bluuebunny/crossref_metadata_2025 huggingface dataset:
Non-Markov character of sound propagation over relatively short ranges in the turbulent atmosphere
Authors: Ostashev Vladimir E. , Wilson D. Keith
Published: April 2018

The Markov approximation is widely used in wave propagation in random media, including sound propagation in the turbulent atmosphere. This approximation, which significantly simplifies formulations for the statistical moments of the propagating field, is valid if the propagation range is greater than the outer length scale of random inhomogeneities. For sound propagation in the turbulent atmosphere, this length scale can be as large as 400–500 m (23% of the height of the atmospheric boundary layer), indicating that the Markov approximation might not be applicable for relatively short ranges. In this paper, the statistical characteristics of a sound field propagating in the turbulent atmosphere, such as the correlation functions of the log-amplitude and phase fluctuations, the mean sound field, and the mutual coherence function are calculated, without using the Markov approximation. The new results are then compared with those obtained in the Markov approximation. The difference between the two formulations and the range of applicability of the Markov approximation are analyzed for different meteorological regimes of the atmospheric boundary layer characterized by the surface heat flux and the friction velocity.
On the use of high-order time-domain impedance boundary conditions
Authors: Dragna Didier , Blanc-Benon Philippe
Published: April 2018

Time-domain simulations are well-suited to study broadband sound propagation. One of the difficulties is the translation of frequency-dependent impedance boundary conditions that leads in the time domain to convolutions. Several methods have been proposed in the literature to have an efficient computation of convolutions. They are based on a multipole approximation of the impedance and on approximations of time-variations of the acoustic variables, allowing the convolution to be simply evaluated by recursive relations. Recently, Dragna et al. [J. Acoust. Soc. Am. 138, 1030–1042 (2015)] have highlighted that recursive convolution methods are low-order methods and have introduced a new method, referred to as the auxiliary differential equation method, which allows one to compute convolutions by integrating in time ordinary differential equations. Its main advantage is that it preserved the order of accuracy. This approach was employed in Troian et al. [J. Sound Vib. 392, 200–216 (2017)] to derive a time-domain impedance boundary condition (TDBIC). This paper aims at evaluating the accuracy and efficiency of this novel TDBIC and to compare it to those of recursive convolution approaches. The novel TDBIC is first presented. A one-dimensional test case, dealing with reflection of an acoustic pulse over an absorbing wall is then investigated. Examples of simulations in three-dimensional geometries in the context of duct acoustics or outdoor sound propagation are then shown.
Computation of dispersion curves of guided waves using discontinuous Galerkin finite element method: Application to functionally graded materials
Authors: Hebaz Salah-eddine , Benmeddour Farouk , Moulin Emmanuel , Assaad Jamal , Duquennoy Marc , Ouaftouh Mohammadi
Published: April 2018

The study of dispersive behavior of guided waves is of great interest in non-destructive testing and development of inspection systems. In this work, a semi-analytical numerical formulation for the computation of dispersion curves and mode shapes in plate-like structures is presented. The formulation is based on a discontinuous Galerkin Finite Element Method. The discretisation of the ordinary differential system in the through-thickness direction yields an eigenvalue problem. The resolution of the latter for a frequency range provides the dispersion curves. The study focuses on Lamb modes in functionally graded material plates with traction-free boundary conditions. The influence of the gradient variations is demonstrated. First, a multilayered approximation consisting of homogeneous laminate is used. Afterwards, the gradients of the parameters are added to the formulation. Numerical examples are presented and compared with those found in the literature. The results are free of spurious modes and show an excellent agreement for all cases, specially for continuously varying properties. The method is very efficient and numerically stable for small and large wave numbers. It was found that a high accuracy is obtained when high-order elements are used on a relatively coarse mesh.
Assessment of learning algorithms to model perception of sound
Authors: Rafaelof Menachem , Schroeder Andrew
Published: April 2018

Predicting human response to complex sound is a nontrivial task. Besides large differences among subjects and practically infinite types of stimuli, human response to sound is typically quantified by a few parameters having nonlinear behavior. Still, such predictions are valuable for the assessment of sound quality, which is a critical step toward the development of systems that offer improved human comfort, productivity and wellness. The overall objective of this work is to learn about human perception of sound through the use of machine learning algorithms. Learning algorithms are ideal for modeling the complex behavior of subjective parameters and identifying new trends with the potential to accumulate knowledge from different experiments. This work compares the performance of four learning algorithms (linear regression, support vector machines, decision trees, and random forests) to predict annoyance due to complex sound. Construction of these models relies on the annoyance response of 38 subjects to 103 sounds described by five known predictors (loudness, roughness, sharpness, tonality, and fluctuation strength). Comparison of these algorithms in terms of prediction accuracy, model interpretability, versatility, and computation time indicates that decision trees and random forests are the best algorithms for this task.
Listener preferences and speech recognition outcomes using self-adjusted hearing aid amplification
Authors: Perry Trevor T. , Nelson Peggy B. , VanTasell Danne
Published: April 2018

Self-adjustment of hearing aid gain empowers users to set amplification parameters according to their personal preferences. Listeners with mild-to-moderate hearing loss demonstrated self-consistency when adjusting gain in noisy backgrounds, although variability across subjects was very high and poorly predicted by user characteristics such as age, hearing thresholds, or previous experience using hearing aids. Despite the wide range of self-fit settings, speech intelligibility in noise was similar between self-fit and audiologist-fit settings. For average-level speech in noisy backgrounds, audibility was driven primarily by the signal-to-noise ratio, not gain settings. Self-adjustment for speech in quiet backgrounds will be compared to results obtained in noise. User interactions with the self-adjustment technology as well as preferences for self-fit settings and audiologist fit settings suggest that users can successfully use self-adjustment technology to achieve more desirable amplification from their hearing aids.
A visually guided beamformer to aid listening in complex acoustic environments
Authors: Jennings Todd R. , Kidd Gerald
Published: April 2018

The purpose of this talk is to provide an overview of recent work on the "visually guided hearing aid" (VGHA; Kidd et al. JASA 133, EL202). The latest prototype VGHA consists of a lightweight, head-mounted, 18-microphone array combined with PC-based software to create a highly-tuned acoustic beamformer steered in real time by an integrated binocular eye tracker. Recent work from our group on the two main functional sub-components of the VGHA (beamforming and eye-gaze control) focused on the benefits that may be obtained by listeners with hearing loss (and in some cases by listeners with normal hearing) attempting to understand speech masked by one or more competing sounds (such as Gaussian noise or additional speakers). These benefits include improved signal-to-noise ratio under different masked conditions and faster steering (compared to fixed directional amplification steered by head turns). The VGHA may be simulated over headphones by PC-based implementations using HRTFs and an optional external eye-tracker. This has allowed testing a number of variations of beamformer properties and the evaluation of multiple types of sound processing strategies including the incorporation of natural binaural cues. Overall, the VGHA holds considerable promise for improving selective listening in a variety of complex listening conditions.
An optimization model for electroencephalography-assisted binaural beamforming
Authors: Pu Wenqiang , Xiao Jinjun , Zhang Tao , Luo Zhi-Quan
Published: April 2018

In a multi-speakers noisy environment, the incoherent correlation between the electroencephalography (EEG) signal and the attended speech envelope provides a user-informed way to design the beamformer for preserving the attended speech and suppressing others. In this work, we exploit such incoherent correlation property in terms of Pearson correlation and propose a unified optimization model for simultaneously designing beamformer and aligning attention preference of the user to each speech source. To balance different design considerations, the proposed optimization formulation makes a trade-off among aligning attention preference, controlling speech distortion, and reducing noise in a weighted manner in the objective function. Specifically, the attention preference is aligned by maximizing the Pearson correlation between the envelope of beamforming output and the linearly transformed EEG signal. To control speech distortion with flexibility, the spatial response of the beamformer to each source is penalized in a min-max sense. And the noise is further reduced by minimizing its mean squares at beamforming output. Experiments on collected EEG signal in two speakers environment demonstrate the effectiveness of the proposed model.
Optimal frequency filtering of auscultation sounds
Authors: Nowak Lukasz
Published: April 2018

Most of the auscultation sounds do not reveal any significant single-frequency components, and their acoustic energy is concentrated in the low frequency region—up to about 100 Hz, falling even below the threshold of hearing. Such character is determined not only by the vibroacoustic behavior of sources, but mostly by high damping introduced by the sound transmission path through tissues underlying the skin surface. The contained diagnostic information is very subtle, and thus it can be easily masked by internal or external noise sources. Not all of those corrupting signals can be efficiently blocked, hence frequency filtering is the most obvious solution for improving the diagnostic capabilities. Many various filtering strategies and techniques were developed and implemented in both acoustic and electronic stethoscopes, however they are based primarily on (not always correct) intuition and subjective evaluation, without implementation of any accurate and objective measurement means or optimization algorithms. The present study introduces various signal to noise ratio (SNR) measures, applicable for different examination cases. Frequency filtering optimization strategies, for maximizing the values of the introduced coefficients for different heart and lung auscultation sounds, are presented.
The function of F0-based pitch
Authors: McPherson Malinda J. , McDermott Josh
Published: April 2018

Pitch is traditionally described as the perceptual correlate of a sound’s fundamental frequency (F0). The importance of perceiving F0 is often attributed to the need to compare sounds with different timbres, for which spectral comparisons might be error-prone. Alternatively, because the F0 is a summary measure of a set of harmonic frequencies, it could provide a compressed representation of sound for memory storage. To clarify the function of F0-based pitch, we tested its importance for discriminating sounds with different spectra, and sounds separated by delays, using both harmonic and inharmonic stimuli. If a task relies on F0-based pitch, performance should be impaired for inharmonic stimuli. We observed an advantage for harmonic stimuli for discrimination of synthetic tones with extreme (and unnatural) spectral variation. However, no such advantage was observed for instrument or speech fragments (with naturally occurring spectral variation). By contrast, discrimination was better for harmonic tones separated by brief (5–10 s) delays, suggesting that F0-based pitch aided storage across the delay. The results indicate that the spectral variation found in natural pitched sounds does not itself make listeners reliant on F0 estimation. The results raise the possibility that F0 primarily serves to facilitate compression of sounds for memory storage.
Primacy and recency effects in serial recall of synthesized vowel sequences
Authors: Shen Yi
Published: April 2018

Recognizing running speech involves not only audibility of speech signals but also cognitive processes that support the encoding and retrieving of speech information to and from memory. The current study aims to separate the effects of audibility and cognitive load on speech recognition performance. Synthesized vowel sequences were presented in noise, and the listeners were instructed to recall all vowels in the sequences in the correct order. In separate conditions, the sequence lengths were 2, 5, and 8 vowels, the presentation rates were 1, 2, 4, and 6 Hz, and the vowel durations were between 12.5 and 100 ms. The signal-to-noise ratios used in the experiment were adjusted for each listener and each of the vowel durations so that the expected recognition performance for isolated vowels was always 80%. A primacy effect was observed and it was more prominent at higher presentation rates and longer sequence lengths. A recency effect was also observed in most listeners, but no significant effect of presentation rate was found.
A study on the reducing stress of music listening by measuring heart rate
Authors: Kim Bong Young , bae myungjin
Published: April 2018

Humans are subject to many types of stress. Stress affects various nervous system transit hormone systems and weakens the immune system and weakens the autonomic nervous system, which slows the activities of various organs. It is already well known that those who overcome stress well have a strong immune system and overcome disease. It also achieves many successful results. This fact indicates how important it is to reduce stress. Music listening is one of the great ways to reduce stress. Music listening is known to stimulate the autonomic nervous system and the motor center to reduce complaints, edgy, and tension. Music listening also has an effect on mental treatment such as post-traumatic stress. In this paper, we tried to find out whether music listening helps to reduce stress through HRV (Heart Rate Variability) SDNN (Standard Deviation N-N Interval) obtained from heart rate measurement. Five subjects measured the heart rate by listening to four experiment song. The SDNN value increased when four of five people listened to all the experiment song. In the remaining one, the SDNN value was lowered for only one experiment song, and for the remaining three songs, the SDNN value was increased. However, the degree of reducing stress was different according to the musical preference of each individual. The results of this experiment showed that listening to music according to taste helps to reduce stress.
The effect of masker type and masker timing on the serial recall of speech
Authors: Roverud Elin , Best Virginia , Conroy Christopher , Kidd Gerald
Published: April 2018

Understanding speech in noise involves a complex set of processes including segregation of sources, selection of the target talker, and recognition and comprehension of the ongoing message. Previous studies often have focused on the segregation, selection and recognition processes by measuring the intelligibility of short strings of words presented concurrently with different types of maskers. However, these studies may not fully capture the effects of masking on all aspects of ongoing speech processing. For instance, masking can cause poorer recall for sequences of words even if a noise/competitor leads to near-perfect recognition of the individual items [e.g., Cousins et al., 2014, Mem Cogn (42)]. Furthermore, in contrast to the typical design of speech recognition tasks, background noise usually does not cease during response to the message. In the present study, we examined listeners’ serial recall of target speech with different maskers presented at different points during the trial. Speech-shaped noise or confusable speech maskers were presented colocated with or spatially separated from the target. The maskers were presented during target presentation, during the response interval, or continuously throughout the trial. Results will compare the influence of different types of masking during encoding and retrieval/response stages of processing.
Acoustic methods for regionalizing an impact force acting on a structure
Authors: Shepard W. S. , Davis Jacob
Published: April 2018

It is often desired to know the location and magnitude of a force acting on a structure. Unfortunately, it is not always possible to install a sensor at the force location, particularly when that location is unknown. A structure subjected to an impact has many different vibrational modes that are excited to different levels based on the excitation location. These vibrations decay with time depending on their different rates of modal damping and the associated acoustic radiation characteristics. In this work, acoustic signatures for a number of structure impact locations were measured, normalized relative to the force magnitude, and processed to examine the ability to correlate the acoustic signal to the force impact location. Various processing techniques, such as the Short Time Fourier Transform, were considered. A primary interest focused on the ability of single-number metrics to aid in identifying the force location. For the experiments, a football helmet structure was used and multiple impact locations were tested. The ability of these acoustic signatures, including those processed into single-number metrics, to aid in identifying the impact location was assessed. The location resolution for which these methods were effective was also evaluated.
Stop laryngeal contrasts of endangered languages of Northern Pakistan
Authors: Hussain Qandeel , Mielke Jeff
Published: April 2018

Iranian, Indo-Aryan (Dardic), and Tibeto-Burman languages of Northern Pakistan are characterized by 2-, 3-, and 4-way laryngeal contrasts at four or five places of articulation (e.g., Balti, Kalasha, Wakhi, etc.). However, the number of stop laryngeal contrasts, their phonetic realization, and the exact place of articulation of retroflex stops (apical or sub-apical) are still controversial in most of the endangered languages of Northern Pakistan. The current study provides the first detailed acoustic and articulatory (Ultrasound for coronals and velars and aerodynamic for all stops) description of the stop laryngeal contrasts of six languages of Northern Pakistan: Balti (Tibeto-Burman), Burushaski (Isolate), Kalasha, Khowar, Shina (Indo-Aryan, Dardic), and Wakhi (Iranian, Pamir). Given that there are languages belonging to different families, we may expect differences in the phonetic realization of stop laryngeal contrasts and the articulation of retroflexes across languages. The acoustic, articulatory, and aerodynamic data were collected in Chitral and Gilgit, Pakistan in 2017. A total of 33 participants were recorded (Balti: 5; Burushaski: 4; Kalasha: 9; Khowar: 5; Shina: 6; and Wakhi: 4). The results of this study will help understand the diverse nature of stop consonants in the endangered languages of Northern Pakistan.
Evidence for active cavity expansion through advanced lingual place of constriction in voiced geminates
Authors: Banerjee Ushasi , Dutta Indranil , Krishnaswamy Meghavarshini
Published: April 2018

Geminate voicing requires both active and passive cavity expansion in order to delay the equalization of subglottal and supraglottal pressure (Ohala 1983, Ham 2001). Studies report differences in stop closure duration (Lahiri & Hankamer 1988) and pre-geminate vowel duration as cues to the laryngeal contrast. Following Schroeder (1967), Mermelstein (1967), and Iskarous (2010), we present results from formant transitions in the VC:V context and the anti-symmetric Fourier component coefficient, an = -2 (Fn-fn)/fn, to partially recover the area function of the vocal tract behind the geminate constriction. Voiced geminates exhibit significantly higher and positive coefficient values than their voiceless counterparts which implies that measured Fn values for voiced geminates are lower than their neutral tract formant values. This finding indicates that voiced geminates have a more anterior constriction compared to their voiceless counterparts. Without ruling out the role of passive cavity expansion through the cheek muscles, we conclude that in order to delay the pressure equalization the lingual constriction in voiced geminates is maintained at an anterior location compared to voiceless geminates in Bangla. Results from the slopes of first order F2 locus-equations are also presented to measure the differences in coarticulatory resistance between the voiced and voiceless geminates.
Articulatory and acoustic investigation of coronals in Hakha Chin
Authors: Smith James , Flego Stefon , Berkson Kelly
Published: April 2018

Coronal obstruents are profoundly common typologically, occurring in most or all of the world’s known languages. Very few languages, however, contrast coronal obstruents at the dental and alveolar places of articulation. Hakha Chin—also known as Lai—is a Tibeto-Burman language spoken in western Myanmar that is reported to do so. Very little phonetic research on Chin exists, but Maddieson and Van Bik (2004) provide acoustic and static articulatory data (palatography and linguography) suggesting that the contrast is between a lamino-dental series and an apico-alveolar series. We follow up on that research using dynamic, volumetric articulatory imaging in the form of 3D/4D ultrasound. Articulatory and acoustic data from two native speakers of Chin are presented in order to contribute to a more thorough understanding of this typologically uncommon contrast.
Pre-coda vowel duration in Nepali
Authors: Schwarz Martha
Published: April 2018

Many previous studies have found that vowels are longer before voiced consonants than voiceless (Chen 1970). Fewer studies have examined vowel duration preceding aspirated vs. unaspirated stops, and results are inconsistent (Maddieson & Grandour 1976, Ohala & Ohala 1992). Durvasala and Luo (2012) find that both voicing and aspiration have a lengthening effect on the preceding vowel in Hindi, and that the voiced aspirated class has the longest preceding vowel duration. In the present study, I measure the duration of vowels preceding word final codas in Nepali, an Indo-Aryan language with the same four-way contrast as Hindi. A phonetic study of final stop realization has not yet been reported in the literature, though Nepali is described as variably neutralizing voiced aspirated to the plain voiced. I present an acoustic analysis of cues in word-final position including voicing, burst, and preceding vowel duration. I find evidence of neutralization of voiced and voiced aspirated stops in terms of aspiration duration, but also find results consistent with Durvasala and Luo (2012), that the voiced aspirated stops have longer preceding vowel than the voiced. The difference in vowel duration between the voiced and voiced aspirated classes suggests incomplete neutralization of voiced aspirated to voiced.
Do Hindi speakers of English realize the acoustic cues for American English /v/ and /w/ contrast?
Authors: Grover Vikas , Shafer Valerie , Whalen D. H. , Levy Erika , Kakadelis Stephanie
Published: April 2018

The Hindi phonological inventory does not include a [v-w] contrast. In a perception study, Hindi speakers identified /v/ and /w/ in American English (AE) words with approximately 50% accuracy (Grover et al., 2016). The present study compares AE /v/ and /w/ productions in bilingual Hindi-English speakers’ with those of AE speakers’ in nonsense words. Two AE listeners annotated recordings of word-initial and word-medial tokens to label F2 onset and presence of frication. F2 onset frequency and F2 slope were calculated. Hindi speakers were divided into two groups, those with length of residence (LOR) in the US of more than five years, and those who reside in India. Results revealed a significant difference between the AE group and both Hindi groups for F2 onset frequency and F2 slope. There was no significant difference between Hindi groups. The AE group had frication for /v/ tokens, but not for /w/. Both Hindi groups produced frication at about the same rate for /v/ and /w/. The finding of no significant difference between the Hindi groups indicates that exposure to the /v/-/w/ contrast in the US did not improve Hindi speakers’ production. This further indicates a need to design targeted training for this contrast.
Consonant-f0 interaction in Malayalam stops: Automatic or controlled?
Authors: Dutta Indranil , Varghese Molly
Published: April 2018

Voicing is known to perturb f0 in the following vowel (House and Fairbanks, 1953). This lowering has been attributed to biomechanical constraints (Svantesson and House, 2006). On the basis of evidence from languages where the [voice] feature does not manifest itself phonetically, yet is accompanied by f0 lowering, it has been argued that f0 lowering serves to enhance the phonological [voice] contrast. We present results from an acoustic study of f0 perturbation in Malayalam, which exhibits predictable voicing in lexical items from its Dravidian stratum. Underlying voiceless stops only appear initially, and are voiced intervocalically. Malayalam also contains a large Sanskrit stratum lexicon, where phonetic voicing is tied to its stratal affiliation. We obtained a significant effect of laryngeal setting on z-score normalized f0 at vowel onset. These results suggest voicing is significant in determining the onset f0 of the vowel, regardless of the phonological status of voicing in a language We conclude that f0 lowering has physiological and biomechanical bases. In that sense, f0 lowering is an automatic consequence of the biomechanics of voicing. At the representational level, such lowering could indeed be enlisted for enhancement, however, that is not the only, or the primary purpose of f0 perturbation.
Effect of contrastive focus on vowel duration in Bangla
Authors: Ghosh Bitapi
Published: April 2018

The present study looks at Bangla, spoken near Kolkata, West Bengal, which has got the status of standard Bangla in present era, belongs to the Indo-Aryan branch of the Indo-European language family. Vowel length distinction is not phonemic in Bangla. However, the impact of various prosodic positions and information structure is quite obvious on the phonetic duration of the vowel. This paper presents an acoustic study of the effect of Contrastive focus on the duration of the high vowels /i/ and /u/ in Bangla. Six monosyllabic and six disyllabic words with the target vowels, altogether 24 words have been recorded in both Contrastive focus and Neutral focus contexts. 24 sentences as response to yes-no questions and for Neutral focus, 24 simple declarative sentences were recorded from six speakers of Standard Kolkata Bangla between age group of 20-25. In Bangla, Focus is realized by low rising pitch pattern where the f0 movement have low f0 valley followed by a rise (Hayes and Lahiri 1991). The present study throws light on the other acoustic cue. The statistical analysis of the results from the study show significant difference in vowel duration of the focussed segment.
Glottal phonation in Punjabi: Tone or aspiration
Authors: Sanyal Paroma , Midha Manujata
Published: April 2018

This study investigates a puzzle in the glottal phonation of the South Asian Language Punjabi. Though phonetically absent from the spoken language, voiced aspirated plosives (VAP) are still perceived as a distinct phoneme by native speakers. This phonological identification is achieved by the introduction of rising and falling tones in complementary phonological environments. Specifically, the VAP lose both voicing and aspiration in word-initial positions and yet speakers are able to discern the expected phoneme due to the epenthesis of a falling tone. Elsewhere, the VAP is phonetically indistinct from the corresponding voiced plosives, but for the rising tone on the same syllable. The vowel bearing the inserted tone in Punjabi must also be stressed. Consequently, the underlying VAP and the corresponding tones can be heterosyllabic. However in such scenarios, the tones associated with the position of VAP is reversed. We propose that an attempted glottal friction is linked to the introduction of the rising and falling pitch contour. In turn, this tone is also correlated with stress, which phonetically corresponds to low pitch in Punjabi.
Performance of WUW and general ASR speech recognition systems in different acoustic environments
Authors: Bohouta Gamal , Këpuska Veton
Published: April 2018

The Automatic Speech Recognition (ARS) Systems have become popular in the different areas of applications, such as education assistance, medical assistance, personal assistant, robotics and vehicles, telecommunications Industry, disabilities systems, home automation, and security access control. All these applications in life will increase the number of speech commands, which are Wake-up-words. The WUW speech recognition is similar to Key-Word spotting. However, in this paper, we propose an approach that will be used to detect the Wake-up-word (WUW) as a command with 100% accuracy and to provide a whole speech recognition system that can work in both WUW and General ASR systems with high accuracy. Also, this new ASR system will be used to solve one of the biggest problems in speech recognition, which is how to discriminate between the uses of a word or phrase in an alerting and a referential context. For example, in the use of the word “Computer'' in an alerting context is "Computer, show me the chart? '' and in a referential context it is “Every computer should have a speaker. Moreover, we tested the new system with different acoustic environments, such as different background noise levels, different speaker distances to the microphone, and different speakers.
Noise adaptive real-time estimation of infrasound using a compact array
Authors: Frazier William G.
Published: April 2018

Previous investigations [Frazier, ASA Pittsburg (2015); ASA San Francisco (2013); ASA Kansas City (2012)] have demonstrated that exploiting the correlated statistical properties of wind noise on short spatiotemporal scales with a compact array permits enhanced detection and reduced estimation error of infrasound transients resulting from explosions. The approach was shown to be effective in real-time using non-parametric wind noise models. More recently [Frazier, ASA Boston (2017); ASA Salt Lake City (2016)], the basic ideas of the approach were extended to estimation of quasi-stationary infrasound signals such as microbaroms using parametric wind noise models and was demonstrated to be effective, but the computational requirements were too demanding for real-time implementation. This presentation will highlight modifications that enable real-time implementation for quasi-stationary signal estimation.
Dimension-reduced adaptive pulse compression with model error compensation
Authors: Botts Jonathan , Summers Jason E. , Bentrem Frank W. , Gaumond Charles F.
Published: April 2018

In active acoustic imaging, range sidelobes from competing clutter and interferers in acoustically complex environments may mask signals of interest. In order to unmask these signals, increase range resolution, and decorrelate the time-series we introduce dimension-reduced adaptive pulse compression (APC) with model-error compensation that operates after a conventional matched filter. Structured covariance estimates minimize the required sample support while dimension reduction occurs naturally and a priori after the matched filter. We compensate for predictable signal distortions such as Doppler through both covariance matrix tapers and truncation of the covariance matrix. Model-based APC is computationally intensive, scaling with the third or fourth power with replica length, so algorithmic modifications are introduced to reduce computation by orders of magnitude. We evaluate the algorithm's robustness and computational feasibility on simulated and archived data. [Portions of this material are based upon work supported by the Naval Sea Systems Command.]
Processing of emitted respiratory noises to monitor health condition and spatial displacement of a scuba diver under water
Authors: Korenbaum Vladimir , Gorovoy Sergey , Kostiv Anatoly , Shiryaev Anton , Borodin Aleksey , Dorozhko Veniamin , Fershalov Andrey
Published: April 2018

Monitoring health condition of a scuba diver and his displacement within the exploited water area is necessary to ensure the safety of diving. Respiratory noises, emitted by the diver into water, may be used for this purpose. In the hydroacoustic basin, powerful signals, having repetition frequency of about 0.12 Hz are found, which correspond to diver’s respiratory rate. The main broadband signal of exhalation is concentrated in the frequency band of 150–1150 Hz. The high-frequency signal in the frequency band of 3.5–4.7 kHz is associated with inspiration. Thus, acoustic estimating the respiratory rate and the ratio of the inspiration/expiration durations are possible, which are important physiological parameters to assess a health condition of the diver. When registering in sea respiratory noises of scuba diver against the background noise, it is possible to trace the acoustic signs of respiratory maneuvers associated with the noise of floating bubbles in the spectrograms at distances up to 100–200 m. The same acoustic signs provide monitoring the displacement of a scuba diver by determining the delays of the maxima of the cross-correlation function at 2 hydrophones. Trails of the delays in correlograms are traced at distances up to 300 m.
Experiments with the Underwater Bubble Low Frequency Sound Resonator
Authors: Morozov Andrey K. , Webb Douglas C.
Published: April 2018

An infra-sound source with a resonator in the form of an underwater bubble made from an elastic material is a simple engineering solution. The elastic membrane supports high volume displacement and a large radiation aperture. The Rayleigh-Plesset equation defines the dynamics of a spherical bubble including surface tension and viscous effects. The shape of a large bubble deforms due to gravitational effects. Its internal pressure oscillations are comparable with the difference of gravity pressure and these effects are part of its dynamics. The real Q-factor of a practical bubble is smaller than theoretical. The underwater bubble resonator was studied experimentally in the pool and at the Woods Hole Oceanographic Institution’s dock. The research includes the 3D finite-element analysis and its comparison with the experimental data. The study gave the real values for the Q-factor, resonance frequencies and sound pressure levels. The experimental resonator has good performance with a maximum SPL 190 dB at the resonance frequency 6-12 Hz. It was shown that a cylindrical resonator can be towed with a speed of 8 knots. The research proved that this is a practical approach for a coherent low frequency sound source, which can find applications in a marine seismic survey.
Creating realistic design goals through the use of locally-measured reverberation time and background noise data
Authors: Whitney Matt , Pyper Ted
Published: April 2018

One of the major difficulties in the design process of an atrium or lobby is providing points of reference to both the architect and the end user to help drive the conversation on an acceptable reverberation time and background noise level for the projected use cases. K2 recently surveyed eight different atrium and lobby spaces in Denver and used this reverberation time and background noise data to develop an appropriate range of reverberation times, for a given room volume, covering most of the common uses for atriums and lobbies. These ranges help provide guidance to both the design team and the end user on design goals and expectations for these spaces. This presentation will present these data, as well as subjective impressions and architectural design of each of the spaces, and describe how K2 used this information to initiate productive conversations with architects and end users and produce realistic design goals.
Inversions for sediment parameters using geophone and tetrahedral array measurements in the New England Mud Patch
Authors: Potty Gopu R. , Miller James H , Lin Ying-Tsong , Bonnel Julien , Newhall Arthur
Published: April 2018

Measurements of acoustic pressure and particle velocity were made during the Seabed Characterization Experiment (SCEx) in the New England Mud Patch south of Cape Cod in about 70 meters of water. The University of Rhode Island and Wood Hole Oceanographic Institution deployed the “geosled” with a four-element geophone array, a tetrahedral array of four hydrophones, and several hydrophone receive units (SHRUs). In addition, a new low frequency source, interface Wave Sediment Profiler (iWaSP) was deployed to excite interface waves (Scholte waves). The iWaSP system consists of a source to generate the interface wave and a four-element accelerometer receive array. Results of inversions for geoacoustic parameters will be presented. Arrival time dispersion from broadband sources (SUS and CSS) and iWaSP will be used for these inversions. Both p-wave and Scholte wave arrivals will be investigated using the geophone and hydrophone data. Seismic data collected at the location will be used to constrain the model parameters in the inversion. Sediment data from cores, other in-situ measurements and inversions using other types of data will be used to compare and validate our inversions. [Work supported by Office of Naval Research.]
Efficient computational algorithms for modeling guided wave dispersion in arterial walls
Authors: Vaziri Ali , Urban Matthew W. , Aquino Wilkins , Greenleaf James F. , Guddati Murthy
Published: April 2018

Arterial stiffness is a well-known biomarker of early cardiovascular disease. Shear wave dispersion ultrasound vibrometry (SDUV) has emerged as a promising technique to estimate local arterial stiffness from the observed dispersion of guided waves. With the ultimate goal of developing real-time inversion for arterial stiffness from SDUV measurements, we develop and validate highly efficient and accurate computational algorithms that compute the wave dispersion in multi-layered immersed tubes. The proposed approach carefully combines Fourier transformation and one-dimensional finite-element discretization to accurately capture fully three-dimensional wave propagation. The method is several orders of magnitude more efficient than three-dimensional finite element simulation, and eliminates other complexities such as the need for absorbing boundary conditions. The method is validated using SDUV experiments on tissue-mimicking phantoms. The validation exercise uncovered an important detail that is often overlooked—the dispersion curve captured through SDUV experiments does not correspond to a single dispersion curve, but a combination of multiple dispersion curves. This implies that proper identification of the dispersion curves could be critical to estimating the arterial stiffness. In this talk, we present the details of the proposed method including formulation, computational complexity, verification and validation.
Characterizing anisotropic, viscoelastic material properties of the tectorial membranes of wild-type and mutant mice using longitudinally propagating waves
Authors: Lemons Charlsie , Sellon Jonathan B. , Ortiz Roberto D. , Boatti Elisa , Freeman Dennis M. , Meaud Julien
Published: April 2018

The tectorial membrane (TM) is an important structural component of the mammalian inner ear. Examination of cochlear physiology in genetically modified mice has demonstrated that mutation of genes affecting TM proteins causes changes in key characteristics of cochlear function. Characterizing the differences in material properties between wild-type and mutant mice could provide insight into the source of reported differences in cochlear physiology. In this study, optical images of isolated TM segments of wild-type and genetically modified mice in response to harmonic radial excitation at acoustic frequencies are used to determine the amplitude and phase of the motion. Wave propagation on the TM segments is modeled using finite element models that take into account the anisotropy, viscoelasticity, and finite dimensions of the TM and the presence of a viscous boundary layer. An automated least-square fitting algorithm is used to find anisotropic, viscoelastic material parameters of wild-type and mutant mice TMs at acoustic frequencies. The resulting material properties are compared to previous estimates of the TM properties.
Fingertip elastography
Authors: Khan Altaf , Royston Thomas J.
Published: April 2018

Dynamic elastography methods attempt to quantitatively map soft tissue viscoelastic properties. Application to the fingertip, relevant to medical diagnostics and to improving tactile interfaces, is a novel and challenging application, given the small target size. In this feasibility study, an annular actuator placed on the surface of the fingertip and driven harmonically at multiple frequencies sequentially creates geometrically focused surface waves. These surface wave propagation patterns are measured using scanning laser Doppler vibrometry. Reconstruction (the inverse problem) is performed in order to estimate fingertip soft tissue viscoelastic properties. The study identifies limitations of an analytical approach versus an optimization approach that utilizes a finite element model. Measurement at multiple frequencies reveals limitations of an assumption of homogeneity of material properties. Identified shear viscoelastic properties increase significantly as frequency increases and the depth of penetration of the surface wave is reduced, indicating that the fingertip is significantly stiffer near its surface. Additional studies using Optical Coherence Tomography (OCT) and Magnetic Resonance Imaging (MRI) provide insights into wave propagation at near sub-surface and deep sub-surface zones.
Articulatory modeling of human and non-human vocal production
Authors: Honda Kiyoshi , Zhang Ju
Published: April 2018

This paper reviews two of the many questions that might be addressed by adapting articulatory models of vocal production across primate species. The first concerns vocal-fold shaping at phonation. The human folds have been modeled to have uniform cross-sectional shapes with round edges along the longitudinal axis, whereas MRI slices reveal an obvious vertical prominence of the mucosa near each vocal process. This shape, due to depressed vocal processes in adduction, somewhat resembles the pointed edges of the vocal membranes in nonhuman primates, which are incorporated in some models of chaotic phonation in those species. This suggests a model of vocal-fold vibration propagating back to the posterior glottis, or coupled oscillation for irregularities in human voices. The second concerns tongue-lip coordination. In human vowel inventories, tongue retraction typically accompanies rounding. In a case report, a trained Chimpanzee was able to produce back vowels /a, o, u/ in whispering, but could not coordinate these articulatory positions with voiced phonation. This suggests an innate neural circuit for patterned tongue-lip coordination as seen in sucking, and such a basic neural unit may be active in humans to simplify the articulatory control of back rounded vowels. [Work supported by NSFC No. 61573254.]
Localization of sources of noise in moving atmospheres using time reversal and geometric acoustics
Authors: Lonzaga Joel B.
Published: April 2018

Due to acoustic refraction and convection in moving, inhomogeneous atmospheres, the backazimuths of recorded signals can be significantly different from the source backazimuth, posing a significant difficulty in source localization. Although an inverse problem, localization of sources of infrasound in such atmospheres has mostly been made using the forward ray-theoretic model of sound propagation. This approach uses search grids for the unknown model parameters that are subsequently estimated by optimizing an objective function. This approach can give rise to a large computational time as well as to the issue of uniqueness, stability, and accuracy of the solutions. The current paper presents a technique which avoids these issues, and which combines time reversal and geometric acoustics for source localization in such atmospheres. This technique, which can also be useful for source localization of other forms of acoustic signals in refractive media, effectively transforms the inverse problem into a forward one by utilizing a flow reversal theorem in the ray tracing equations. Beside the atmospheric conditions, the only parameters needed in this technique are the trace velocities of the signals recorded by at least two arrays. Comparison of numerical simulations with observational data will be presented.
Understanding the effect of room parameters on acoustic time reversal
Authors: Denison Michael , Anderson Brian E.
Published: April 2018

Time Reversal (TR) is a technique used to focus an acoustic signal at a particular point in space. Of the many variables that contribute to the quality of TR focusing, the most important are the number of sound sources, signal bandwidth. and properties of the medium. Although much research has been done to quantify the effects that the number of sound sources and signal bandwidth have on the TR process, little has been done in regard to the effect that room parameters have on TR. This is largely due to the difficulty involved with changing room acoustic parameters to measure their effects. We use the image source method (using the algorithm proposed by Allen and Berkley) to simulate the TR process in a variety of rooms with different acoustic and geometric room parameters. We define and calculate the maximum focal amplitude, the temporal focus quality and the spatial focus quality for each simulation. We compare the results and determine the effects of absorption and room volume on TR. We find that less absorption improves max response and spatial quality while it decreases temporal quality and that larger volumes have decreased max response and spatial quality while having increased temporal quality.