bluuebunny/crossref_metadata_2025 huggingface dataset:
Interactions between auditory quality, linguistic context, short-term memory, and speech recognition
Authors: Bosen Adam K.
Published: April 2019

The ability to recognize speech depends on several factors, including the quality of the auditory input, linguistic context, and the listener’s cognitive abilities. These factors correlate with speech recognition, but less consideration has been given to how these factors interact. Here, we present evidence from two studies indicating that interactions exist between quality, context, and short-term memory. In study one, we demonstrate that contextual expectations across stimulus sets can determine the relationship between auditory quality and short-term memory. Specifically, recall of lists of digits is not affected by noise-band vocoding, whereas vocoding impairs both item identification and recall for lists of single syllable words drawn from a large, untrained set. In study two, we demonstrate that correlations between digit list recall and PRESTO sentence recognition are strongest when auditory quality is poor, whereas correlations between digit and word list recall weaken with decreasing auditory quality. This finding suggests that auditory quality and semantic context moderate the relationship between memory and speech recognition. We conclude that incorporating experimental measures of auditory quality, short-term memory, and recognition of speech materials with different contexts will provide a clearer perspective on how these factors relate to speech recognition in listeners with hearing loss.
Examining use of context in sentences: The recruitment of perceptual and cognitive-linguistic abilities in younger and older adults
Authors: Fogerty Daniel , Miller Rachel E. , Ahlstrom Jayne B. , Dubno Judy R.
Published: April 2019

Auditory-verbal contextual information facilitates speech recognition. In everyday listening, perceptual and cognitive-linguistic abilities influence speech recognition, but the role of these abilities in contextual use remains unclear. To assess the use of context, younger and older adults listened to sentences in multitalker babble and identified the final words from either high- or low-predictability sentence frames; difference scores between the two conditions indicate use of context. We also assessed (1) speech processing abilities, including masking release, talker segregation, and auditory-verbal closure, (2) cognitive abilities, including memory, inhibition, and speed of processing, and (3) linguistic abilities, assessed through verbal fluency and vocabulary knowledge. Compared to younger adults, older adults had better contextual use and vocabulary knowledge but had poorer recognition of final words with increasing age. These measures, combined with thresholds and age, were entered as predictor variables to explain performance for high- and low-context sentences. Preliminary results indicate that vocabulary knowledge and auditory-verbal closure were associated with contextual use and final word recognition in low-context sentences, respectively. Final word recognition in high-context sentences involved a combination of abilities. Thus, recruitment of perceptual and cognitive-linguistic abilities depends, in part, on the availability of sentence context. [Work supported by NIH/NIDCD.]
Acoustic characteristics of voiceless English fricatives (/f θ s/) produced by native English and native Japanese speakers
Authors: Aoyama Katsura , Flege James E. , Akahane-Yamada Reiko , Yamada Tsuneo
Published: April 2019

This study examined productions of three English voiceless fricatives (/f θ s/) produced by native Japanese (NJ) and native English (NE) adults and children (16 participants each in 4 groups). The purpose of this study was to investigate acoustic characteristics of these fricative productions that were evaluated using intelligibility ratings in Aoyama et al. (2008). The following acoustic parameters were selected based on their importance in differentiating English fricatives (Jongman et al., 2000): noise duration, spectral moments, normalized amplitude, and relative amplitude. A total of 768 tokens were acoustically analyzed (256 each of /f θ s/). The results showed that there are many differences in acoustic characteristics between NJ and NE speakers. First, the durations of /θ/ were longer in the NJ speakers’ productions than in the NE speakers’ productions on average. Second, the spectral mean for /s/ was lower in the NJ speakers’ productions than NE speakers’ productions. Third, the NJ speakers’ productions of /s/ did not differ from /f/ and /θ/ in normalized amplitude, whereas the NE speakers’ productions of /s/ showed higher amplitude than /f/ and /θ/. These findings will be discussed and compared to the findings based on intelligibility ratings in Aoyama et al. (2008).
Non-native perception of noise-vocoded speech
Authors: Jacewicz Ewa , Fox Robert A. , Lee Joy
Published: April 2019

Research has established that non-native speakers consistently underperform compared to native speakers in challenging listening environments such as in noisy backgrounds. Furthermore, when presented with variable talkers and regional accents, non-native listeners are unable to benefit from available phonetic details to the same extent as do native listeners [Jacewicz et al., J. Acous. Soc. Am. 144, 1864 (2018)]. The current study further inquired into this non-native processing deficit using noise-vocoded speech. Noise-vocoding not only models cochlear implant processing but is also informative regarding perception of degraded speech in normal-hearing listeners. Proficient Korean-English bilinguals were asked to identify talker dialect (Ohio, North Carolina) and talker sex responding to utterances processed through a noise-source vocoder at 4, 8, 12, and 16 channels. A separate task assessed intelligibility of this noise-vocoded speech. Since cues to talker dialect are found primarily in fine spectral structure, we expected them to perform increasingly worse than native listeners with the decreasing number of channels. Both native American English controls and Korean-English listeners adhered to the predicted pattern across tasks and conditions. The study provides further evidence that the ability to utilize partial spectral information is compromised in non-native speech processing.
Changes to perceptual assimilation following training
Authors: Kharlamenko Julia , Kabakoff Heather M. , Levy Erika , Levi Susannah V.
Published: April 2019

Adult speakers of American-English have difficulty perceiving front-back rounding contrasts in vowels, such as French /œ/-/o/. These difficulties stem in part from how learners map sounds in a second language onto native-language sound categories. We used a perceptual-assimilation task to test how perceptual training changes listeners’ mapping of nonnative speech sounds. One listener group was trained with a bimodal distribution of stimuli drawn from an /œ/-/o/ acoustic continuum, a second was trained with a unimodal distribution, and a third (control) group completed no training. The training incorporated active learning with feedback and lexical support. In the assimilation task, listeners heard French vowels and were asked to select which of eight English /hVd/ words best matched the French stimulus. Results revealed general perceptual training effects and differences in both trained groups. For /o/, both trained groups differed significantly from untrained controls. However, for /œ/, listeners in the unimodal group did not differ from the untrained controls, but the bimodal group differed from the untrained controls. Thus, exposure to a unimodal distribution can alter perception, but a bimodal distribution may have stronger effects on perception. Bimodal distribution may be better for facilitating assimilation of nonnative sounds to native-like categories.
A comparison between native and non-native speech for automatic speech recognition
Authors: Park Seongjin , Culnan John
Published: April 2019

This study investigates differences in sentence and story production between native and non-native speakers of English for use with a system of Automatic Speech Recognition (ASR). Previous studies have shown that production errors by non-native speakers of English include misproduced segments (Flege, 1995), longer pause duration (Anderson-Hsieh and Venkatagiri, 1994), abnormal pause location within clauses (Kang, 2010), and non-reduction of function words (Jang, 2009). The present study uses phonemically balanced sentences from TIMIT (Garofolo et al., 1993) and a story to provide an additional comparison of the differences in production by native and non-native speakers of English. Consistent with previous research, preliminary results suggest that non-native speakers of English fail to produce flaps and reduced vowels, insert or delete segments, engage in more self-correction, and place pauses in different locations from native speakers. Non-native English speakers furthermore produce different patterns of intonation from native speakers and produce errors indicative of transfer from their L1 phonology, such as coda deletion and vowel epenthesis. Native speaker productions also contained errors, the majority of which were content-related. These results indicate that difficulties posed by English ASR systems in recognizing non-native speech are due largely to the heterogeneity of non-native production.
Formant discrimination of speech and non-speech sounds in temporally modulated noise: Effect of language experience
Authors: Li Mingshuang , Xu Can , Liu Chang , Tao Sha
Published: April 2019

Recent studies showed that the residency experience of 1–3 year in English-speaking countries could improve temporal dip listening against energetic masking of noise (i.e., masking release), resulting in better performances in identification of vowel, consonant, and sentence in temporally fluctuating noise. The current study aimed to explore whether this advantage would be extended to speech and non-speech perception discrimination. Three groups of listeners were recruited: English-native listeners (EN), Chinese-native listeners in US with long US residency experience (i.e., 2.5–4 years; CNUL), and Chinese-native listeners in the US with short US residency experience (i.e., less than half a year; CNUS). Thresholds of spectral discrimination of speech and non-speech stimuli were measured. EN listeners are expected to have the lowest thresholds and highest masking release in vowel formant discrimination, followed by CNUL listeners, with highest thresholds and lowest masking release for CNUS listeners. Furthermore, the current study will also examine whether the effect of language experience on masking release would be presented for non-speech spectral discrimination in noise. [Work supported by the China National Natural Science Foundation (31628009).]
Modelling the effects of spectators on speech intelligibility in a typical soccer stadium
Authors: Hammond Ross , Mapp Peter , Hill Adam J.
Published: April 2019

Public address system performance is frequently simulated using acoustic computer models to assess coverage and predict potential intelligibility. Simulations are most-often completed in unoccupied spaces as this provides worst-case scenario intelligibility due to the reduced absorption. When the typical 0.5 speech transmission index (STI) criterion cannot be achieved in voice alarm systems, due to design difficulties, justification must be made to allow contractual obligations to be met. An expected increase in STI with occupancy can be used as an explanation, though the associated increase in noise levels must also be considered. However, numerous approaches exist when modelling the people which can produce significant discrepancies. This work demonstrates typical changes in STI for different spectator conditions in a calibrated stadium computer model. This includes different audience modelling approaches, distribution, capacity, posture (standing/seated), and atmospheric conditions. The effects of ambient noise are also considered. The results can be used to approximate expected changes in STI caused by different spectator conditions.
Assessing FES-induced muscle fatigue using ultrasound to determine the inverse neuromuscular model for optimal FES input
Authors: Kim Kang , Sheng Zhiyu , Sharma Nitin
Published: April 2019

Functional electrical stimulation (FES) has been successful in activating paralyzed or paretic muscles to restore limb functions of individuals with impaired gait function. However, when activating the limb joint motion through externally stimulating muscle, rapid onset of muscle fatigue becomes a critical issue that results in injury. To overcome this challenge, an optimal FES input to the neuromuscular system needs to be determined and updated in real-time in order to maintain an effective, safe limb function. The inverse neuromuscular model between the desired limb joint motion and the FES input depends on time varying muscle contractility or fatigue level. In this study, ultrasound speckle tracking is proposed to assess muscle contractility and to establish a dynamic model. To demonstrate the feasibility, isometric knee extension experiments of healthy human participants were conducted with ultrasound imaging on the quadriceps muscle. The consistent decrease in peaks in strain and maximum knee joint torque during each contraction cycle suggest a potential correlation between the strain field and fatigue level of the target muscle. With further validation, ultrasound strain field can be used to solve for the dynamic neuromuscular model and further to determine the optimal FES input. Some technical challenges will also be discussed.
Super-resolution ultrasound imaging for in vivo microvasculature assessment in acute kidney injury mouse model
Authors: Chen Qiyang , Rush Brittney M. , Yu Jaesok , Tan Roderick , Kim Kang
Published: April 2019

Acute kidney injury (AKI) is a disease with a high mortality rate and increasing incidence. It also generates a high risk of developing into chronic kidney disease (CKD). The deterioration from AKI to CKD is associated with rarefaction of microvasculature in renal cortex. However, there is lack of well-established diagnostic method that can evaluate the microvasculature changes noninvasively and conveniently with a high spatial resolution during the progression from AKI to CKD. Ultrasound super-solution imaging is an emerging technology that can achieve a high spatial resolution of the vasculature beyond the acoustic diffraction limit by localizing the center of the signals from microbubbles. In this study, deconvolution based super-resolution ultrasound imaging is used to noninvasively assess the microvasculature changes in mouse kidney after AKI. Ultrasound scans on mouse kidneys at 3 weeks and 6 weeks post ischemia-reperfusion injury are compared with control mouse kidneys. Obvious microvasculature reduction due to AKI is identified, which is evidenced by histology. The feasibility of ultrasound super-resolution imaging as a potential diagnostic method for progressive renal disease after AKI is demonstrated in the AKI mouse model.
A parametric evaluation of shear wave speeds estimated with the time-of-flight approach in viscoelastic media
Authors: Wiseman Luke M. , Urban Matthew W. , McGough Robert
Published: April 2019

The time-of-flight approach estimates the shear elasticity in tissue mimicking elastography phantoms and in soft tissue. The time-of-flight approach is effective in elastic phantoms, but the time-of-flight approach tends to overestimate the shear elasticity in viscoelastic phantoms and in viscoelastic soft tissues. To characterize errors in estimated parameters for different values of the shear elasticity and the shear viscosity, three-dimensional (3D) shear wave simulations are evaluated for twelve different parameter combinations. The 3D acoustic radiation force is calculated for an L7-4 transducer using the fast nearfield method and the angular spectrum approach, and then, 3D shear wave propagation in a viscoelastic medium is simulated with Green's functions for a Kelvin-Voigt model. The time-of-flight method is then evaluated within a two-dimensional plane. The results show that the accuracy of the time-of-flight method depends on the values of the shear elasticity and the shear viscosity. In particular, the error in the estimated shear elasticity increases as the shear viscosity increases, where the largest errors are observed when larger values of the shear viscosity are combined with smaller values of the shear elasticity. [Work supported in part by NIH Grants DK092255, EB023051, and EB012079.]
In situ calibration to account for transmission losses in backscatter coefficient estimation
Authors: Nguyen Trong , Tam Alex , Oelze Michael L.
Published: April 2019

The backscatter coefficient (BSC) has demonstrated the ability to classify disease state and identify the response of cancer to therapy. However, estimating the BSC in vivo using a reference phantom technique does not account for transmission losses due to intervening layers, leading to increase in bias and variance of BSC-based estimates from one sample to the next. To account for transmission losses, an in situ calibration approach is proposed using a titanium sphere that is well-characterized ultrasonically, biocompatible, and embedded inside the sample. Ultrasound scattered from the sphere encounters the same transmission loss and attenuation as the investigated sample and can be used as a reference spectrum. To test the calibration procedures, phantoms were scanned with and without lossy layers on top, and BSCs were estimated using the in situ calibration approach and the reference phantom approach and compared. The differences of the BSCs, using the BSC from the reference phantom without a layer as baseline, were 0.16 ± 2.29 dB, -1.95 ± 2.99 dB and -10.90 ± 3.64 dB using the in situ calibration approach without the layer, with the layer, and using the reference phantom approach with the layer, respectively. The results indicate that an in situ calibration target can account for overlaying tissue losses thereby improving the robustness of BSC-based estimates.
Detection of remotely piloted aircraft using bio-acoustic techniques
Authors: Fang Jian , Driscoll Michael , Brinkworth Russell , Finn Anthony
Published: April 2019

This paper describes a biologically inspired approach for acoustically detecting and tracking small remotely piloted aircraft based on processing found in the insect visual system. Previous work has shown the insect visual system is excellent at enhancing and isolating signals in complex and noisy visual scenes. By constructing spectrograms of audio signals, we essentially converted audio data into images, which could then be processed in the same way as visual data sets. Traditional time-frequency analysis was used to characterise the signatures of remotely piloted aircraft observed by multiple sets of small microphone arrays located on the ground. A model based on multiple layers of non-linear dynamic adaptive components measured from responses of insect visual neurons was then applied to the observed spectrograms to enhance the related acoustic harmonics and suppress the unrelated noise. The result was crisp low-amplitude signal detection and classification of these difficult target sets. In contrast to traditional systems that operate uniformly across the entire spectrum—attempting to capture the world as faithfully as possible—the bio-inspired processing uses multiple time scales and operates independently on each time-frequency cell ("pixel"). This reveals unseen harmonics otherwise hidden by noise, thereby extending the maximum range at which even slow-moving, low amplitude targets are detected and tracked.
Estimating multiple bearings-of-arrival from tornadic storms using the complex Wishart distribution
Authors: Frazier William G. , Talmadge Carrick L. , Hetzer Claus , Waxler Roger M.
Published: April 2019

Several array signal processing methods can be used to estimate bearings-of-arrival (BOA) in the presence of multiple infrasound sources, and their effectiveness depends upon several factors including array geometry, relative signal and noise power spectra, and noise cross-spectra. One of the most effective and computationally efficient methods is Multiple Signal Classification (MUSIC). However, MUSIC's performance not only degrades with the decreasing signal-to-noise ratio, as all methods, but also degrades as the noise model deviates from the assumption of uncorrelated, equal noise power on all channels. Uncorrelated, but unequal, noise power levels are a common situation with infrasound arrays, and the degradation of MUSIC's performance has been observed when estimating 2–10 Hz acoustic emissions from tornadic storms. This presentation examines the performance of formal maximum-likelihood estimation of multiple BOAs using the complex Wishart distribution as a model for the array's cross-spectral density matrix estimates. Estimation and computational performance comparisons with MUSIC are also reported.
Using acoustic waveform inversion to quantify volcanic emissions
Authors: Iezzi Alexandra M. , Fee David
Published: April 2019

Volcanic eruptions produce immense sound, particularly in the infrasound band. Acoustic waveform inversion shows promise for improved eruption characterization by providing robust estimates of erupted volume and mass. Previous inversion studies have generally assumed a simple volumetric acoustic source (monopole) that radiates sound equally in all directions. However, more complex and complete source reconstructions are possible with a combination of equivalent sources (multipole). Recent work has made progress using Finite-Difference Time-Domain modeling over high-resolution topography to obtain the full three-dimensional Green’s functions. The source-time function can then be inverted for and converted to a volume and mass flow rate. We review the acoustic waveform inversion as it has been applied to volcanic eruptions and discuss current limitations and how they can be mitigated. In most cases, the simple (monopole) source mechanism is a good approximation for discrete volcanic explosions, but a small directionality (dipole) component may remain. Furthermore, the neglecting effects of topography can lead to the overestimation of both the monopole and dipole strengths. Volcano infrasound source mechanisms are also not well constrained due to infrasound sensors usually being deployed on the surface. The methods discussed here can be extended to anthropogenic explosions and monitoring efforts, potentially in near-real time.
The effects of spatial and temporal frequency of meteorological data sampling on accurate prediction of infrasound propagation
Authors: Alter Ross E. , Swearingen Michelle E. , McKenna Mihan H. , Simpson Christopher , Quinn Brian G.
Published: April 2019

Accurately characterizing the environment is essential for robust predictions of infrasound propagation. In particular, meteorological data often exhibit large spatiotemporal variations; thus, accurate characterization can be challenging given the limitations of current measurement techniques. Though numerical weather models can provide realistic representations of the meteorological environment, the spatial and temporal sampling frequencies needed for accurate prediction of infrasound propagation are currently unknown. To address this issue, various simulations were conducted with the Weather Research and Forecasting (WRF) meteorological model, and the output was incorporated into a wide-angle parabolic equation method (PE) model to predict local infrasound propagation between 1 and 20 Hz. The sensitivity of infrasound propagation to spatial (1–15 km) and temporal (≤1 min) sampling frequency within the WRF model was then calculated. Furthermore, the results of these simulations were compared to calibrated field measurements of meteorological and infrasound data at ranges up to 15 km to identify which sampling frequencies and locations along the propagation path provided the most accurate propagation results. Next steps for applying these results to future infrasound simulations will also be discussed, particularly regarding the optimization of horizontal and vertical meteorological sampling. Distribution Statement A: Approved for public release; Distribution is unlimited.
Perceptual weighting of elevation localization cues across frequency
Authors: Ahrens Axel , Brimijoin Owen
Published: April 2019

Spectral cues are thought to be of particular importance in the perception of the elevation of a sound source. While some work has been done on demonstrating the importance of individual frequency bands, the relative importance of bands across a wide range of frequencies has not been firmly established. To estimate this, we built a broadband signal consisting of seven 1-ERB-wide noise bands that could each be assigned to a different elevation. The frequency range was either from 1 to 16 kHz with 3-ERB-wide spectral gaps or a higher-resolution range from 3 to 12 kHz with 1-ERB-wide spectral gaps. On each trial, each frequency band was independently convolved with a randomly chosen personalized head-related transfer function from one of seven elevations (±60 deg, 15 deg steps). In a 1-interval, 2-alternative forced choice task, listeners were asked to judge whether the sound was perceived above or below a reference stimulus presented on the horizontal plane. Two azimuth angles at -15 deg and -45 deg were considered. Perceptual weights for each frequency band were then calculated using a regression analysis method. Results showed that listeners tended to weight the 6.5 kHz band the highest for both azimuth directions and frequency resolution conditions.
Disentangling the contribution of head shadow, loudness summation, and binaural unmasking to spatial release from masking in children
Authors: Peng Z. Ellen , Litovsky Ruth
Published: April 2019

Segregating target speech from noise is crucial for children’s ability to communicate effectively in everyday environments. Past research clearly shows that when target sources are spatially separated from maskers, compared with target-masker being co-located, children as young as 2–3 years old demonstrate improved speech understanding. This effect is known as spatial release from masking (SRM). Generally, studies have used free-field or dichotic listening; hence, the contributions of head shadow, loudness summation, and binaural unmasking to SRM are unknown in children. This study aimed to quantify these factors in virtual auditory space. By varying the target-masker spatial configurations (co-located versus separated) and ear conditions (monaural versus binaural), speech understanding benefit was defined as improvement in the signal-to-noise ratio to achieve an accuracy of 50%. Results from 29 children with normal hearing (6–15 years old) show that head shadow cues are dominant in providing benefit, followed by binaural unmasking. Loudness summation, through increased intensity by listening with both ears, provided little to no benefit. No age effects were found. Results also suggest a re-balancing between cues depending on listening strategies adopted by children. For example, children who relied more on binaural unmasking received less benefit for speech understanding from head shadow. [Work supported by NIH-NIDCD.]
A deep learning based segregation algorithm to increase speech intelligibility for hearing-impaired listeners in reverberant-noisy conditions
Authors: Zhao Yan , Wang DeLiang , Johnson Eric , Healy Eric
Published: April 2019

Recently, deep learning based speech segregation has been shown to improve human speech intelligibility in noisy environments. However, one important factor not yet considered is room reverberation, which characterizes typical daily environments. The combination of reverberation and background noise can severely degrade speech intelligibility for hearing-impaired (HI) listeners. In the current study, a deep learning based time-frequency masking algorithm was proposed to address both room reverberation and background noise. Specifically, a deep neural network was trained to estimate the ideal ratio mask, where anechoic-clean speech was considered as the desired signal. Intelligibility testing was conducted under reverberant-noisy conditions with reverberation time T60 = 0.6 s, plus speech-shaped noise or babble noise at various signal-to-noise ratios. The experiments demonstrated that substantial speech intelligibility improvements were obtained for HI listeners. The algorithm was also somewhat beneficial for normal-hearing (NH) listeners. In addition, sentence intelligibility scores for HI listeners with algorithm processing approached or matched those of young-adult NH listeners without processing. The current study represents a step toward deploying deep learning algorithms to help the speech understanding of HI listeners in everyday conditions. [Work supported by NIH.]
Effects of age and hearing loss on spatial release from speech-on-speech masking, performance in envelope-based psychophysical tasks, and EEG envelope-following responses
Authors: Patro Chhayakanta , Klang Alix , Kreft Heather A. , Wojtczak Magdalena
Published: April 2019

Behavioral measures of amplitude modulation (AM) detection and envelope interaural-phase-difference (eIPD) detection reflect listeners’ ability to process temporal information. Robust encoding of temporal envelopes is necessary for understanding speech in a complex acoustic environment and for spatial segregation of a target speech from interfering background. It has been suggested that a large variability in performance in psychophysical tasks involving temporal envelope processing and in spatial release from masking for speech intelligibility may arise from cochlear synaptopathy. However, many studies have not found significant correlations between these measures and the amount of self-reported noise exposure in young listeners with audiometrically normal hearing. Similarly, electroencephalographic envelope-following responses did not significantly correlate with noise exposure or with behavioral performance reliant on envelope processing young normal-hearing population. In this study, behavioral measures in psychophysical tasks (AM and eIPD detection) and speech intelligibility in two-talker babble were measured for listeners with normal and near-normal hearing across a wide age range (20 to 69 years). Correlational analyses were performed using behavioral measures and envelope-following responses collected from the same listeners. Results will be discussed in terms of sensitivity of these measures to effects of aging and high-frequency hearing loss. [Work supported by NIH Grant R01 DC015987.]
Reverberation detection threshold estimates in normal-hearing listeners
Authors: Zahorik Pavel , Shehorn James
Published: April 2019

The study of human sensitivity to a single acoustic reflection (echo) has a long and rich history. The influence of time delay, level, direction, and source material are well documented. Unfortunately, real world listening seldom involves only a single reflection. Multiple reflections and reverberation are instead the norm. It is therefore surprising that the detection threshold for acoustical room effects (early reflections plus reverberation) has not been extensively studied, if at all. This study represents an initial step to fill this gap in knowledge. Using virtual auditory space techniques to simulate room acoustic sounds fields over headphones, the detection threshold for reflected/reverberant sound energy was measured for three sound field conditions: a small office-sized room (broadband T60 = 0.5 s), a concert hall (broadband T60 = 1.5 s), and a reference condition with a single echo at 40 degrees to the right of midline. The source signal was a 220 Hz complex tone, 250 ms in duration. Thresholds for the single-echo reference condition and the small room condition were found to be comparable, whereas the concert hall produced thresholds that were at least 20 dB lower. Temporal integration and binaural effects are considered as potential explanations for these results.
Selective deficits in a clinical population with self-reported hearing difficulties but normal audiometric thresholds
Authors: Shehorn James , Strelcyk Olaf , Zahorik Pavel
Published: April 2019

Audiologists often encounter patients who report hearing difficulties despite having normal audiometric thresholds. Many of these patients are told that they have normal hearing, although it is possible that the current typical audiometric test battery is not sensitive to these hearing difficulties. A test battery including immittance testing, binaural listening tasks, cognitive testing and subjective questionnaires of speech understanding, spatial hearing, and annual noise exposure was administered to 26 patients (age range: 18–53 years) who had sought out audiologic assessment at a community clinic but had normal hearing sensitivity. Despite normal hearing sensitivity, the participants in the study exhibited deficits in several binaural listening tasks, weakened middle-ear muscle reflexes, and rated their speech understanding and spatial hearing as being significantly worse than a normative population with normal hearing sensitivity. This patient group did not perform significantly worse on any of the cognitive measures nor did they report significantly more noise exposure than a normative young adult population; however, nearly two-thirds of participants were classified as “high risk” for noise exposure. These findings confirm self-reported hearing difficulties reported by this audiometrically “normal” population and suggest that several of the measures used in this study should be considered for standard audiologic evaluation.
Predicting speech-cue weighting in older people with impaired hearing
Authors: Gallun Frederick J. , Wright Richard , Souza Pamela
Published: April 2019

Previously published data (Souza et al., 2015; 2018) revealed individualized patterns of cue-weighting in the identification of synthetic speech ("bah," "dah," "lah," "wah"). The speech tokens were constructed such that temporal modulation (TM; envelope rise-time) and spectrotemporal modulation (STM; formant transitions) were systematically varied. Here, a discrimination task with the speech stimuli and a modulation detection task with noise tokens were related to cue weighting for thirty people aged 61–90 years with impaired hearing. Discrimination of STM in the speech stimuli was correlated with both STM sensitivity and cue weighting, while TM discrimination and the audiogram were not. The direct link between STM sensitivity and cue weighting was not supported, however. Results suggest that there was no substitute for actually measuring the ability to discriminate the speech stimuli on the basis of spectrotemporal information. The discrimination task may provide a more time-efficient measure of cue weighting than the cue identification task. These results have important implications for the rehabilitation of impaired hearing, which is generally guided only by the audiogram, which was the worst predictor of cue weighting tested. [Work supported by NIH R01 DC006014 (PI: Souza) and NIH R01 DC015051 (PI: Gallun).]
Effects of training on sensitivities to spatial changes in auditory scenes
Authors: Ball Natalie J. , Wisniewski Matthew , Simpson Brian , Mercado Eduardo
Published: April 2019

Previous work has shown that change deafness can occur with changes in the spatial location of objects within auditory scenes. Whether performance can be improved with training has yet to be directly tested. In the present study, the impact of training was examined in a “flicker”-like paradigm, where an initial scene comprising environmental sounds presented on the horizontal plane alternated with the presentation of a comparison scene, which was either the same or contained a change in the location of one or more sounds. Trained participants were trained on a set of sounds on day 1, while control subjects completed an unrelated visual task. On day 2, participants were tested using the same paradigm, with trained subjects hearing either the sounds they were trained on or new sounds, and controls being exposed to the paradigm for the first time. Overall, trained participants performed better (93% accuracy) than untrained participants (79%). Both groups had lower reaction times on correct-response change trials than on correct-response no-change trials. Trained subjects performed no better on trained sounds than new sounds. These data indicate that training can improve task performance, but improvements may be not be limited to sounds experienced during training.
Non-stationary source localization using direction of arrival measurements from multiple in-air vector sensors
Authors: Penhale Miles B. , Barnard Andrew R.
Published: April 2019

Vector sensors utilize a combination of pressure sensors, particle velocity sensors, or both, to determine the acoustic intensity magnitude and direction pointing toward an acoustic source. This acoustic intensity vector is often referred to as the Direction of Arrival (DOA). By combining DOA information from multiple vector sensor measurement locations, a sound source may be instantaneously localized. The majority of vector sensor research has been conducted for underwater applications. A few studies of in-air vector sensors, which utilize multiple microphones, have been conducted; however, the majority of them study stationary sound sources in a laboratory environment or non-real-world settings. The focus of this paper is to study in-air vector sensor capabilities when sensing non-stationary mechanical noise sources—specifically ground vehicles—in a non-laboratory environment where ambient noise may be present. The DOA measurements at multiple vector sensor locations are used to test the acoustic source localization potential for this method.
Ultrasonic communications for real-time video-rate data transmission through tissue
Authors: Tabak Gizem , Oelze Michael L. , Singer Andrew
Published: April 2019

Wireless implanted medical devices (IMDs), which communicate data wirelessly from sensors within the body to a receiver outside of the body, are poised to be significant contributors to medical diagnostics and treatment procedures. Currently, radio frequency (RF) electromagnetic waves are the most frequently used communication method for wireless IMDs. However, high attenuation of RF in the body and strict regulations on the RF frequency spectrum limit the data rates to 267 kbps for in vivo applications. Considering standard definition video streaming requires 1.2 Mbps, and HD requires greater than 3 Mbps, it is not possible to use RF communication, for example, in applications that require real-time video transmission and, possibly, intervention such as real-time video capsule endoscopy. In our work, we use ultrasonic waves to communicate through tissue at video-capable data rates (>1.2 Mbps). Previously, we demonstrated a 4 Mbps data rate with BER less than 1 10-4through beef liver using small, 2-mm biocompatible transducers at 1.3 MHz. In this study, we will investigate the effects of target motion on data rates and demonstrate real-time communication links in situ in a dead animal and in vivo.
Variability of the modal response of a shallow water acoustic waveguide in the presence of uncertain sediment properties
Authors: Martinelli Sheri , Wixom Andrew S. , Langhirt Mark , Holland Charles W.
Published: April 2019

Physical properties of the seabed comprise an important input set for physics-based modeling of sound propagation in littoral environments. Unfortunately this knowledge is often incomplete due to inherent space-time variability, yet deterministic models are still very much the standard for complex environments. Normal mode decomposition provides a well-established and understood framework for the study of underwater propagation. This work applies a generalized polynomial chaos expansion with stochastic collocation to propagate uncertain variables through a normal mode model, thus constructing expressions for the transmission loss (TL) and mode shapes themselves in terms of the input random variables. The goal of this work is to demonstrate the impact of imperfect knowledge of physical parameters on a deterministic computational model of the in-water acoustic field, and further study how well important information about model performance is captured by considering only first and second moments of the output distribution. An emphasis on material properties and geometry of sediment layering in the ocean bottom serves to isolate the effects of sediment uncertainty. Such a study can provide guidance for the use of deterministic models in performance prediction and has consequences for geo-acoustic inversion.
Current trends in binaural auralization of microphone array recordings
Authors: Ahrens Jens , Andersson Carl , Helmholz Hannes
Published: April 2019

Many approaches for the capture and auralization of real acoustic spaces have been proposed over the past century. Limited spatial resolution on the capture side has typically been the factor that caused compromises in the achievable authenticity of the auralization. Recent advancements in the field of microphone arrays provide new perspective particularly for headphone-based auralization. It has been shown that head-tracked binaural auralization of the data captured by a bowling-ball-sized spherical array of around 90 microphones allows for creating signals at the ears of the listener that are perceptually almost indistinguishable from the ear signals that arise in the original space. Promising results have also been obtained based on smaller arrays with fewer microphones. In the present contribution, we provide an overview of the current activities in the research community and demonstrate the latest advancements and remaining challenges.
Benchmark analytical solutions for steady-state high frequency broadband sound fields in three rectangular enclosures
Authors: Raudales David , Bliss Donald B. , Michalis Krista , Rouse Jerry W. , Franzoni Linda P.
Published: April 2019

Analytical solutions are presented for interior broadband sound fields in three rectangular enclosures with absorption applied on the floor and ceiling, rigid sidewalls, and a vertically oriented dipole source. The solutions are intended to serve as benchmarks that can be used to assess the performance of broadband techniques, particularly energy-based methods, in a relatively straightforward configuration with precisely specified boundary conditions. A broadband Helmholtz solution is developed using a frequency-by-frequency modal approach to determine the exact band averaged mean-square pressures along spatial trajectories within each enclosure. Due to the specific choice of enclosure configuration and absorption distribution, an approximate specular solution can be obtained through a summation of uncorrelated image sources. Comparisons between the band averaged Helmholtz solution and the uncorrelated image solution reveal excellent agreement for a wide range of absorption levels and improve the understanding of correlation effects in broadband sound fields. A boundary element solution with diffuse boundaries is also presented which produces consistently higher mean-square pressures in comparison with the specular solution, emphasizing the careful attention that must be placed on correctly modeling reflecting boundaries and demonstrating the errors that can result from assuming a Lambertian surface.
Acoustic microstreaming due to an oscillating contrast microbubbles near a substrate: Velocity, vorticity and closed streamlines
Authors: Mobadersany Nima , Sarkar Kausik
Published: April 2019

Intravenously injected microbubbles used as ultrasound contrast enhancing as well as drug delivery agents are encapsulated by a nanometer-thick layer of lipids, proteins, or polymers to stabilize them against premature dissolution. Here, acoustic microstreaming due to an oscillating microbubble, either coated or free, responsible for sonoporation and other bioeffects is analytically investigated. The detailed flow field is obtained, and the closed streamlines due to the ring vortex are plotted in both Eulerian and Lagrangian descriptions. Analytical expressions are found for the ring vortex showing that its length depends only on the separation of the microbubble from the wall and the dependence is linear. The circulation as a scalar measure of the vortex is computed quantitatively identifying its spatial location. The functional dependence of circulation on bubble separation and coating parameters was shown to be similar to that of the shear stress. [Work supported partially by NSF CBET 1602884 and GWU.]
Sounding out bacteria: Microstructural effects of therapeutic ultrasound on bacterial biofilms
Authors: Bharatula Lakshmi Deepika , Rice Scott , Marsili Enrico , Kwan James J.
Published: April 2019

Treatment of chronic infections due to formation of bacterial biofilms are a huge risk due to the growing concerns with antimicrobial resistance. Biofilms grow in a complex and dynamic environment that weaken the effect of antimicrobials. Yet, the current strategy to tackle the problem is the development of novel drugs. However, the increasing prevalence of antimicrobial resistance suggests that an alternative treatment strategy without, or in synergy with, antibiotics is necessary to combat the biofilm infections. We and others have proposed high intensity focused ultrasound (HIFU) as a means to disrupt the biofilm matrix and improve therapy. Yet to date, there is limited knowledge on the cellular activity triggered by the biofilm-acoustic interactions. Here, we report the effect of HIFU at 500 kHz center frequency in absence of antibiotics or microbubbles on the microstructure of biofilms formed by Pseudomonas aeruginosa. Changes to the biofilm after acoustic exposure were characterized by confocal microscopy and electrochemical impedance spectroscopy. We observed a drop in the biomass at pressures where non-linear acoustics were dominant, and an increase in cellular activity. Our results suggest that there are acoustic bio-effects present in these bacteria that have not yet been reported.
Rapid calculating of loudness according to ANSI S3.4-2007 with the Glasberg and Moore 2002 extension to time-varying signals in MATLAB
Authors: Swift S. Hales , Gee Kent L.
Published: April 2019

The ANSI S3.4-2007 standard gives a method for calculating the predicted loudness of stationary sounds for an average listener. Glasberg and Moore (2002) provide an extension of the method to time-varying sounds. The mathematical structure of the excitation in the loudness calculation is amenable to significant acceleration in MATLAB by expressing portions of the calculation, notably those representing the cochlear filtering process, in terms of matrices. Thus, procedures to achieve rapid processing of loudness are set forth. Possible extensions of this approach to other metrics within the same family are considered.