bluuebunny/crossref_metadata_2025 huggingface dataset:
Early acoustic hearing and spoken language skills of children with cochlear implants
Authors: Uchanski Rosalie M. , Davidson Lisa S.
Published: June 2017

Development of spoken language is difficult for children born with hearing loss. While most clinicians agree on the goal of improving the audibility of spoken language as early as possible, there is less agreement on the types of devices (bilateral cochlear implants vs. bilateral hearing aids vs. one hearing aid with one cochlear implant) they would recommend to achieve improved audibility. Additionally, acoustic properties of speech are conveyed differently by hearings aids (HAs) and cochlear implants (CIs); voice-pitch and prosodic properties, assumed critical for learning words from continuous speech, are conveyed better with HAs than CIs while broad spectral properties of individual speech segments are conveyed better with CIs than HAs. The relation between a simple model of a child’s early (birth to ~3 years old) acoustic hearing experience (includes HA use, CI surgery dates, severity of hearing loss, etc.) and eventual spoken language skills (tested at later ages of 8-10 years old) will be examined, especially in the context of which devices might be best for spoken language development. This examination reflects Dr. Louis Braida’s long-standing interest in understanding the acoustic properties of speech and its perception, especially for the benefit of those with hearing loss.
Automated extraction of information from Language ENvironment Analysis (LENA) home recordings of older children with autism
Authors: Clements Mark A. , Pawar Rahul , Caulley Desmond , Jones Rebecca , Lord Catherine
Published: June 2017

It has been established that children and adolescents with Autism Spectrum Disorder show a wide range of abilities to use spoken words and establish interactive conversations. Automatic measurement of such abilities in naturalistic environments would greatly facilitate assessment and monitoring of such individuals. If sufficient accuracy could be achieved, studies could be performed whose sample sizes are large enough to draw meaningful conclusions. In the current study, 16-hour home audio recordings using the LENA (Language ENvironment Analysis) device are examined for older children and adolescents. Specific enhancements to the existing LENA analysis platform include the ability to diarize recordings for subjects aged 5 through 13, to detect non-verbal vocalizations such as laughter and whining, to identify child-directed speech, and to determine when questions are posed. Other higher-level descriptors involve extraction of affect, computation of conversational interaction measures, detection of cross-talk and interruption events, and identifying emotional outbursts. Subject-specific diarization based on a small amount of hand-labeled data yields acceptable accuracy. However, a newly developed system based on i-vectors, specifically designed for the environment at hand, requires no such labeling at the onset.
The auditory-visual articulation Index
Authors: Grant Ken W. , Bernstein Joshua G.
Published: June 2017

Hearing aids (HAs) are the primary method for treating hearing impairment. However, in complex environments with competing sound sources, HAs provide marginal benefits at best. Under these conditions, clinicians recommend facing the speaker to extract visual speech information. Combined auditory-visual (AV) speech generally provides a signal that is much more resistant to noise and reverberation than an auditory-only signal. The Articulation Index (AI) established that different frequency regions of speech vary in their degree of importance for intelligibility. However, frequencies most important for auditory-only speech intelligibility differ from those that are most important for AV speech intelligibility. Thus, the optimal signal-processing solution may differ between AV and auditory-only conditions. Braida and colleagues sought to develop an AV version of the AI to enable HA signal-processing strategies to be compared without the time and expense required for behavioral testing. This presentation describes this and other work inspired by Braida’s efforts to predict AV speech intelligibility using only auditory-only and visual-only information. [Work supported by a grant from CDMRP, #DM130027. The views expressed in this abstract are those of the authors and do not reflect the official policy of the Department of Army/Navy/Air Force, Department of Defense, or U.S. Government.]
Contributions of Louis Braida to improved signal processing for hearing aids: Addressing the problem of reduced dynamic range in listeners with sensorineural hearing loss
Authors: Reed Charlotte M. , Desloge Joseph G. , D'Aquila Laura A.
Published: June 2017

Lou’s early work in the area of improved signal processing for hearing aids included his research on compression amplification to combat the effects of loudness recruitment in listeners with sensorineural hearing loss. Working with his doctoral students (including Rich Lippmann, Steve De Gennaro, and Diane Bustamante), Lou made major contributions towards an analytical understanding of the benefits and limitations of compression amplification as a component of hearing aids. Recently, Lou has been involved in work on a new signal-processing scheme which operates to equalize the energy in a speech signal over time. This energy-equalization (EEQ) scheme shares a similar goal with compression amplification in that they both attempt to match the range of speech levels into the reduced dynamic range of a listener with sensorineural loss. Their operation, however, is different: while compression amplification is based on the actual sound-pressure level of the signal, the EEQ scheme operates on relative energy calculations designed to reduce the variations in overall signal level. In this talk, we will describe the EEQ processing system together with results obtained on its evaluation with hearing-impaired listeners for speech reception in backgrounds of continuous and fluctuating noise.
Validating a perceptual distraction model in a personal two-zone sound system
Authors: Rämö Jussi , Christensen Lasse , Bech Søren , Jensen Søren H.
Published: June 2017

This paper focuses on validating a perceptual distraction model, which aims to predict user’s perceived distraction caused by audio-on-audio interference, e.g., two competing audio sources within the same listening space. Originally, the distraction model was trained with music-on-music stimuli using a simple loudspeaker setup, consisting of only two loudspeakers, one for the target sound source and the other for the interfering sound source. Recently, the model was successfully validated in a complex personal sound-zone system with speech-on-music stimuli. Second round of validations were conducted by physically altering the sound-zone system and running a set of new listening experiments utilizing two sound zones within the sound-zone system. Thus, validating the model using a different sound-zone system with both speech-on-music and music-on-speech stimuli sets. Preliminary results show that the model performance is equally good in both zones, i.e., with both speech-on-music and music-on-speech stimuli, and comparable to the previous validation round (RMSE approximately 10%). The results further confirm that the distraction model can be used as a valuable tool in evaluating and optimizing the performance of personal sound-zone systems.
Sound environment and sound field reproduction using transducer arrays: Correlation between physical and perceptual evaluations
Authors: Gauthier Philippe-Aubert , Berry Alain
Published: June 2017

Sound Environment Reproduction (SER) using Sound Field Reproduction (SFR) is aimed at the spatial reconstruction of a target sound field captured using a microphone array. SFR has recently gained attention for SER in industrial or engineering contexts for sound comfort or sound quality studies. The challenge is to create a reproduced sound field that first satisfies an assessment based on physical evaluation, for example to satisfy any regulation based on physical quantities. However, the reproduced sound environment should also success in perceptual evaluation. In this work, SER was applied to spatial sound field simulation in a vehicle mock-up. Both physical and perceptual evaluations were completed. Physical metrics such as frequency-dependent averaged reproduction error (both phase and magnitude) and averaged magnitude error (ignoring phase) were measured. Perceptual evaluations were based on similarity listening tests while comparing SER with an original reference (the target sound field). Perceptual evaluations were compiled as similarity scores. Correlation of similarity scores based on various physical evaluations suggests that the frequency-averaged and spatially averaged magnitude error is the physical evaluation metric that is the most correlated with the result of listening tests. This suggests that spatially reproducing the accurate frequency spectrum is the first criterion for immersing SER.
A study of complex system dynamics: Metronome synchronization
Authors: Sonne Noah A. , Ryan Teresa J.
Published: June 2017

This work investigates energy exchange within a complex vibrating system. The system is made up of a mass, called the primary oscillator, and a number of attached smaller structures, called the subordinate oscillator array. Specifically, a rectangular rigid foam base is the primary mass and mechanical metronomes are used as the subordinate oscillators. This work explores how the orientation and arrangement of the metronomes on the master structure affects the time it takes for metronome synchronization as well as the resulting amplitude of oscillation of the vibration of the primary mass. A MATLAB based image processing approach is used to measure these system parameter. [This work was supported by the Robert W. Young Award for Undergraduate Student Research.]
Structural-acoustic optimization using cluster computing
Authors: Campbell Robert , Shepherd Micah R. , Hambric Stephen
Published: June 2017

Structural-acoustic optimization using state-of-the-art evolutionary algorithms may require tens of thousands of system solutions, which can be time-limiting for full-scale systems. To reduce the time required for each function evaluation, parallel processing techniques are used to solve the system in a highly-scalable fashion. The system acoustic radiation is modeled as a stochastic problem using finite elements for the structural vibration and boundary elements for the fluid loading and acoustic analysis. The approach is demonstrated by minimizing the sound radiated from curved panel under the influence of a turbulent boundary layer in the presence of added point masses. Details of the point mass magnitudes and distribution are outcomes of the optimization. Solver scaling information is provided that demonstrates the utility of the parallel processing approach.
Dissipation as energy transport to molecular scale
Authors: Akay Adnan
Published: June 2017

Dissipation describes irreversible transfer of ordered kinetic energy from a larger-scale to thermalized vibrations at the molecular scale. Considering dissipation as transfer of energy from one form of vibrations to another form, models can be developed without the need for qualitative empirical constants. Examples of “lossless” damping mechanisms will be derived to illustrate energy conversion at the molecular scale.
Second language pronunciation training using acoustic-to-articulatory inversion
Authors: Berry Jeffrey J. , Stoll Abigail , Jones Deriq , Alikiaamiri Seyedramin , Johnson Michael T.
Published: June 2017

The current work presents articulatory kinematic, acoustic, and perceptual data characterizing the effects of how visual biofeedback derived from acoustic-to-articulatory inversion may influence vowel pronunciation training for native-Mandarin speakers of English. Ten participants were engaged in a six-week pronunciation training program that included a focus on English vowel production. As an addition to traditional pronunciation training techniques often used by speech-language pathologists, half of the participants were also provided with visual biofeedback displays detailing aspects of their current tongue position as well as idealized positions for vowel targets. Visual displays were obtained using an acoustic-to-articulatory inversion model based on the Parallel Reference Speaker Weighting (PRSW) method for model adaptation. Pre- and post-training changes in articulation were compared between participants that used only traditional pronunciation training methods and those who were given visual biofeedback based on acoustic-to-articulatory inversion. Data analyses focus on articulatory-kinematic measures, obtained via electromagnetic articulography, measures of vowel formant frequencies, and perceptual assessments based on phonetic transcriptions from expert listeners. The results of the current work provide insights regarding the value of PRSW-based adaptation of acoustic-to-articulatory inversion models and the resulting visual feedback displays as tools in second-language learning of English pronunciation.
Interaction of drift and distinctiveness in L1 English-L2 Japanese learners
Authors: Huffman Marie K. , Schuhmann Katharina , Keller Kayla , Chen Chanda
Published: June 2017

Dynamic L2 effects on L1 phonetics appear in experienced and novice second language learners, raising the question of what linguistic and cognitive factors determine their occurrence, degree, and direction (assimilatory versus dissimilatory). Unlike Chang (2012), our longitudinal data from voiceless stops in early L1 English:L2 Japanese learners show primarily dissimilatory increase in English VOTs, an effect found most strongly early in their first semester. Assimilatory L1 drift toward the lower VOTs of L2 Japanese may be disfavored because a decrease in English voiceless stop VOT could threaten the L1 contrast between long and short lag stops. Furthermore, dissimilatory VOT increase on voiceless stops allows English speakers to distinguish phonetically similar L1 and L2 voiceless stops (e.g., Flege and Eefting 1987). These two principles predict that our voiced stop data could show increased L1 prevoicing (an assimilatory effect that would not endanger the English voicing contrast), while also displaying non-identical prevoicing/short-lag values for L1 and L2 voiced stops (separating the languages, as for Huffman and Schuhmann’s (2016) English-Spanish learners). Overall, our data suggest that L1 changes in early L2 learning can be dissimilatory, and that phonetic properties of L1 and L2 contrasts affect how L1 values restructure during early L2 acquisition.
Is it blow or below? Non-native listeners' perception of words that contrast in syllable count
Authors: Tajima Keiichi , Shattuck-Hufnagel Stefanie
Published: June 2017

Non-native speakers often have difficulty accurately producing and perceiving the syllable structure of a second language. For example, Japanese learners of English often insert epenthetic vowels when producing English words, e.g., stress produced as /sutoresu/. Similarly, when asked to count syllables in spoken English words, they frequently overestimate the number of syllables, suggesting that they tend to perceptually insert epenthetic vowels between adjacent consonants. These tendencies suggest the possibility that learners may have difficulty distinguishing between English words that contrast in syllable count, i.e., words that differ in the presence/absence of a vowel, e.g., blow-below, sport-support. Furthermore, if listeners perceptually insert epenthetic vowels, then they should misperceive blow as below more often than below as blow. To test these predictions, Japanese listeners participated in a 2AFC identification task, using 78 English minimal pairs contrasting in syllable count such as blow-below. Results showed that Japanese listeners indeed had difficulty with this task. However, misidentification of blow-type words as below was less frequent than misidentification of below-type words as blow, contrary to predictions based on perceptual epenthesis. These results suggest that simple comparison of syllable structure between languages may not suffice to predict difficulties in L2 speech perception. [Work supported by JSPS.]
Temporal-order processing of American-English vowel sequences by native and non-native English-speaking listeners
Authors: Rogers Catherine L. , Cheon Bogyeong , Donaldson Gail
Published: June 2017

To understand the development of native-like proficiency in speech processing, we must consider the apparent ease with which native speakers process speech sounds under a variety of conditions. In the present study, auditory temporal-order processing of American-English vowel sequences was compared across three listener groups: monolingual English speakers and relatively early vs. later learners of English as a second language. Using the methods of Fogerty, Humes and Kewley-Port [2010, J. Acoust. Soc. Am., 127, 2509-2520], 70-ms resynthesized versions of the syllables “pit, pet, put,” and “pot” were presented in a two-syllable temporal-order processing task. Task difficulty was increased by decreasing syllable-onset asynchrony (SOA), i.e., the duration between syllable onsets. SOA thresholds for accuracy of syllable-sequence identification were estimated using the method of constant stimuli on each of four 72-trial blocks. Similar SOA thresholds were obtained for native English speakers and early learners of English, but SOA thresholds increased by a factor of two or more for later learners of English. Furthermore, the average SOA threshold of the later learners is similar to that of the older listeners in Fogerty et al. (2010), suggesting that increased processing time partially accounts for both groups’ increased difficulty in processing speech in noisy environments.
Effects of experience on the production of Korean stop consonant contrasts by Mandarin Chinese learners
Authors: Oh Grace E.
Published: June 2017

The effect of L2 experience on the segmental and prosodic production of second language was investigated. Thirty two Chinese learners of Korean varying in the amount of experienced (3 months vs. 2 years) were compared to sixteen age-matched native Korean speakers in their production of three-way contrastive stops (aspirated, lenis, and fortis) in Korean. To examine both segmental and prosodic aspects, Korean four-syllable phrases (i.e., Accentual Phrase) beginning with each stop type in word-initial position were elicited. VOT, f0, H1-H2 were analyzed and compared across groups (native Korean, experienced, inexperienced groups). For further analyses of the prosodic domain, f0 of the first two syllables (High-High for aspirated, fortis and Low-High for lenis) was compared. The results revealed that the experienced Chinese learners showed an early mastery of fortis stops, producing more native-like VOT and H1-H2 than the inexperienced group. Also, Chinese groups were able to produce a sequence of AP with a native-like f0 pattern regardless of the amount of experience. The production of f0 for stop contrasts, on the other hand, was non-native like even after 2 years of experience, supporting previous observations that segments require greater L2 experience than intonation to be acquired in a native-like manner.
Overweighting of pitch cues in stop identification by Korean learners of Mandarin Chinese
Authors: Lee-Kim Sang-Im
Published: June 2017

The present study reports on a novel case where learners of a tonal language not only develop keen sensitivity to F0 cues in general, but the increased sensitivity may also foster perceptual reorganization of cue weighting in stop identification. Korean learners of Mandarin and novice listeners participated in identification tasks for which pitch contours of Mandarin words containing unaspirated stops were digitally manipulated. In word-initial position, learners showed considerably higher sensitivity to onset pitch cues, showing a near-categorical perception from lenis to fortis judgment along with higher onset pitch. In word-medial position, tone contours emerged as a significant predictor of the stops; predominant fortis judgment in high-level tone vs. lenis judgment in mid-rising tone. The novice listeners showed similar patterns in both cases, but to a much smaller degree. The effect of tone contours was discussed with reference to an auditory contrast whereby onset pitch of mid-rising tone is perceptually lower, providing positive evidence for lenis stops, due to the contrast with the following high pitch. Taken together, the learners’ differential behavior suggests substantial reorganization of perceptual cues whereby onset pitch is promoted to a primary cue for the lenis-fortis contrast, concurrent with significant underweighting of VOT cues.
The effect of noise and second language on turn taking in task-oriented dialog
Authors: Sørensen A Josefine , Fereczkowski Michal , MacDonald Ewen N.
Published: June 2017

Previous studies of floor transfer offsets (FTO), the interval between one talker stopping and the other starting, suggest that normal conversation requires interlocutors to predict when each other will finish their turn. We hypothesized that noise and/or speaking in a second language (L2) would result in longer FTOs due to increased processing demands. Conversations from 20 pairs of normal hearing, native-Danish talkers were elicited using the Diapix task in four conditions consisting of combinations of language (Danish vs. English) and noise background (quiet vs. ICRA 7 noise presented at 70 dBA). Overall, participants took longer to complete the task in both noise and in L2 indicating that both factors reduced communication efficiency. However, L2 had very little effect beyond completion time, likely because the participants were very good in English. In contrast to our predictions, in the presence of noise, the median of the FTO distribution decreased by approximately 30ms and the standard deviation decreased by approximately 10%. However, the average duration of interpausal units (i.e., utterances of continuous speech) increased by 40% in noise. These findings are consistent with talkers holding their turn for longer, allowing more time for speech planning.
The power of a unimodal distribution in cue reweighting: Unimodality vs prediction error as signs of cue irrelevance
Authors: Harmon Zara , Idemaru Kaori , Kapatsinski Vsevolod
Published: June 2017

Maye & Gerken (2000) proposed that sound categories can be learned from probability distributions: a unimodal distribution suggests a single category, while a bimodal one suggests two contrasting ones. Research on distributional learning has focused on developing a contrast through exposure to a bimodal distribution. Here, we instead investigate how exposure to a unimodal distribution affects perception of a pre-existing multidimensional contrast (voicing, for which the primary cue is VOT). A total of 60 adult native English speakers were exposed to either bimodal or unimodal VOT distributions spanning the unaspirated/aspirated boundary (bear/pear). However, we paired acoustic stimuli with pictures of bears and pears independently of VOT in training. For each stimulus, participants were asked to guess the referent and received (random) feedback, generating an error signal that suggested VOT is no longer informative and should be downweighed. In this design, the bimodal distribution suggests the existence of two categories but provides a clearer error signal: in the unimodal condition, most training tokens have ambiguous VOT, preventing clear predictions of voicing, thereby reducing prediction error. Nonetheless, participants downweighed VOT (and upweighed a secondary cue, F0) only with unimodal training. We conclude that unimodality is a very strong cue to dimensional irrelevance.
Bayesian model selection for multilayer microperforated panel sound absorber design
Authors: Fackler Cameron J. , Hou Yiqiao , Xiang Ning
Published: June 2017

Microperforated panel (MPP) sound absorbers are capable of providing high sound absorption coefficients without the use of fibrous materials; however, they typically function in narrow frequency ranges. By combining multiple MPPs into a multilayer absorber, the frequency bandwidth may be increased while maintaining a high absorption coefficient. Modeling the acoustic properties of an MPP absorber requires four physical parameters per MPP layer. Since each additional MPP layer in a multilayer absorber increases the complexity of the acoustic model, Bayesian model selection is well-suited to the task of designing a multilayer MPP absorber. In such a design, minimizing the number of layers used while still satisfying the design goals is desirable, in order to optimize material usage, cost, and space required by the absorber. In a full Bayesian design framework, model selection determines the number of MPP layers required, while parameter estimation determines the (physical) design parameters for each layer. In this work, an example design scheme is specified to satisfy a practical need for acoustic absorption. The Bayesian framework produces a three-layer MPP design which meets the target requirements. This absorber design is constructed, and impedance tube measurements are obtained to validate the acoustic absorption properties.
Minimum mean square error estimation of a sparse broadband acoustic response with a hierarchical mixture Gaussian prior
Authors: Gendron Paul J. , Silva Jacob L.
Published: June 2017

Considered here is a minimum mean square error (MMSE) estimator of a broadband acoustic response function over a vertical aperture based on an adaptive sparsity prior. The prior is a hierarchical Gaussian mixture distribution built on the assumption that acoustic paths can be partitioned into a relatively coherent set of arrivals that on average exhibit Doppler spreading about a mean rate and a set of incoherent paths that exhibit a flat Doppler spectra. The hierarchy establishes constraints on the parameters of each of these Gaussian models such that coherent components of the response are both sparse and in the ensemble obey the Doppler spread profile. An empirical Bayes approach is developed to estimate the latent parameters of the hierarchy, from which the shared time varying dilation process can ameliorated thereby enhancing coherent multi-path combining. The model is tested with acoustic communication recordings taken in shallow water at very low signal-to-noise ratios.
Inference on speed, depth, and range of a submerged object from a limited vertical aperture under uncertain noise power
Authors: Barros Abner C. , Anchieta David C. , Gendron Paul J.
Published: June 2017

Considered here is an narrow band directed source and hydrophone receiver arrangement employed to infer the depth, speed, and range of an oncoming submerged object. Tracking the scattering body by means of a continuous wave transmission is challenging due to the difficulty of inferring the frequencies and angles of the two returned closely spaced wave vectors. Computation of the posterior pdf of these two wave vectors is accomplished by a judicious Gibbs sampling scheme that accounts for the uncertainty in the ambient acoustic noise level. Computational improvements are accomplished by taking full advantage of the prior distribution of the wave vectors associated with the specific target scenario. Very short duration observations of approximately 10 milliseconds are considered over which the Doppler rate of change of the two wave vectors can be considered negligible. This Bayesian scheme takes advantage of the analytic tractability of the conditional density of the received amplitudes and phases and of the noise powers. The conditional densities of the ordered wave vectors however are constructed numerically by 2 dimensional inverse quantile sampling. The inferred joint density of depth, range, and speed of the target is accomplished by constructing an inverse transformation of the acoustic propagation model.
Machine learning applied to estimating broadband source signature characteristics in a shallow ocean environment
Authors: Knobles David P. , Badiey Mohsen
Published: June 2017

Probabilistic machine and deep learning methods are critical elements in the area of automation and artificial intelligence. The regression task considered here is to use supervised machine learning to predict a frequency dependent acoustic source level from received broadband signals that have propagated through a shallow ocean waveguide possessing random features. The details are intimately linked to a previously introduced sequential maximum entropy—Bayesian approach employed to generate marginal probability distributions for both environmental and source parameter values. To meet the requirement to have both low training and generalization errors, several regularization and non-linear optimization methods are considered to enhance performance. This includes convolutional networks. Optimization includes not only the issue of sampling but also finding an effective model capacity, which is one of the most significant challenges to successful machine learning applications. The methodology is applied to measured broadband data collected in about 75 m of water about 60 miles south of Cape Cod Massachusetts in an area called the mud pond. Both the training samples and the samples not used in training have known source levels with which to measure both the training error and the generalization error and thus quantify performance.
Clustering technology for the analysis and classification of bioacoustic vocalizations
Authors: Agranat Ian
Published: June 2017

A new technique for clustering and classifying bioacoustics vocalizations from large audio recording data is presented. Over 30,000 terrestrial passive audio and ultrasonic recorders deployed worldwide for the monitoring of birds, frogs and bats generate many petabytes of data annually. Our methods seek to efficiently and automatically detect candidate vocalizations from these massive datasets and sort them into clusters of similar vocalization patterns. Researchers can then quickly survey biodiversity and find species of interest by examining each of the resulting clusters. After clustering, species-specific classifiers can be built by labeling whole clusters or training classifiers after labeling individual detections. Once a classifier is built, it can be used to search new data for species of interest. The algorithms detect vocalizations comprising one or more short syllables. A variable-length observation sequence is extracted comprising spectral features changing through time. A Hidden Markov Model is trained on these sequences and Fisher scores are calculated for each detection. The dot product of two Fisher scores provides a measure of similarity between two vocalizations and forms the basis for clustering. The algorithms are designed to run in parallel on multi-core processors and stream the data to minimize memory requirements.
Environmental indicators from underwater soundscapes and simultaneously collected non-acoustic data
Authors: Freeman Simon E. , Freeman Lauren A.
Published: June 2017

Drawing environmental conclusions from underwater acoustic recordings alone can to be challenging as most received sounds, especially in shallow water, are from sources that cannot easily be classified. This presentation will discuss two ongoing projects in which soundscapes were recorded simultaneously with non-acoustic, validating environmental data. In one case, acoustic data were collected simultaneously with bottom composition, organism censuses, and night-time time-lapse imagery over shallow reefs in the Hawaiian Islands. Multivariate analysis showed that areas defined by a “cool tropics” oceanographic regime grouped along a principal component parallel to monotonically increasing acoustic frequency: protected or more remote sites produced soundscapes that featured greater levels of low frequency (<2 kHz) biological sound. Degraded and/or algae dominated sites produced soundscapes featuring higher levels of high frequency (2-20 kHz) sound. The second case involves experimental work using hyperspectral optical data and simultaneously obtained acoustic recordings in shallow water, combined with in-situ spectrophotometer readings and visual benthic surveys. Optical reflectance and sound are produced through completely different mechanisms, yet the underlying environmental phenomena we wish to evaluate exhibit both. This presentation will discuss the potential of fusing these data in order to enhance our understanding of the shallow water environment. [Work supported by ONR.]
A wind-driven noise model in deep water
Authors: Li Fenghua , Xu Dong , Guo Yonggang
Published: June 2017

Surface processes, including non-linear surface wave interactions and bubble oscillations, dominate the deep water ambient noise in the absence of shipping and wildlife. A single-parameter noise model, determined by the wind speed above the ocean surface, is proposed to interpret the deep ocean acoustics from 1 Hz to 4 kHz. Below hundreds of hertz, the sound is primarily generated by surface wave interactions, which is the function of surface wave spectrum. A theoretical expression of wave spectrum is proposed to contain the gravity and capillary waves, which is in accord with the features of measured wave spectra. The noise spectrum for frequency above hundreds of hertz due to the effective bubble oscillation within the bubble cloud is proposed. The noise spectrum has been estimated as a function of frequency and wind speed based on available information of the bubble distribution. The model/data comparison shows that the proposed single-parameter noise model are in reasonable agreement with the data. [Work supported by National Natural Science Foundation of China, Grant No. 11125420.]
Modeling on low-frequency underwater noise radiated from a typical fishing boat based on measurement in shallow water
Authors: Peng Zilong , Wang Bin , Fan Jun , Zhao Kaiqi
Published: June 2017

As evidenced in documents during the past decades, the impact of man-made underwater noise on the marine environment has always attracted more and more interest of the global researchers. About ten kinds of models have been proposed to predict underwater radiated noise (URN) by ships, and most of which are applicable above 100 Hz. This paper is aimed to modeling the low-frequency URN. Extensive measurements were made of the URN of a small fishing boat (length 43 m, displacement 500 tons) at South China Sea. The URN data show the high-level noise below 100 Hz is mainly contributed by the mechanical noise (e.g., main engine and service diesel generator) and propeller cavitation, and performs complex varying characteristics with the speed. The effect on the Transmission Loss (TL) from the sound-speed profile and bottom has been analyzed, compared with the empirical function, which show the estimated TL has an important influence on the spectral source levels (SSLs). Inspired by the method in AQUO (Achieve QUieter Oceans) project, a predicted model applied to typical fishing boat was built.
Estimation of the entropy of seismic ambient noise: Application to passive imaging
Authors: Seydoux Leonard , Shapiro Nikolai , de Rosny Julien
Published: June 2017

Recovering Green’s functions from diffuse ambient noise correlation is an efficient technique for passive seismic imaging. However, the field diffuseness is not completely fulfilled in practice because the noise is generated in preferential areas and contaminated by some highly coherent signals due to earthquakes, for instance. Here we show that the noise entropy is a robust estimator of the diffuseness. In the first part, we directly use the entropy as a metric of seismic noise activity to detect tremors around Piton de la Fournaise Volcano or highly coherent noise sources at continental scale using data collected by USArray seismic array. Then in the second part, we take the benefit of the entropy to propose an original equalization process based on the analysis of the covariance matrix to mitigate the effect a poorly diffuse noise field. The efficiency of the method is validated with several numerical tests. We apply the method to the data collected by the USArray, when a strong earthquake occurred. The method shows a clear improvement compared with the classical equalization to attenuate the highly energetic and coherent waves incoming from the earthquake, and allows to perform reliable travel-time measurement.
Environmental seismology: What can we learn from ambient noise?
Authors: Larose Eric
Published: June 2017

Environmental seismology consists in studying the coupling between the solid Earth and the cryosphere, or the hydrosphere, the anthropospher. In practice, we monitor the modifications of the wave propagation due to environmental forcing such as temperature and hydrology, using ambient seismic noise that constitute a continuous, cheap and relatively reproducible source of vibrations. Recent developments in data processing [1], together with increasing computational power and sensor concentration have led to original observations that allow for this new field of seismology. In this paper, we will review how we can track and interpret tiny changes in the subsurface of the Earth related to external changes from modifications of the seismic wave propagation, with application to geomechanics, hydrology, and natural hazard [2]. We will demonstrate that, using ambient noise, we can track: thermal variations in the subsoil, in buildings or in rock columns with application to damage estimation; the temporal and spatial evolution of a water table; the evolution of the rigidity of the soil constituting a landslide, and especially the drop of rigidity preceding a failure event. [1] Shapiro, N. M., Campillo, M., 2004. Geophys. Res. Lett. 31, L7 614. [2] E. Larose et al.: J. Appl. Geophys. 116, 62-74 (2015).
Relocating drifting sensor networks with ambient noise correlations
Authors: Nichols Brendan , Martin James S. , Verlinden Christopher M. , Sabra Karim G.
Published: June 2017

A network of drifting sensors, such as hydrophones mounted to freely drifting buoys, can be used as an array for locating acoustic sources underwater. However, for accurate localization of such a source using coherent processing, the positions of the sensors need to be known to a high degree of accuracy, typically more accurately than provided by dead reckoning or GPS alone. Past work has demonstrated the inter-sensor distances can be obtained from long-term ambient noise correlations on fixed arrays [Sabra et al., IEEE J. Ocean Engineering, 2005, 30]. Here, the approach was extended for tracking drifting sensor motion by combining a stochastic search algorithm with ambient noise correlation processing. Optimization of the stochastic search method was explored and performance compared to acoustic data collected from a volumetric hydrophone vs. vector sensor array deployed in the Long Island Sound.
Interferometric reconstruction of plate waves from cross correlation of diffuse field on a thin aluminum plate
Authors: Hejazi Nooghabi Aida , de Rosny Julien , Boschi Lapo , Roux Philippe
Published: June 2017

This study contributes to evaluating the robustness and accuracy of Green's function (GF) reconstruction by cross-correlation of noise, disentangling the respective roles of ballistic and reverberated (“coda”) signals. We conduct a suite of experiments on a highly reverberating thin aluminum plate, where we generate an approximately diffuse flexural wavefield. We validate ambient-noise theory by comparing cross correlation to the directly measured Green’s function. We develop analytically a theoretical model, predicting the dependence of the symmetry of the cross correlations on the number of sources and signal-to-noise ratio. We validate this model against experimental results. We next study the effects of cross-correlating our data over time windows of variable length, possibly very short, and taken at different points in the coda of recordings. We find that, even so, a relatively dense/uniform source distribution could result in a good estimate of the GF; we demonstrate that this window does not have to include the direct-arrival signal for the estimated GF to be a good approximation of the exact one. Afterwards, we explicitly study the role of non-deterministic noise on cross correlations and establish a model which confirms that the relative effect of noise is stronger when the late coda is cross-correlated.
Robust multiple focusing with adaptive time-reversal mirror using a genetic algorithm
Authors: Byun Gi Hoon , kim Jea Soo
Published: June 2017

Kim and Shin [J. Acoust. Soc. Am. 115 (2), 600-606 (2003)] suggested an extension of the single constraint adaptive time-reversal mirror (ATRM) to simultaneous multiple focusing, by considering multiple constraints. In their proposed method, the optimization is performed using the linearly constrained minimum variance (LCMV) method, a well known optimization method in the field of adaptive signal processing that allows multiple linear constraints. However, highly correlated signal vectors from the probe source locations cause prominent spatial sidelobes in multiple TR focusing. In this study, a genetic algorithm is extended to LCMV method to calculate the backpropagation vector which satisfies new constraint responses. Numerical simulations demonstrate that multiple TR focusing combined with a genetic algorithm can significantly suppress sidelobes, especially when the focal locations are close to each other.
Modeling method on acoustic scattering from penetrable objects using a hybrid Kirchhoff/ray approach
Authors: Wang Bin , Zhao Kaiqi , Fan Jun , Zheng Guoyin
Published: June 2017

A method is put forward to investigate the acoustic scattering from double-sided water loaded targets which is penetrable even in high frequency. The present approach is an extension of the TriKirch method which was elaborated for non-penetrable targets [J. Acoust. Soc. Am. 140(3), 1878-1886 (2016)] and complex targets are dispersed into many triangular planar facets. Reflection coefficient of plane facet is introduced to calculate the scattering amplitude of each non-rigid triangular facets combined with TirKirch method, and the total scattering amplitude of target is obtained by superposing the amplitude of all facets’ contributions coherently. Double bounce (DB) contributions to the scattering is calculating by ray-tracing method. Computations are made for a double-sided water loaded finite cylindrical shell with perforated ring ribs, and are compared with the experimental results，and both results show good agreement.
Mode coherence in random matrix theory simulations
Authors: Wage Kathleen E. , Van Uffelen Lora J.
Published: June 2017

Random matrix theory (RMT) can be used to simulate the effect of internal waves on broadband acoustic mode time series in deep water, as described by Hegewisch and Tomsovic in several papers [Europhys. Lett. 2012; J. Acoust. Soc. Am. 2013]. Using RMT, narrowband mode propagation consists of the multiplication of a series of propagator matrices designed to model the mode coupling due to internal waves at a single frequency. In a recent paper, we varied the correlation of the mode coupling matrices with frequency and examined the effect on the time series generated via Fourier synthesis [Van Uffelen & Wage, Inst. of Acoustics Conf., 2016]. This talk focuses on another key aspect of the RMT model, i.e., mode-to-mode coherence at a single frequency. Modes decorrelate as they propagate through internal waves. Building on Creamer's work, Colosi and Morozov use transport theory to analyze mode-to-mode coherence and mode energy in deep water environments [J. Acoust. Soc. Am., 2009]. This talk compares the energy and coherence results obtained with RMT methods to those from transport theory. [Work sponsored by ONR.]
Making, mapping, and using acoustic nanobubbles for therapy
Authors: Coussios Constantin
Published: June 2017

Acoustically driven bubbles continue to find new therapeutic uses, including drug delivery to tumors, opening the blood-brain barrier, and direct fractionation of tissues for surgical applications. Creating acoustic cavitation at length scales and pressure amplitudes compatible with biology remains a major challenge and could be greatly facilitated by a new generation of nano-scale cavitation nuclei that stretch our current understanding of bubble stability, acoustic microstreaming, and the interaction between cavitating bubbles and biological media. Furthermore, in order to ensure patient safety and treatment efficacy, monitoring the location and activity of the nanobubbles during ultrasonic excitation is essential. It will be shown that Passive Acoustic Mapping (PAM) of sources of nonlinear acoustic emissions enables real-time imaging and control of cavitation activity at depth within the body, thereby making it possible to monitor therapy using solely acoustical means. Combined, these techniques for making, mapping, and using acoustic nanobubbles can enable improved delivery of modern immuno-oncology agents to cells and tumors, needle-free transdermal drug delivery and vaccination, and new spinal therapies to treat the spinal cord or repair the intervertebral disc.
Plane-wave decomposition and ambisonics output from spherical and nonspherical microphone arrays
Authors: Gumerov Nail A. , Zotkin Dmitry N. , Duraiswami Ramani
Published: June 2017

Any acoustic sensor embedded in a baffle disturbs the spatial acoustic field to a certain extent, and the recorded field is different from a field that would have existed if the baffle were absent. Recovery of the original (incident) field is a fundamental task in spatial audio. For some sensor baffle geometries, such as the sphere, the disturbance of the field by the sensor can be characterized analytically and its influence can be undone to recover the incident field. However, for arbitrary shaped baffles, numerical methods have to be employed. In the current work, the baffle influence on the field is characterized using boundary element methods, and a framework to recover the incident field from measurements from sensors embedded on the baffle in the plane-wave function basis is developed. Field recovery also allows generation of high-order ambisonics representations of the spatial audio scene. Experimental results using both complex and spherical scatterers will be presented.
Three-dimensional remote ensemble system using the immersive auditory display “Sound Cask.”
Authors: Ise Shiro , Watanabe Yuko , Ueno Kanako
Published: June 2017

A 3D sound field simulation system using the immersive auditory display system, “sound cask,” has been developed for creating a virtual environment that would reproduce the 3D acoustics of concert halls for musicians. The simulation system is based on the boundary surface control principle. The original sound field was measured using a microphone array consisting of 80 omnidirectional microphones installed at the nodes of the C80 fullerene structure. The virtual sound field was then constructed in a cask-shaped space (approx. 2 × 2 m), with 96 channel full-range loudspeakers installed in the space. The 3D acoustic waves of music, including the acoustic condition on the stage, were created virtually inside the sound cask. For this, the first step was to design inverse filters of the MIMO system between the 96 loudspeakers and 80 microphones located in the sound cask. Next, the inverse filters and the impulse responses measured in actual concert halls and signals from instruments played by musicians were convolved in real time. Using two sound casks connected each other, three-dimensional remote ensemble system can be realized. Room acoustic indices between the actual and virtual conditions were compared, and subjective experiments involving professional musicians were performed.