bluuebunny/crossref_metadata_2025 huggingface dataset:
The relationship between vowel space area and intelligibility of speech in children with velopharyngeal insufficiency
Authors: Hashemi Hosseinabad Hedieh , Boyce Suzanne E. , Kummer Ann W. , Washington Karla , Taite Winter
Published: November 2019

Speech of individuals with velopharyngeal insufficiency is characterized by nasalized vowels. It is well known that velopharyngeal coupling significantly alters the acoustic spectrum of vowels. Aside from changes in formant amplitude and bandwidth, formant frequencies may be modified. Centralized F1 and lowered F2 due to effects of nasalization could change the dimensions of the quadrilateral space. Such variations might cause the vowel space area to be compressed which might contribute to speech unintelligibility. In pursuing research on speech intelligibility in cleft palate, this study was conducted to address the relation between speech intelligibility and spectral characteristics of American English vowels in nasal speech. The hypotheses is that narrower vowel space will result in higher degrees of unintelligibility. Ten children aged 4-12 years with different degrees of hypernasality repeated words with four corner vowels. F1 and F2 midpoint frequencies were measured, and vowel space areas were calculated using the formula for an irregular quadrilateral. In order to determine intelligibility, seventy naïve listeners were asked to orthographically transcribe the sentences produced by the participants. Correlational analysis were used to determine the relationship between the vowel space and intelligibility deficits. Further results will be discussed in the meeting.
Acoustic variability in electrolaryngeal speech
Authors: Cox Steven R. , Shadle Christine H. , Chen Wei-Rong
Published: November 2019

The electrolarynx (EL) is a hand-held electronic device that provides individuals with a means of communicating verbally postlaryngectomy. The EL produces a vibratory sound source that can be transmitted through the neck, where it excites vocal tract resonances generating speech. While some users attain a high level of intelligibility, the sound is unnatural due to numerous acoustic defects. In this study users of two EL devices were compared: the Servox emits a constant F0; the TruTone’s F0 can be varied by the user. The 10 users studied (5 Servox, 5 TruTone) used their EL for at least 24 months and were judged to be proficient. Though a previous study [Watson and Schlauch, Am. J. Speech Lang. Pathol., 18(2) (2009)] showed that intelligibility increased for a user with variable F0, TruTone use was not correlated with greater intelligibility in this study, perhaps because these TruTone users did not vary F0 much (F0 range was 3 to 20 Hz for a corpus of Harvard sentences). Detailed acoustic analyses of EL speech were considered along with other factors such as age and duration of EL experience. Results suggest directions for refinement of EL devices and training protocols for both types of EL users.
A comparison between the Korean digits-in-noise test and Korean speech perception-in-noise test in normal-hearing and hearing-impaired listeners
Authors: Seo Jae-Hyun , Oh Yonghee
Published: November 2019

The purpose of the present study was to determine whether the diagnostic efficacy of Korean version of the DIN (K-DIN) test is equivalent to that of the Korean speech perception-in-noise (K-SPIN) test, which is representative tool for speech-in-noise test. Twenty-seven subjects (15 normal-hearing listeners and 12 hearing-aid users) participated. 50 target digit triplets were presented at the most comfortable levels to each subject while presenting speech-shaped background noise at four levels of SNRs. K-SPIN test also was conducted at same procedure to the K-DIN test. The results of K-DIN and K-SPIN tests were compared by a Pearson-correlation test. The K-DIN test and the K-SPIN test were performed 74 times for each monaural and bilateral listening conditions. The results showed a statistically significant correlation between two tests in all listening conditions (left: r = 0.788, p < 0.001; right: r = 0.814, p < 0.001; bilateral: r = 0.727, p < 0.001). In the time consuming, the K-DIN test and the K-SPIN tests were about 5 minutes and 30 minutes, respectively. The current findings support that the K-DIN test is significantly correlated with the K-SPIN test in hearing-in-noise test performance, which implying that the K-DIN test can be used as a simpler and time-efficient hearing-in-noise test in Korea.
The emergence of conversational turns as a function of parent coaching on language
Authors: Ferjan Ramirez Naja , Kuhl Patricia K.
Published: November 2019

Previous research shows that parents’ use of parentese, a nearly universal speaking style distinguished by higher pitch, slower tempo, and exaggerated intonation contours, is associated with advances in children’s language learning. We recently showed that when parents are “coached” about the use of parentese when their infants are 6- and 10-months of age, they increase their use of parentese in child-directed speech, which has an immediate and positive effect on child babbling at 14 months. The present longitudinal follow-up study illuminates the mechanisms by which parentese enhanced children’s language growth in families who received parent coaching on the use of parentese. We demonstrate that, by 18-months of age, coached parents and their infants (n = 55) engaged in significantly more conversational turns compared to uncoached parents and infants (n = 24), p = 0.01, as measured by first-person LENA recordings in families’ homes. Furthermore, infants of coached parents showed enhanced language outcomes at 18 and 24 months, as measured by LENA recordings, the MacArthur-Bates Communicative Developmental Questionnaires, and a vocabulary assessment in the laboratory. We propose that parentese enhances parent-child engagement in communicative turn-taking, thereby creating a positive feedback loop, which could further advance children’s language learning.
Acoustic characterization of Singaporean children's English with American and British counterparts: A case study on approximants
Authors: Gu Yuling , Chen Nancy
Published: November 2019

We investigate English pronunciation patterns in Singaporean children in relation to their American and British counterparts by conducting acoustic analysis on /l/ and /ɹ/. A total of 110,405 utterances from American (140 speakers), British (82 speakers) and Singaporean (192 speakers) children aged 6–13 were studied. We find that similar to British children, Singaporean children do not lower F3 as much as American children when producing syllable-final /ɹ/s, suggesting a lack of rhoticity. A one-way ANOVA test demonstrated that these differences are statistically significant across speaker groups: F(2,411) = 750.9, p < 0.001, and a post-hoc Tukey's HSD test showed that all differences in pairwise comparisons are significant too. Interestingly, similarity between Singaporean and British pronunciation patterns are not observed for syllable-final laterals (dark /l/). We observe that Singaporean children's dark /l/s are produced with characteristics distinct from both those of American and British children. While American and British children demonstrate lowering of F1 and F2 in transitions into dark /l/s, these are not exhibited in Singaporean children's pronunciation. One-way ANOVA tests showed that these differences are significant: F(2, 1519) = 55.09, p < 0.001 for F1; F(2, 1519) = 118.7, p < 0.001 for F2, and post-hoc Tukey's tests reached statistical significance too.
Audiovisual enhancement in the perception of children’s speech
Authors: Karisny Jacquelyn , Smith Gisela , Oeding Kristi , Hagen Alexandra , Munson Benjamin
Published: November 2019

It is nearly axiomatic that audiovisual (AV) speech is more intelligible than audio-only (A-only) speech, particularly when the speech is presented in a challenging listening environment, such as in background noise [e.g., MacLeod and Summerfield, Br. J. Audiol., 2 (1987)]. No previous research on audiovisual speech perception has examined the perception of children’s speech. Children may elicit a smaller AV benefit than adults, as their visual articulatory movements are more variable than adults’ [e.g., Smith and Goffman, J. Speech Lang. Hear. Res. 41 (1998)], and hence are less informative perceptual cues. Alternatively, the overall lower intelligibility of children’s A-only speech might lead them to elicit overall higher AV benefits than adults. To examine this question, we collected developmentally appropriate sentence productions from five, 4-6 year old children, and five sex-matched adults. Ongoing work is examining the intelligibility of these sentences in multitalker babble in A-only and AV conditions in a variety of signal-to-noise ratios, so that we can compare AV benefits for children and adults when A-only intelligibility is matched. Both sentence intelligibility and eye gaze during perception are being measured. Results will help us understand the role of individual-speaker variation on the magnitude of AV benefit.
A few comments on passive synthetic aperture sonar in honor of Ed Sullivan
Authors: D'Spain Gerald L.
Published: November 2019

I came to know Ed Sullivan after submitting a paper on passive synthetic aperture sonar for publication in the Journal of the Acoustical Society of America. At that time, he was an Associate Editor of the journal. I knew of him, by reputation, of course. Only after the paper reviews came back did I get to know him, the person, the hard-core, old-school signal processor and physicist. At first, I simply tried to humor him to get the paper accepted for publication, figuring that quotes from a few of his papers sprinkled here and there would suffice. But after spending time expanding my knowledge of his work, I developed a deep appreciation for the foundation in signal processing he and colleagues had built. The published version of our paper was significantly improved as a result. This talk presents some of Ed Sullivan's results on model-based signal processing and its application to passive synthetic aperture sonar, along with some of our published results. Its main purpose, however, is to serve as a reminder that we all stand on the foundation built by those who have gone before. [Work supported by the Office of Naval Research.]
Deconvolution of underwater acoustic marine mammal signals from the Northern Gulf of Mexico
Authors: Leftwich Kendal , Ioup Juliette W. , Haas Michael J.
Published: November 2019

Extracting useful information from electronically recorded data measurements often requires the signal to be separated from the noise and the blurring impulse response deconvolved out. In order to denoise LADC-GEMM underwater acoustic data, we apply and compare several techniques from Fourier frequency filtering and wavelet denoising. We also look at methods of estimating the acoustic impulse response (system response, transfer function, etc.) for the northern Gulf of Mexico to be used in implementing several deconvolution methods applied to the same dataset. Deconvolution techniques tested include Fourier deconvolution, Wiener filtering, various blind deconvolution methods, as well as combination algorithms. Comparisons of techniques will be presented, showing that the different methods may yield different results. Metrics for analysis will include both the final quality of results along with computational efficiency. Some recommendations will be made within the framework of marine mammal signal detection; although specifics of denoising and deconvolution will often be data dependent, these ideas can be applied more broadly to a variety of situations.
Reconstruction of volume scattering coefficient with considering seabottom scattering layer
Authors: Vornovskikh Polina , Sushchenko Andrei , Prokhorov Igor
Published: November 2019

A problem of remote sensing in the ocean by a point isotropic sound source is considered. To describe the process of remote sensing of the ocean, the authors selected a kinetic model based on the integro-differential radiation transfer equation. The inverse problem is formulated, which consists in reconstructing the coefficient of volume scattering in the seabottom layer. An equation for simulating the single scattered signal is obtained. Computational experiments are carried out to analyze the solution of the inverse problem.
Renovation of a college gymnasium into a multimedia maker-space
Authors: Nelson Smid Jennifer , Miller Gregory A.
Published: November 2019

The Haverford College Visual Culture Arts and Media (VCAM) project is an adaptive-reuse of an old gymnasium building. The building is stone construction, built in 1900, and unused for decades. The redevelopment included 25 000 SF with a wide range of collaborative, classroom, and laboratory spaces that support the creation, exploration, and study of visual and media arts. The main large volume of the original gymnasium became a lounge and presentation space for multi-media artwork. Significant portions of the original gym floor were removed, allowing a three-story box to be inserted into the volume to house (from top to bottom): a Media, Film Production and Object Study Room; a Film and Media Edit Room; and a 65-seat Screening Room. An original suspended running track was repurposed into flexible seating and circulation to offices and studios. Many acoustic challenges arose from repurposing the large volume gymnasium including reverberation control, structure borne noise and vibration isolation of stacked studios, and acoustic isolation of a noise sensitive screening room located in vicinity of maker spaces utilizing drills, buzz saws, and laser cutters. Audio/Video systems supported a wide range of activities, including a number of sophisticated recording, filming, editing and playback scenarios.
Working towards a standardized approach to reporting of sound measurements in wildlife studies
Authors: McKenna Megan F. , Fristrup Kurt M.
Published: November 2019

We present guidance we have developed and adapted through attempts to synthesize and distill the diverse literature documenting wildlife responses to noise into concise support for conservation planning. Our systematic review revealed significant inconsistencies in the use of—and specifications for—acoustic metrics. Studies typically report noise levels in decibels (dB), yet these values can arise from a diverse array of methods and processing procedures. Though some of this diversity reflects varied objectives and acoustical contexts, we believe that normative guidance and better awareness of standards would enhance the interpretation, repeatability of published studies, and incorporation of results into conservation planning. Nested within this larger issue is the vexing problem of calibration, especially in relation to consumer audio recorders or emerging products for bioacoustical monitoring. While it is possible to extract useful information about acoustic phenology or species presence from uncalibrated audio recordings, calibration can deliver a much broader range of information. We review the suite of calibration options presently available, and our experience with implementing and sharing solutions.
Music information retrieval—The impact of technology, crowdsourcing, big data, and the cloud in art
Authors: Kostek Bozena
Published: November 2019

The exponential growth of computer processing power, cloud data storage, and crowdsourcing model of gathering data bring new possibilities to Music Information Retrieval (MIR) field. MIR is no longer music content retrieval only; the area also comprises the discovery of expressing feelings and emotions contained in music, incorporating other than hearing modalities for helping this issue, users’ profiling, merging music with social media and qualitative recommendations in music services. Moreover, 5G telecommunications networks, characterized by “near-instant and everything in the vicinity talks with one another,” with exponentially faster download and upload speeds, may change the existing models and create a new age of interconnectedness. This paper aims at showing some of the already highly exploited technologies and crowdsourcing models applied to music processing. Several studies are discussed in details, such as, e.g., deep learning applied to music, a way to generate an expanded training sets using 2-D data such spectrograms, mel-cepstrograms, chromagrams, and waveform-based representations of the signal instead of feature vectors in machine learning, allowing to retain all nuances related musical articulation in the signal. Also, a discussion is to be outlined, expanding the issue of the impact of these new technologies on the artistic and aesthetic values of music.
Fearless steps, NASA’s first heroes: Conversational speech analysis of the Apollo-11 mission control personnel
Authors: Joglekar Aditya , Hansen John H. L.
Published: November 2019

Between 1963 and 1972, a massive team of dedicated scientists, engineers, and specialists at the NASA Mission Control Center (MCC) worked seamlessly together in a cohesive manner to successfully carry out multiple manned missions to the moon. All communications between personnel were carried out over multiple inter-connected audio channels and recorded on two 30-track analog tapes. Digitization of the entire Apollo-11 mission tapes made possible through the UTDallas-CRSS Fearless Steps (FS) initiative contains the recordings of all the crew members including the three astronauts. With over 600 speakers in constant communication ensuring the astronauts’ safety throughout the mission, the success of the Apollo missions can be attributed largely to the MCC crew members. The focus of this effort is thus, to analyze their high-level group dynamics and intricate communication characteristics and gain insight into the success parameters involved in such large-scale time-critical operations. The 100 hour FS Challenge corpus highlights multiple salient moments like Lift-Off and Lunar-Landing and poses its’ own challenges for core speech tasks. This effort aims at analyzing spontaneous multi-speaker conversations in recordings often degraded by various noise types. Analysis of speech characteristics under varying stress, overlap, and noise conditions are observed to develop novel speech-activity and speaker models. These domain-specific models are leveraged to develop state-of-the-art SLT systems.
Effect of signal to noise ratio on a convolutional neural network for source ranging and environmental classification
Authors: Howarth Kira , Van Komen David F. , Neilsen Tracianne B. , Knobles David P. , Dahl Peter H. , Dall'Osto David R.
Published: November 2019

In ocean acoustics, simultaneous estimation of both source-receiver range and environment are complicated by low signal-to-noise ratio (SNR). Range and environment class can be found with a convolutional neural network (CNN), which is chosen because of its ability to find patterns in grid-structured data. The CNN acts on synthetic pressure time series data from a single receiver generated for four canonical environments: deep mud, mud over sand, sandy silt, and sand. Data were split into training and validation sets. The CNN is trained to identify source range and environmental class. The change in performance for different SNR values is evaluated by adding Gaussian-distributed noise. A study is done regarding the impact of having different SNR values for the training and validation datasets. The trained CNN is applied to pressure time series data measured on the APL-UW Intensity Vector Autonomous Receiver system at SBCEX. This study shows that performance depends more on the suitability of the training dataset than on the SNR value, implying that a CNN has potential to both estimate range and environmental class, even when there is low SNR. [Work supported from Office of Naval Research.]
Approximate models for the structural-acoustic response of elastic barriers suitable for hybrid analytical-computational techniques
Authors: Villa Mauricio , Bliss Donald B.
Published: November 2019

Exact and approximate models are considered for the fluid-loaded response of finite panels driven by an obliquely incident acoustic wave. To evaluate the acoustic back-coupling of these fluid-loaded structures, the kernel function of the integral-differential equation is approximated using new and efficient trigonometric representations for small arguments. This representation allows for accurate matching to the asymptotic expansions with very few terms. The approach efficiently treats arbitrary boundary conditions by interpreting them as deviations from a simply supported panel corrected by a smooth force distribution. Simple, closed-form approximate models that represent the dominant radiation mechanism are also pursued. The fluid-loaded reverberation of the panel is modeled by approximating the inertial and radiation damping effects of the bounding fluids based on an understanding of the radiation mechanisms for subsonic and supersonic flexural waves. Comparison to benchmark solutions confirms the simple radiation models successfully describe the dominant characteristics of the structural-acoustic response. These models are demonstrated to work extremely well in the mid to high frequency range. Their success has important implications for the creation of hybrid analytical/computational models that are efficient and accurate for multi-dimensional mid-frequency structural acoustics.
An efficient model order reduction of frequency-dependent double panel systems
Authors: Cavalheiro Thiago , Lenzi Marcos S. , Martins Gustavo C. , Lenzi Arcanjo
Published: November 2019

This paper treats a typical industrial vibroacoustic problem, consisting of a double panel under forced vibration which radiates to the far-field. A finite element model is proposed, where both structural links and porous material are represented as equivalent models. The addition of frequency-dependent elements, as mechanical links and porous material in the cavity, leads to a balance between airborne and structural-borne transmission through panels in the mid-frequency range, which demands a suitable model to represent these phenomena. In order to cope with such large-scale models, a dedicated parametric model order reduction is proposed. The frequency-dependent terms are separated from the system matrices, allowing an offline reduction of the system. The Second-Order Arnoldi procedure is employed to calculate the reduced bases at multiple frequencies, chosen by a rational interpolation scheme. The expansion frequencies are selected by a novel adaptive windowing algorithm, which aims at minimizing the number of frequencies within the frequency range. The proposed approach has shown to be effective in reducing significantly the computational time while maintaining precision compared to the direct method.
Disyllabic word context effects on cue weighting in Mandarin tone perception and production
Authors: Zhang Xiaojuan , Cheng Bing
Published: November 2019

This study aims to examine effects of the disyllabic tonal context on native Chinese speakers' cue weighting for perceiving and producing target Mandarin Tone 2 and Tone 3 that share similar pitch contours. The speech materials were disyllabic Mandarin words, matched by word frequency and familiarity. Acoustic manipulations for the perception test included four key parameters, including fundamental frequency (F0) at tone onset, F0 at turning point, the timing of turning point and F0 at tone offset, which characterized the pitch contours of Tone 2 and Tone 3. Thirty native Chinese speakers participated in this study. These synthetic disyllabic stimuli were used in an identification task. Production of the target tones in the disyllabic context was also recorded from each participant. Cue weights for the key acoustic parameters were assessed with logistic regression analyses and linear discriminant analyses to model the extent to which each acoustic cue predicted category membership in the perception and production data. The results demonstrated strong perceptual weighting sensitivity to the disyllabic context in native Chinese speakers. The results are discussed with respect to the dynamics of tone perception and production in word contexts and potential implications for Chinese as a second language learners.
Variability of formant values at different time points of vowels
Authors: Jibson Jonathan
Published: November 2019

Vowel inherent spectral change research has led to two apparently contradictory conclusions about the mental representation of vowels: perception studies support a two-target model of vowels, while production studies focus on calculating continuous formant contours. This study explores production data in a new way by analyzing where in vowel duration variability is lowest across tokens. Portions of vowels that are less variable across tokens can be taken as reflecting greater importance to or planning by speakers. Born-and-raised Wisconsinites aged 18–30 were recorded reading a word-in-frame reading list eliciting 30 tokens of 14 American English vowels (all but /ə, ɚ/) in the context hd. F1 and F2 were sampled at 50 evenly spaced intervals for each vowel, covering the entire duration. For every speaker, the standard deviations of F1 and F2 were taken. Values for F1 were almost entirely below the perceptual threshold (0.28 Bark), while values for F2 were mostly above. The all-speaker-averaged values were subsequently plotted as a pair of bar charts for every vowel. F1 plots are primarily U-shaped with vertices near midpoints, as are half of F2 plots, with the other half being flat. Neither shape supports a two-target representation of vowels from a production standpoint.
A curvilinear comparison of normalized English formant trajectories
Authors: Yang Byunggon
Published: November 2019

Formant trajectories reflect the continuous variation of speakers’ articulatory movements over time. This study examined formant trajectories of English vowels produced by ninety-three American men and women in Hillenbrand et al. (1995). Praat scripts were used to read the sound data, and to collect formant values at the six equal time points within each vowel segment. The scale function in R was used to normalize the first two formant data sets of men and women separately. Then, the scaled trajectories were compared using generalized additive mixed models (GAMMs). The results indicate that women yielded proportionately higher formant values than men. The standard deviations of each group showed similar patterns at the first formant (F1) and the second formant (F2) axes and at the measurement points. GAMMs of all the scaled formant data produced various patterns of deviation along the measurement points. Generally, more group difference exists in F1 than in F2. Also, women’s trajectories appear more dynamic along the vertical and horizontal axes than those of men. We conclude that scaling and nonlinear testing are useful tools for pinpointing differences between speaker group’s formant trajectories. This research could be useful as a foundation for future studies comparing curvilinear datasets.
Voice onset time variation in natural southern speech
Authors: Lipani Lisa , Olsen Michael , Olsen Rachel M.
Published: November 2019

Sociophonetic research has traditionally emphasized vowels and only recently begun to examine consonants. The emerging literature on consonants has found systematic regional variation in consonant production (Jacewicz et al., 2009; Eddington and Turner, 2017). These studies have focused on present-day speakers producing lab-recorded speech and thus provide little historical or naturalistic insight to consonant variation. One way that the production of stop consonants varies is in voice onset time (VOT). In this study, we utilize the Digital Archive of Southern Speech (DASS) (Kretzschmar et al., 2013), a collection of 64 sociolinguistic interviews recorded between 1968 and 1983, to explore consonant variation in an unscripted historical setting. DASS was force-aligned using the Montreal Forced Aligner (McAuliffe et al., 2017), and the VOT of pre-vocalic, word-initial stop consonants in three-minute audio clips from each speaker was measured using AutoVOT (Keshet et al., 2014). VOT was normalized by dividing duration by speaking rate, and this normalized measurement was included as the dependent variable in a mixed effects model. As expected, stress, voicelessness, and a dorsal place of articulation were significant predictors. Our preliminary analysis reveals significant regional and age differences, adding evidence for speaker-specific variation of VOT as a result of sociophonetic variables.
The roles of vowel length and sentential context in onset pitch perturbations in Thai
Authors: Silpachai Alif
Published: November 2019

This study investigates the relationship between fundamental frequency at the onset of voicing (onset f0) and Voice Onset Time (VOT) in a tonal language with prevoiced, short-lag, and long-lag stops. Recent research on Thai and Vietnamese has suggested that higher f0 in the following vowel is conditioned by long-lag stops, but this effect occurs more in higher, not lower, tones and in words produced in isolation, not in a carrier phrase. An examination of previous studies, however, suggests that the effect may be moderated by vowel length and the type of carrier phrase. To determine whether this is true, this study compares onset f0 measured 40 ms after voicing onset in Thai low tone words with phonemically short and long vowels that occur in two types of carrier phrases and in isolation. The results show that prevoiced, not short- or long-lag, stops condition higher onset f0 in short, not long, vowels, and this effect takes place in words occurring in both types of carrier phrases, not in isolation. This suggests that vowel length may be a relevant factor. The results will be discussed further, and implications for onset f0 control will be offered.
The SCAT model of wideband biosonar achieves shape imaging, clutter rejection, and mitigation of pulse-echo ambiguity by frequency hopping
Authors: Simmons James A. , Ming Chen , Tuninetti Amaro , Simmons Andrea
Published: November 2019

Echolocating big brown bats emit wideband (22–100 kHz) FM sounds and perceive echo delay over the full band by auditory spectrogram correlation. However, echoes must contain the lowest FM frequencies (22–30 kHz) as a condition for delay perception; echoes containing fully the higher 90% of frequencies are rejected as having delay images. For echo spectral interference nulls, bats perceive the delay separation of glint reflections corresponding to the frequency spacing of the nulls, a process of spectral transformation. They also perceive echo phase shifts (0 deg to 180 deg) as corresponding alternations in delay (0 ms to ±15 ms). This sensitivity requires registering phase at the lowest frequencies first and is essential for binaural localization and synthetic-aperture shape reconstruction. In dense clutter, bats alternate their lowest frequencies in successive broadcasts from about 22–25 kHz to about 27–30 kHz for “frequency hopping” to mitigate pulse-echo ambiguity for overlapping echo streams in clutter. The SCAT biosonar model replicates all these critical performances. By initiating echo processing at the lowest frequencies and then working up the FM sweeps to complete the delay estimate, it accommodates phase delay for azimuth, synthetic-aperture imaging, and pulse-echo ambiguity when frequency hopping occurs. [Work supported by ONR.]
Incorporating uncertainty into detection probabilities and receiver-operator characteristic curves
Authors: Worthmann Brian M.
Published: November 2019

Receiver-operator characteristic (ROC) curves are used in a variety of fields to assess the performance of binary classification algorithms. A variety of metrics can be defined that quantitatively summarize ROC curve performance, including area under the curve (AUC). In measured data, the detection statistic for each trial typically has some uncertainty associated with it. ROC curves generated from limited measured data that ignore this uncertainty can appear stair-stepped, and produce a single value for the AUC, which lacks any estimate of its variance. Incorporating the uncertainty in the detection statistic directly can lead to smooth ROC curves parameterized by a mean and variance, which then lead to AUC’s with both a mean and variance. With this, the significance between two or more ROC curves, and their AUC’s, can be assessed statistically. If the classification algorithm that produced the detection statistics have tuning parameters, the optimal values for these tuning parameters can also be determined, along with their associated variances. This methodology is presented alongside other techniques from literature. [This work was performed under the auspices of the U.S. Department of Energy by Lawrence Livermore National Laboratory under Contract No. DE-AC52-07NA27344 and is supported by the Office of Naval Research.]
Automatic detection and tracking of contacts based in clusterization applied in passive sonar
Authors: Silva Fabio O. , Bozzi Fabrício , Monteiro Fernando d. , Filho William S. , Soares Carlos F.
Published: November 2019

In passive sonar, the detection and tracking of contacts based on the energy of Direction of Arrival (DoA) is a challenge for the spatiotemporal clusterization algorithms. Typically, the tracking of spatiotemporal data involves two steps, first a clusterization of a temporal frame to reduce the data to micro-cluster and a second step tracking the micro-cluster moving. That tracking approach has problems to predict the motion and resolve merge and split of clusters. In this paper, we propose, analyze e test an algorithm to explore the characteristics of passive sonar` signals and improve the tracking performance. To resolve the bearing crossing we proposed an automatic parameter extraction on Low-Frequency Analysis and Recording (LOFAR) and Detection of Envelope Modulation on Noise (DEMON) with the same tracking abording to extract the contact acoustic signature and in case of crossing reassociate the contact based on his signature. The proposed algorithm is analyzed in a theoretical dataset to parameters tuning and evaluated in a real dataset collected by a flank array in a navigation channel.
Distortion of the ambient low-frequency acoustic field by moorings
Authors: Godin Oleg A.
Published: November 2019

Moored receivers are often employed to collect long time series of underwater acoustic data to monitor the ocean. Moorings typically contain a float and other elements, acoustic parameters of which are rather distinct from those of seawater. In the frequency range of primary interest in seismo-acoustics and noise interferometry, the size of a mooring and dimensions of its float may be of the order of the acoustic wavelength or smaller. Then, acoustic measurements are performed in the near field of a strong scatterer. Assuming a spherical shape of the float, this paper aims to quantify the difference between the ambient and measured characteristics of low-frequency acoustic fields. Diffraction is shown to significantly affect low-frequency acoustic fields in a vicinity of a compact object even when the object is small compared to the wavelength. Scattering-induced low-frequency perturbations in oscillatory velocity are generally much larger than pressure perturbations in the vicinity of sub-wavelength objects. These perturbations prevent mounted vector sensors from measuring correctly the direction of the free-field oscillatory velocity. The feasibility of a posteriori compensation of the distortions in scalar and vector sensor measurements will be discussed. [Work supported by ONR.]
The soundscape of Sen. Salgado Filho Avenue, Natal/RNBrazil: The acoustic impact caused by the insertion of semi-exclusive bus lane
Authors: Vieira Bruna P. , Barbosa Edy J.
Published: November 2019

Measures have been adopted in Brazilian roads in order to prioritize collective transportation over the individual. In Natal/RN, the lane next to the sidewalk was destined to the bus circulation at Senador Salgado Filho Avenue, raising questions about the noise impact felt by people at bus stops. Therefore, the objective is to understand the influence of the road configuration on the environmental sound quality to point out elements for mobility plans. The methodology used was: measuring of sound pressure levels at bus stops; questionnaires to verify the evaluation of the pedestrians on the sound impact of the semi-exclusive track; development of acoustic maps based on the current scenario and the reallocation of public bus to the central site. The results demonstrated sound levels on sidewalks higher than those recommended by WHO, and that people are disturbed by the noise due to the proximity of the bus. Through the calibrated models, it was found that the location of the semi-exclusive lane on the right is less noisy than in the central bed, a solution adopted in others roads, showing how the studies of sound impact can be used in urban and mobility planning to increase the pedestrians comfort on sidewalks.
Improving the clinical environment soundscape
Authors: Taylor Ally , Moore Dr David , Simpson Liz , Smith Wendy
Published: November 2019

This project examines the impact of the clinical acoustic environment on both patients and staff within a large modern inner-city acute hospital (1677 beds). It aims to both summarize and improve the hospital soundscape. Three key parts of experimental work were carried out: noise level measurements were taken in the Queen Elizabeth University Hospital atrium in Glasgow, interviews were undertaken with medical personnel and listening experiments were conducted to assess if adding natural sounds into the environment could reduce the negative effects noise can have on the occupants of the hospital. The atrium area of the hospital was found to be far louder than needed for rest and relaxation, with all recorded equivalent noise levels over 55 dB(A). Relaxing sounds of nature were added to the hospital soundscape to harness the restorative power of nature. The effect of this was tested using the perceived restorativeness soundscape scale; there was mixed evidence that the introduction of natural sounds led to an increase in the environment’s restorative power, with four out of eight attention restoration theory components tested displaying a statistically significant difference. The medical personnel interviewed indicated that the idea of introducing natural sounds is promising and requires further research.
Experimental study of bubble sources localization in a water tank for oil spill detection
Authors: Yang Xing , Cao Lei , Lu Zhiqu
Published: November 2019

Since the Deepwater Horizon oil spill in 2010, there have been increasing demand for a real-time monitoring system that can detect, locate, and assess oil leakages. To this goal, we launched a three-year research project to develop a passive acoustic technique in 2017. The leaked crude oil creates underwater sound through bubble oscillations in an oil spill event. This paper focuses on studying the localization of the source of bubble-induced sounds in an experimental environment. Hydrophone arrays were deployed randomly and used to receive the acoustic signals in a 2 m × 2 m × 2 m water tank. We used spatial spectrum analysis algorithms such as Multiple Signal Classification (MUSIC) and Estimation of Signal Parameters by Rotational Invariance Techniques (ESPRIT) to determine the direction of arrival (DOA) by a hydrophone Uniform Linear Array (ULA), and utilized the multiple 2-D angles to localize the bubbles in 3-D space. We will also investigate the sensitivity of the localization algorithm with respect to the variations of physical features such as the cluster of frequency components and noise level. Experiment results will be presented and discussed.
Auditory feedback control of vocal intensity during speech and sustained-vowel production
Authors: Hilger Allison I. , Levant Sam , Kim Jason , Lester-Smith Rosemary A. , Larson Charles R.
Published: November 2019

Vocal intensity is an important speech component for communicative effectiveness, such as speaking above ambient noise in a loud restaurant, and also for communicative appropriateness, such as speaking quietly in a library. Speakers adjust their vocal intensity using information from their auditory feedback of how loud they perceive their speech. In this study, we were interested in the control of auditory feedback for vocal intensity. Participants repeatedly produced both a simple, sustained-vowel (i.e., “ahhh”) and a short target phrase (i.e., “You know Nina?”) while their voice auditory feedback was briefly perturbed up or down in loudness. We then measured their reflexive intensity response produced quickly after the perturbation to correct for the unexpected change in loudness. In a similar study on pitch perturbations, speakers produced larger reflexive pitch responses as a function of perturbation magnitude in phrase production but did not produce this differential pattern in sustained vowel production (Chen et al., 2007). This result suggests that auditory feedback is more sensitive to changes in suprasegmental features of speech when produced with linguistic intent. In our current study, we are interested in whether this same pattern will be observed for control of vocal intensity.
Frequency-wavenumber spectrum estimation using blended dominant mode rejection beamforming
Authors: Wage Kathleen E.
Published: November 2019

Capon [Proc. IEEE (1969)] designed the minimum variance distortionless response (MVDR) beamformer to obtain spectral estimates with better resolution than the conventional averaged-periodogram estimator. The MVDR spectrum is a function of the inverse of the sample covariance matrix (SCM), which often must be regularized prior to inversion. To address conditioning problems, Abraham and Owsley [IEEE Oceans (1990)] developed a modified MVDR approach called dominant mode rejection (DMR). The DMR beamformer defines its weights using a structured covariance consisting of a low-rank interference subspace plus an orthogonal noise subspace. It assumes the rank of the interference is known. Recently, Buck and Singer [IEEE SAM (2018)] proposed the blended DMR beamformer that eliminates the need for rank estimation by defining a weight vector that is an affine combination of fixed-rank DMR beamformers. This talk investigates frequency-wavenumber estimation using blended DMR, focusing particularly on efficient implementations for large linear arrays. Adapting the approach described by Therrien [Prentice Hall (1992)] for MVDR, the blended DMR spectrum for an equally spaced array can be computed using fast Fourier transforms of the sample eigenvectors. Results will be illustrated using experimental data from underwater vertical arrays. [Work supported by ONR.]
High-amplitude focusing of ultrasound in air using time reversal
Authors: Wallace Carla B. , Anderson Brian E.
Published: November 2019

In this study, time reversal (TR) signal processing methods are used to create a focus of airborne ultrasound (between approximately 30 kHz and 50 kHz) in a room. There are unique challenges presented by ultrasonic sources, such as the highly directional nature of the sources, and attenuation of ultrasound in air. The aim of this study is to create a TR focus whose amplitude is as high as possible using the clipping processing method. Because ultrasonic sources tend to be very directional, this study explores the impact of using methods to make the sources more omnidirectional, which increases the number of reflective paths that contribute to the TR focus. This study explores various configurations of source and microphone in the room to determine which configuration yields the highest amplitude. Because thermoviscous loses for ultrasound are higher than for audible frequencies, a smaller room is used in the experiments to increase focal amplitudes.
Using the frequency-difference autoproduct to passively range remote sources in the deep ocean
Authors: Geroski David J. , Dowling David R.
Published: November 2019

Signal processing techniques utilizing the nonlinear frequency-difference acoustic-field autoproduct have been applied in recent years to successfully localize remote sources in the shallow ocean [Worthmann et al., JASA 138, 3549–3562 (2015)] and in the deep ocean [Geroski and Dowling, JASA (to appear)] by switching the signal processing to below the signal band frequencies. The successful source localization results obtained here indicates the robustness of frequency differencing techniques to the problem of mismatch between measured and computed acoustic fields that plagues source localization algorithms based on correlating computed and measured field. The promising deep ocean results were obtained by analyzing PhilSea10 experimental acoustic recordings from a water-column spanning vertical array with 149 receivers and an approximate length of 5 km. Given that such an extensive receiving array is unlikely to be routinely available, an investigation was conducted to determine how reduced array aperture and transducer count influence source ranging results from frequency-differencing techniques for 100-Hz bandwidth signals having center frequencies from 170 and 260 Hz, and source to array ranges between 100 and 500 km. Ranging results from this study are presented for both simulated and measured in-band signals, and are calculated using single-digit Hz difference frequencies. [Sponsored by ONR.]
An algebraic geometry approach to passive source localization
Authors: Cheney Margaret , Kirsteins Ivars P.
Published: November 2019

We examine the application of algebraic geometry-based localization methods to 3-D underwater acoustic source localization. An approach to passive source localization that is commonly used in the electromagnetics community is to first determine the time difference of arrivals (TDOAs) by cross-correlating pairs of received signals, and then from the TDOAs use algebraic geometric methods to determine the source location. In particular, a TDOA for a receiver pair determines a hyperboloid on which the source must lie. Consequently the source must lie at the intersection of the TDOA hyperboloids. Because hyperboloids are examples of polynomials, systems of hyperboloids can be solved with numerical algebraic geometry software such as Bertini or Macaulay2. Such software typically works by starting with a reference system of polynomials whose exact solution is known, and tracking the solution while gradually deforming the reference system into the desired one. This approach is less well-known in the sonar community because the sonar wave propagation environment is more complicated. We begin by applying this algebraic geometry approach to the case of an iso-velocity range-independent sound speed environment and then investigate extensions of this approach to underwater waveguides with non-homogeneous sound speed profiles.
Adapting underwater acoustic communication networks to changing oceanic conditions Using opportunistic multipath signaling schemes
Authors: Sen Gupta Ananya
Published: November 2019

A fundamental bottleneck to high data-rate shallow water acoustic communications is rapidly fluctuating delay spread, which is caused by highly unpredictable multipath scattering by the moving sea surface, sea bottom as well as fluid motion. A rich literature exists on how to track and mitigate the mutipath interference and thus recover the direct arrival between the transmitter and receiver. This talk will review some of the recent methods proposed, in particular light of different representations of the channel multipath in delay, time, frequency domains and combinations thereof. The talk will also present how the different representations of channel multipath can be exploited to design opportunistic signaling schemes that exploit rather that mitigate multipath effects. We will also discuss the impact of such opportunistic signaling, particularly in the context of diversity strategies in communication networks.