (ITPC), accuracy, and reaction time data were recorded and analyzed. Results indicated that prosody (relative to semantics) and happiness (relative to neutrality and sadness) enjoyed greater processing advantages during the sensory processing of acoustic signals, initial derivation of emotional significance, and cognitive evaluation of emotional speech. However, these channel and category-specific effects were attenuated in implicit tasks and sadness processing. Explicit tasks also displayed greater perceptual sensitivities in early auditory processing, but exhibited greater processing difficulty during decision making. In addition, ITPC values served as significant predictors of AERP amplitudes across the three contextual factors. These findings reveal the neural and behavioral underpinnings of emotional speech processing, which is mediated by the dynamic interplay among channels, emotions and attention.
Using ideal binary masking based on signal-to-noise ratio of temporal amplitude envelope to improve the intelligibility of speech in noise
Authors: Soleymanpour Rahim , Golzari Kia , Kim Insoo , Heiney Erin , Marquis Hillary , J. Brammer Anthony
Published: November 2021

Ideal Binary Masking (IBM), using prior information, improves speech intelligibility by attenuating noisy components with a scaling factor applied to the noise. The main challenge is to construct an appropriate decision-making model to identify noise- or speech- dominant components. In this study, we utilized the signal-to-noise ratio (SNR) of the temporal amplitude envelope in the frequency-time domain. We firstly divided the noisy speech from 200 Hz to 6 kHz, processed by MATLAB, into 16 contiguous subbands each with bandwidth approximately 1.5 times an equivalent rectangular bandwidth. The subband envelopes were produced by means of the absolute value of the signal. SNRs of the temporal envelope were calculated for 40 ms windows. The mask was unity when the SNR was greater than −5dB; otherwise, it was 0.5. We evaluated the performance of the proposed IBM on word scores obtained with different speech in speech-spectrum shaped noise SNR values of −2, −4, −6, and −8 dB. Sixteen native speakers (age 28 ± 3 years) with normal hearing were recruited for the study and underwent the Modified Rhyme Test to assess intelligibility. Statistically significant increases of up to 20% in mean word scores were obtained by this IBM. [Work supported by NIOSH.]
Effects of transparent and opaque facemasks on audiovisual speech perception
Authors: Koenig Laura , Randazzo Melissa , Smith Paul J. , Priefer Ryan
Published: November 2021

Face masks reduce the spread of COVID-19 but hinder speech perception. Masks reduce visual feedback, filter the acoustic signal, and affect speech movements. Older adults, particularly those with aging related hearing loss, are impacted by the reduction of auditory and visual cues when communicating with masks. Face masks also pose a challenge to medical professionals when relaying important information to patients. Transparent masks with a clear window across the mouth may facilitate masked communication; however it is unknown if visual feedback through the transparent mask facilitates speech perception given the reduced acoustic signal. We examined speech intelligibility through masks using unpredictable sentences, recorded by two adults, that limit guessing words based on context. We presented sentences mixed with multitalker babble in four conditions (no mask, disposable, transparent, and N95 mask). Listeners, recruited online, typed what they heard. Percent words correct was scored by a text-matching algorithm, allowing for morphological errors and homonyms. We will determine the visual benefit of the transparent masks over the other masked conditions. Results of this study may increase awareness among healthcare providers about how different mask types can affect communication and empower individuals to decide which masks to wear in different communicative contexts.
Recommendations for improving communication while performing tasks under noisy, reverberant and delayed communication transmission conditions
Authors: Rawool Vishakha
Published: November 2021

This study, which is part of a larger study, was designed to generate recommendations for improving remote collaborative communication. Ninety (45 men and 45 women) participants were divided into 30 teams with 3 members in each team. Each team built different LEGO models under three communication transmission delay conditions in the following order: No delay, 750 ms delay and 5000 ms delay. One individual served as the LEGO builder, who did not have access to the instructions. The remaining two team members provided instructions for building the models. Participants were in separate rooms and communicated with each other via earphones. Background white noise (50 dB SPL) was introduced in the ear canals and reverberation was applied to the transmitted communication. After the three delay conditions, participants responded to questions. Their responses were coded into different categories for conceptual analyses. Several recommendations emerged including providing accurate, concise and clear instructions, ensuring ongoing alignment of the spatial orientation of the LEGO model across team members, standardizing/aligning phrases or terms across the team members, and increasing team and task familiarity. Additional recommendations will be discussed during the presentation with examples. [Acknowledgement: Graduate students Sydney Osbarn and Melissa Reyes assisted in data collection.]
Perceived effects of communication transmission delays during remote collaboration under noisy and reverberant settings on communication, team and task performance, and well-being
Authors: Rawool Vishakha
Published: November 2021

This study, which is part of a larger study, was designed to investigate the perceived effects of communication delays on well-being, communication, and task and team performance. Ninety (45 men and 45 women) participants were divided into 30 teams with 3 members in each team. Each team built different LEGO models under three communication transmission delay conditions in the following order: No delay, 750 ms delay and 5000 ms delay. One individual served as the LEGO builder, who did not have access to the instruction manual. The remaining two team members provided instructions for building the models. Participants were in separate rooms and communicated via earphones. Background white noise (50 dB SPL) was introduced in their ear canals and reverberation was applied to the transmitted communication. After each of the delay conditions, participants completed a visual analogue scale with 39 items related to well-being, communication and team and task performance. ANOVA revealed significant perceived effects of 5000 ms delay on communication, team dynamics, task performance and well-being. In the well-being categories, women yielded poorer scores than men. Detailed results will be included in the presentation along with recommendations. [Acknowledgement: Graduate students Sydney Osbarn and Melissa Reyes assisted in data collection.]
Oceanographic property estimation with ambient noise on the New England shelf
Authors: Gebbie John , Siderius Martin
Published: November 2021

Acoustic propagation through the ocean waveguide is strongly affected by the vertical water sound velocity profile and the geoacoustic properties of the seabed. Direct measurement is often prohibitive, however, the ambient noise field itself contains information about these properties, and therefore can be estimated via passive acoustics. The ambient noise field is driven by wind and wave action at the surface. This energy propagates through the waveguide ultimately affecting the vertical distribution of the noise energy, but also picking up environmental information along the way. A vertical line array is a useful tool for observing the noise field and estimating properties of the environment. The observed element data consists of correlated stochastic processes. The fundamental limit on the accuracy with which environmental properties can be estimated is given by the Cramér-Rao Lower Bounds (CRLB). The CRLB thus provides useful insights into which environmental properties can be independently or jointly estimated. A maximum likelihood (ML) estimator will asymptotically approach the CRLB as the number of snapshots increases. The ability to estimate different environmental properties and ML performance is demonstrated with simulations and real data collected in May 2021 off the New England shelf. [Work sponsored by the Office of Naval Research.]
Localizing a quiet moving source with range-coherent matched field processing
Authors: Akins Franklin H. , Kuperman William
Published: November 2021

We introduce range-coherent matched field processing (MFP) to lower the signal-to-noise ratio (SNR) threshold required to successfully localize a narrowband source in a well-characterized shallow water environment. To accomplish this, range-coherent MFP coherently processes snapshots formed from a moving source. This approach differs from existing MFP approaches that treat each snapshot as having a random phase. Range-coherent MFP requires determination of the source phase acquired between snapshots. With that information, MFP can be applied to the cross-spectrum of the ''supervector'' of concatenated snapshots acquired at different times. Viewed another way, range-coherent MFP is simply MFP applied to a passive synthetic aperture formed from a moving source. The synthetic aperture geometry depends on source velocity, which is included in the MFP search space. Range-coherent MFP produces robust velocity estimates at low SNR, which permits the use of a longer FFT in pre-processing. The synthetic aperture array gain plus the increased input SNR afforded by the enhanced pre-processing significantly lower the signal level required for successful localization. We present the application of range-coherent MFP to data from the SWellEx-96 experiment to localize a source that is too quiet for conventional methods to localize.
A non-Euclidean approach to source localization in an ocean waveguide
Authors: Finette Steven I.
Published: November 2021

In recent years, there has been significant interest in the natural extension of signal processing techniques based on Euclidean distances to non-Euclidean metrics. Covariance or cross-spectral density matrices (CSDMs), often the basis for localization and detection processors, can be interpreted as “points” in a Riemannian manifold on which a distance metric can be defined. This mathematical structure has proven quite useful in a number of processing schemes involving detection, source localization and classification of data. In this presentation, I consider a fully geometric, non-Euclidean approach to source localization in a simulated shallow water ocean waveguide where the sound speed field is stochastic due to the presence of internal gravity waves. A number of different CSDM estimators are constructed by formulas describing the mean or median of a set of sample dyads obtained from Green function realizations of the propagating field from a source point to a vertical array of acoustic sensors. These CSDM estimates are not derived by statistics, but through the geometric structure of the CSDM. The resulting estimates are incorporated into matched-field localization processors whose measure of similarity or distance between CSDMs is determined on the manifold. Work supported by funds from the Office of Naval Research.
Gaussian process model selection for parameter estimation in ocean acoustics
Authors: Jenkins William F. , Gerstoft Peter
Published: November 2021

The dynamic nature of the ocean environment presents numerous challenges to parameter estimation in ocean acoustics. We present a variational Bayesian method which selects an optimal model for parameter estimation in ocean acoustics using Gaussian process (GP) regression. GP model selection is based on Bayesian model selection, which treats the parameters and kernel selection of the model as hyperparameters and uses Bayesian inference to compute a model’s likelihood with respect to its hyperparameters. However, in Bayesian model selection, the integrals over parameter space are often intractable and require approximation or sampling methods such as Markov chain Monte Carlo. GP renders these integrals tractable by treating hyperparameters as multivariate Gaussian distributions, which greatly speeds up the optimization. Furthermore, model selection using GPs allows for straightforward evaluation of uncertainty in the selected model’s likelihood. In this study, we compare results of acoustic parameter estimation from Bayesian and GP model selection using an ocean acoustic propagation model.
On the invariance of the waveguide invariant
Authors: Rouseff Daniel , Zurk Lisa
Published: November 2021

Constructive and destructive interference between different acoustic ray paths or modes is a common feature of propagation in the ocean waveguide. Russian investigators first distilled these interference effects into a single quantity, the so-called waveguide invariant. Over the last two decades, investigators have incorporated the waveguide invariant into signal processing algorithms for applications including localization, tracking, and environmental inversion. An important caveat is that a specific numerical value for the waveguide invariant sometimes applies for only a subset of all propagating modes; different numerical values can apply for different subsets. Under appropriate conditions, the waveguide invariant can appear to bifurcate with acoustic data simultaneously showing traits from both values. In the present work, a derivation is sketched that emphasizes what is actually invariant about the waveguide invariant. The proposed approach permits a more ready integration of the waveguide invariant into signal processing algorithms. Examples are presented for both deep- and shallow-water scenarios.
Nature of waveguide invariant for multilayer sediment on New England shelf in 20–4000 Hz band
Authors: Knobles David P. , Bonnel Julien , Lin Ying-Tsong , Hodgkiss William , Neilsen Tracianne B. , Wilson Preston S.
Published: November 2021

The waveguide invariant βmn (f), m,n = 1,2,…,N where N is the number of observed modes is inferred from sound emitted by merchant ships and from a towed mid-frequency source emitting CHIRP signals. The acoustic data were recorded on VLAs deployed in the New England Mudpatch during the Seabed Characterization Experiment 2017 in about 75 m of water. The received spectrograms show clear striation patterns out to 4 kHz. The seabed consists of about 12 m of mud over multiple sand layers extending down to about 300 m. Starting at β12(F) = ∞ at F about 25 Hz, βmn(f) is extended to 4 kHz. Below 500 Hz βmn exhibits a non-Pekeris behavior which results from the depth dependence of the multilayer seabed. Above 500 Hz the behavior of βmn begins to exhibit more of a Pekeris behavior, although there exist features that result from the finite thickness of the mud layer. A conclusion is that βmn (f) is sensitive to the geoacoustic structure over a large bandwidth, and thus has the potential to be utilized for remote sensing of the seabed with wideband sources.
Focusing Green's function using the waveguide invariant theory
Authors: Kim Daehwan , Kim Donghyeon , Kim J. S.
Published: November 2021

Sound propagated in an ocean waveguide is greatly influenced by the characteristics of the underwater acoustic channel. Thus, knowledge of the propagation characteristics (i.e., impulse response or Green’s function) is key to various ocean remote sensing applications. Recently a relationship was verified that the waveguide invariant property allows a time-domain Green’s function observed at one location to be extrapolated to adjacent ranges [Song et al., J. Acoust. Soc. Am. 149, 2173–2178 (2021)]. Here, we exploit this relationship to coherently combine the time-domain Green’s function from adjacent ranges to improve the SNR of the multipath impulse response. Numerical simulation results are presented and discussed.