bluuebunny/crossref_metadata_2025 huggingface dataset:
Experimental study on the acoustic field caused by the interference of supersonic jet with a perforated plate—The optical measurement of sound absorption
Authors: Murata Yo , Murayama Kaku , Ishii Tatsuya , Okamoto Koji , Daiguji Hirofumi , KANEKO SHIGEHIKO
Published: November 2019

The supersonic jet is known to be a source to high intensity sounds. According to previous studies, when this jet collides with a perforated plate, a complex flow field is formed, and sound waves with much higher amplitudes are generated. In our study, an optical visualization test based on the Schlieren method was carried out, using a simple model consisting of a CD nozzle and a perforated plate. During the test, a sound-absorbing material was flush-mounted in the plate as an approach to alleviate the acoustic field. Visualized videos tried to account for absorption of high-amplitude sound waves by the sound-absorbing material. The videos were analyzed and compared with acoustic measurement for verification.
Entropy noise from shock containing nozzles of military jets at afterburner
Authors: Tam Christopher
Published: November 2019

This paper considers the noise of military jets operating at afterburner. Boosting by the afterburner, the jet velocity and temperature are much higher than those of laboratory jets. Under these conditions, do we anticipate the noise components of these jets the same as those of laboratory jets, i.e., consisting mainly turbulent mixing noise? Do we expect new noise sources in these jets, since they operate at much higher velocities and temperatures and that the combustion processes in the afterburner are highly unsteady. In this paper, we explore one plausible new noise mechanism; the generation of entropy noise inside the shock containing nozzles of military jets. It is well-known that hot and cold spots (entropy waves) from unsteady combustion when convecting through a non-uniform mean flow would create fluctuating pressure and hence acoustic radiation. We investigate this possibility by performing numerical simulation of the flow and noise inside a rectangular nozzle typical of that of the F-22 Raptor. These nozzles are imperfectly expanded. There are shocks inside. We find that because shocks have very steep velocity gradients, their presence leads to the generation and emission of strong entropy noise when entropy waves from the afterburner pass through them.
Analysis of turbulence in convergent-divergent nozzles with fluid insert noise reduction technology
Authors: Prasad Chitrarth , Morris Philip
Published: November 2019

The noise generated by tactical fighter aircraft engines is harmful to personnel working in the vicinity of the aircraft and can cause annoyance to communities in the vicinity of air bases. The primary noise sources in supersonic jets are related to the supersonic convection of large-scale turbulent structures, which generate intense noise in the jet downstream direction, and the interaction of the turbulence with the jet’s shock cell structure that results in broadband shock-associated noise, which radiates predominantly in the sideline and upstream directions. The use of fluid inserts in the divergent section of variable area exhaust nozzles has been shown experimentally, at small and moderate scale, to reduce noise radiation from both noise sources. To understand the fluid insert noise reduction mechanisms, a Large Eddy Simulation database is developed and analyzed. The focus is on differences between turbulence properties in a baseline and a fluid insert nozzle. The flow database is analyzed using Spectral Proper Orthogonal Decomposition and Doak’s Momentum Potential Theory. The latter theory permits the decomposition of the flow into hydrodynamic, thermal and acoustic components. The changes in the acoustic component are related to the observed changes in the radiated noise.
Ultra-compact ultrasonic metalens for underwater focusing
Authors: Chen Jian , Fan Zheng
Published: November 2019

There are significant challenges to design acoustic metasurface in the ultrasonic regime where the wavelength is small. Moreover, the routine design rule based on effective medium approximation is invalid for underwater ultrasound. Our recent development of metasurface provides a great opportunity to solve above-mentioned challenges and the results demonstrate the great potential to apply this metasurface in the field of ultrasonic focusing. Here, we presented a focusing metasurface lens with extreme simplicity (slot structure) and ultra-compact size (deep-subwavelength spacing and thickness). Instead of tuning the acoustic path length individually, we exploited the strong wave couplings between the deep-subwavelength-spaced units for the phase modulation. A microscopic coupled-wave theory was used to predict the phase profile, based on which the metasurface lens for ultrasonic focusing was optimized. Particularly, broadband focusing was also exhibited due to the non-resonant arrangement of the proposed metasurface lens. The numerical and experimental results agreed well with each other, effectively validating the feasibility of the proposed metasurface lens. The proposed approach provides a feasible avenue for the design of simple and ultra-compact ultrasonic lenses that would be suitable in various fields of ultrasonics.
Aqueous acoustic metasurface for the anomalous reflection of sound
Authors: Sieck Caleb F. , Martin Theodore P. , Wissman James P. , Ikei Alec , Rogers Jeffrey S. , Guild Matthew D. , Rohde Charles
Published: November 2019

Ultrathin structures known as acoustic metasurfaces offer the same ability to achieve extreme properties as acoustic metamaterials, yet have the added potential benefit of being very thin (much less than a wavelength in thickness). In this work, we discuss an aqueous acoustic metasurface that utilizes subwavelength non-resonant embedded cavities to achieve anomalous reflection of incident acoustic waves impinging over a wide range of angles. The metasurface was constructed using air-filled cavities in a low-shear matrix material and designed to shift the angle of the reflected wave by 30 deg. The use of a low sound speed matrix material and non-resonant cavities enabled operation over a wider range of frequencies and incident angles while maintaining subwavelength thickness. The cavity heights were varied to achieve 0 to 2π phase shifts in a unit cell according to the generalized form of Snell’s Law. The fabricated structure was mounted on a brass plate and experimentally tested in water. Very good agreement was found between finite element models and the experimental ultrasonic tank tests of the aqueous acoustic metasurface. [Work supported by the Office of Naval Research.]
Broadband monostatic acoustic scattering simulations of underwater unexploded ordnance using time-domain spectral-elements
Authors: Bottero Alexis , Williams Earl G. , Kuperman William A. , Rakotonarivo Sandrine
Published: November 2019

Acoustic detection of unexploded ordnance (UXO) that contaminate the world’s waterways is vital. Critical to the implementation of detection methods is the numerical simulation of the scattering response of these targets using finite element methods. We introduce a parametric approach based on the time-domain spectral finite element method (SEM). This technique allows for the computation of broadband acoustic response of complex heterogeneous 3-D fluid-solid objects, in particular a 5 inch rocket UXO used for this study. The scattered field is obtained at arbitrary distances using the Kirchhoff–Helmholtz integral. The main interest of the SEM is that it is particularly adapted to high-performance computing (CPU or GPU). Contrary to most of the other finite element methods it scales perfectly; using twice as many processors leads to a halving of computing time. In addition, working in the time domain is a direct simulation of common sonars and experiments. The method is first benchmarked against the commercial finite-element code COMSOL on the monostatic response of a rigid target over a full 180 deg. Finally, results obtained for the 5 inch rocket are compared to actual measurements obtained in the NRL underwater acoustics tank facility. [Work partially supported by the Office of Naval Research.]
A substructuring approach based on mechanical admittances to solve vibro-acoustic problems in the mid-frequency range
Authors: Meyer Valentin , Laurent Maxit
Published: November 2019

In the low frequency range, discretization methods such as the Finite Element Method are the preferred techniques to solve vibro-acoustic problems. The main limitation of such approaches is that the mesh needs to be refined when the frequency increases, leading to prohibitive calculation costs. This paper presents the Condensed Transfer Function (CTF) method for substructuring vibro-acoustic problems. The problem is split in several structural, acoustic or vibro-acoustic subsystems that are studied individually before being assembled. The vibro-acoustic behavior at the boundaries of the subsystems is expressed in terms of mechanical admittance for structures and acoustic impedance for fluid domains. A set of condensation functions is used as a basis to express the values at the boundaries. The coupling between subsystems is written through the continuity equations of mechanics. Compared to classical reduction methods such as Craig-Bampton, this approach does not require the prior knowledge of the modal basis of the subsystems. Also, the CTF method allows coupling subsystems that are described by different techniques, i.e., analytically, numerically or experimentally. The principle of the method will be recalled and example of applications will be given. In particular, numerical and experimental results on a stiffened cylindrical shell will be compared.
Dynamical energy analysis applied to real-world structures—New developments
Authors: Richter Martin , Tanner Gregor , Chappell David J.
Published: November 2019

Dynamical Energy Analysis (DEA) has been established as a mesh-based ray tracing method working at mid-to-high excitation frequencies for structure-borne sound on complex vehicle structures. Unless standard ray-tracing used, for example, in room acoustics and following long rays through a series of specular reflections, DEA is based on an integral representation by (a) reformulating the energy transport problem in terms of ray densities and linear matrix equations and by (b) applying the ray tracing directly on preexisting meshes used also in, for example, FEM calculations. The advantage is that DEA can be used as a drop-in replacement for FEM tool working on the same meshes but determining results at very high frequencies. In order to achieve this, a so-called transfer matrix has to be created and later solved for the energy equilibrium distribution. In this presentation, we address recent developments in making DEA an efficient black-box tool for mid- to high-frequency vibrations including convergence issues and suitable preprocessor applications, sound radiation from DEA calculations as well as verification and validation studies.
Using H1 instead of H1–H2 as an acoustic correlate of glottal constriction
Authors: Chai Yuan , Garellek Marc
Published: November 2019

Bickley (1982) proposed H1–H2 as an index of the stronger H1 observed for breathy vowels compared to modal ones. Because harmonic amplitudes also vary by overall sound pressure level (SPL), H1 amplitude was normalized to that of H2. Studies have also shown that H1–H2 is well perceived and correlates with glottal open quotient, and thus can be used to infer the articulations leading to changes in perceived quality. Yet in some cases, H1–H2 does not distinguish two voice qualities at statistically significant levels. We claim that this might be due not to the same open quotient between two voice qualities, but to the fact that H1–H2 can be especially noisy to estimate: for example, when f0 is irregular (as in creaky voice qualities), H1–H2 often has large variance that isn’t well modeled statistically. Using several corpora, we show that H1 alone (while controlling for SPL) allows for stronger statistical differentiation of voice categories than H1–H2. We further confirm that H1 can stand in place of H1–H2 as a correlate of glottal constriction. When differentiating voice qualities acoustically, researchers should consider measuring H1 (while controlling for differences in overall SPL) instead of, or in addition to, H1–H2.
The influence of speech prosodic structure and syllable composition on consonant VOT in Japanese
Authors: Chen Xi S. , Chen Si , LI Bei
Published: November 2019

This research elaborates the joint influences of the position relative to the accented prosodic component and syllable composition on the stops VOT in Japanese. Monosyllables and disyllables start with three types of plosives, namely bilabial, alveolar, and velar, are studied on different positions in prosodic sentences—pre-focus, on-focus, and post-focus. Results show that narrow and contrastive focus prolong the voiced on-focus stops (including bilabial, alveolar, and velar) which bear positive values in broad-focus sentences; but compress the on-focus voiceless consonants (including bilabial, alveolar, and velar) which have negative values neutrally. On the position of pre-focus, voiced consonants shorten the VOT towards to zero, especially for the velar voiced consonants; the voiceless ones follow this pattern exactly. On the position of post-focus, noticeably, the voiced and the voiceless plosives always show the opposite patterns—both of narrow and contrastive focus elicit a longer VOT on following voiced stops but reduce the VOT of voiceless consonants following them. The voiced alveolar show the most apparent compression at pre-focus, which always close to zero. Kinds of results describe significant interaction influences among different factors that bear the potential to change consonant VOT in Japanese.
Centering diphthongization in Hui Chinese dialects
Authors: Hu Fang , Zhang Minghui
Published: November 2019

This paper reports centering diphthongization in three Hui Chinese dialects. It argues that centering diphthongization serves as clear evidence that monophthongs and diphthongs are not a dichotomy, but a continuum, and thus supports a dynamic theory of vowel production (Hu, 2017). Centering diphthongs in Hui dialects occur as an intermediate vowel category between monophthongs and diphthongs. Fine-grained phonetic details reveal that the process of centering diphthongization is gradient in the three Hui dialects. The Yi county dialect is at an early stage: centering diphthongization is essentially a phonetic process of adding an additional neutralized [ɐ]-like production to the distinctive vowel element, namely [i u y e o] > [i:ɐ u:ɐ y:ɐ e:ɐ o:ɐ], respectively. The Qimen dialect is at a phonologically developed stage: monophthongs and diphthongized vowels contrast in dynamics, namely the static [i u y] versus the dynamic [i:ɐ u:ɐ y:ɐ], but diphthongized vowels have a temporal structure different to that of plain diphthongs. The Xiuning dialect is at a final stage: monophthongs and diphthongized vowels contrast in dynamics, and diphthongized vowels exhibit a temporal organization similar to that of plain diphthongs.
Acoustic characteristics of belching in speech
Authors: Kidner Brooke L.
Published: November 2019

“When such nonspeech sounds occur connected in speech they sometimes carry important social connotations (Pike, 1943: 39).” This sentiment is attested over and over again in linguistic literature that provides cases of nontraditional sounds (such as coughing, moans, cries) being co-opted into the continuous stream of speech within discourse (Ohala,1995; Ogden, 2013; Prelock and Hutchins, 2018; Pinto and Vigil, 2018). This reveals an unanswered problem: While many of these sounds have been able to be expressed by their phonetic parameters in paralinguistics (Trager, 1958; Poyatos, 1975), belching—as evident from a case study as being intentionally interjected into the speech stream with social connotations (Kidner, 2018)—has consistently eluded detailed description using acoustic parameters. Acoustic data were taken from this case study, and belching was distinguished in Praat (Boersma and Weenik, 2018) from modal speech by an increase in glottal pulses and jitter (>4.4%), an increase in shimmer (>15%), low pitch (≤300 Hz), and an irregular amplitude contour. These criteria for belch identification (while not exhaustive) allow us not just to classify belches phonetically within the corpus, but moves the process of acoustically defining otherwise difficult paralinguistic sounds forward.
Effects of stop voicing on long-distance acoustics
Authors: Lipani Lisa
Published: November 2019

In Coleman (2003), the phonological feature [voice] was found to have a long-distance coarticulatory effect. However, voicing is not always realized as categorically present or absent. Davidson (2016) classifies partially voiced stops as having different patterns of voicing: bleed, trough, negative VOT, and hump. The present study answers the question: How do voicing patterns in stops affect the long-distance coarticulatory properties of the stop? Following the methodology of Coleman (2003), LPC coefficients, voicing, F0, harmonics-to-noise ratio, and z-score normalized RMS amplitude were measured in the read data from the Rainbow Passage in the Nationwide Speech Project (Clopper and Pisoni, 2006) in order to examine the stops and their coarticulatory effects. In the second sentence of the Rainbow Passage, the phones [b], [d], and [k] and their surrounding acoustics were measured in 10 millisecond frames that were aligned in time using dynamic time warping. A preliminary analysis using data visualization with smoothed conditional means reveals differences in the acoustics. There were several differences in both anticipatory and carryover coarticulation in the first through sixth and ninth through eleventh LPC coefficients. Subtle differences in acoustics as a result of long-distance coarticulation have implications for what makes both human and synthetic speech sound natural.
Oral-velum actions in the articulation of nasal juncture geminates and singletons
Authors: Oh Miran , Byrd Dani
Published: November 2019

Articulatory studies of consonant production generally focus on oral gestures, without further attention to the action of non-oral gestures—such as velum gestures. In large measure, this lacuna has arisen due to a lack of instrumental data. This study deploys real-time MRI video to investigate nasal consonant production in Korean, allowing us to observe spatiotemporal aspects of both oral and velum gestures. We examine the articulatory dynamics of concatenated and assimilated juncture geminate nasals (/n#n/ & /t#n/ → [n.n]) in comparison with singleton nasals (/#n/ & /n#p/). We predict that geminates and singletons differ in the duration of oral and velum gestures and that concatenated and assimilated geminates show different articulatory characteristics. Findings indicate that Tongue Tip (TT) duration is longer in both geminate types relative to singletons, but Velum (VEL) lowering duration does not consistently distinguish geminates from singletons, though its plateau duration does. Further, both TT constriction degree and VEL lowering displacement differ for assimilated geminates as compared to concatenated geminates and singleton codas, with greater TT magnitude and lesser VEL lowering in assimilated geminates. Results suggest that velum and oral components do not behave in parallel in the production of these nasal consonant sequences. [Work supported by NIH.]
Acoustic correlates of Apurímac Quechua ejective stops
Authors: Marcinko Mackenzie , Ojeda Ludeña Jermani , Alshahrani Abdulrhman
Published: November 2019

Apurímac Quechua ejective and pulmonic stops are compared with respect to VOT, rise time (the following vowel’s peak amplitude – onset amplitude), and F0 difference (following vowel’s midpoint amplitude – onset amplitude). These three properties distinguish ejective from pulmonic stops in other languages (e.g., Ingush) but were previously unanalyzed in Quechuan languages. Results show that only rise time and VOT distinguish Apurímac Quechua ejective stops. All syllables (stressed and unstressed) of CV.CV.CV words were analyzed. The vowel /a/ followed each stop onset. 60 pulmonic plus 40 ejective stops were compared (all places of articulation) (α = 0.05). Rise time is the strongest acoustic cue for ejective stops: vowels following ejective stops had significantly longer rise times (meanejective= 7.74 dB, SD = 1.27) than pulmonic (meanpulmonic = 4.85 dB, SD = 2.94). Ejective stops therefore rise to peak amplitude more slowly than pulmonic. F0 difference did not significantly differ between vowels following each stop type (meanejective = -0.32 Hz, SD = 0.05; meanpulmonic = 0.40 Hz, SD = 0.03). Ejective stops had significantly longer VOTs (meanejective = 27.82 m/s, SD = 2.01) than pulmonic (meanpulmonic = 23.98 m/s, SD = 4.09). This difference is small, indicating that VOT is a less-robust acoustic cue for ejective stops.
Effects of alcohol and emotions on voice features in Mexican Spanish accented speech: A forensically relevant case study
Authors: Miglio Viola G.
Published: November 2019

This is a case study of how alcohol intoxication and emotions affect the speech of a male Mexican Spanish speaker arrested on suspicion of vehicular manslaughter, and how his dialectal variety may have influenced the assumption of the arresting officers that alcohol intoxication was revealed by his altered speech patterns. There are still few standards in Forensic Linguistics and Voice Recognition/Identification (cf. the survey by Hollien et al., 2014). We do know, however, that listeners often overestimate the amount of alcohol speakers ingested, sober speakers can successfully mimic drunken speech, and conversely it is possible for a speaker under the influence to sound more sober than s/he actually is by modifying certain parameters (ibid., p. 180). It has also been demonstrated that dialectal differences can impair intelligibility and speaker recognition (Bahr et al. 2002, Betancourt and Bahr, 2010): the present case study analyses the acoustic segmental and suprasegmental aspects of the existing recordings of the arrest, and shows that while the speaker may have been physiologically drunk, he was rather “linguistically sober.” Further correlations are also drawn between dialectal characteristics and “slurred speech,” as well as the influence of emotions and possibly deception on speech rhythm and variations in vocal parameters.
Shannon-Zipf comparison of human and humpback whale voiced phonetic and sub-unit complexity
Authors: Pines Howard S.
Published: November 2019

The striking similarities of time-frequency spectrograms of voiced human speech and humpback whale vocalizations indicated a common targeted, frequency-modulated phonetic/sub-unit basis. To map the sub-unit structure of voiced humpback whale song units, a time-frequency-log-power contour segmentation, extraction, and classification procedure was tested on voiced human speech and then applied to humpback vocalizations. When the extracted “target” tone-pairs of the two most energetic “vocal fold” harmonic frequencies were plotted in x-y coordinates and subjected to cluster analysis, the plot exhibited properties of a Shannon-Hartley-compliant “modem symbol constellation” diagram of 14 distinct tone-pair and single-tone sub-regions. The humpback voiced symbol constellation is structurally comparable to the Peterson-Barney, F1 vs F2, formant mapping of English language vowels. The humpback’s core set of 14 fixed-pitch sub-units is augmented by numerous intra- and inter-sub-regional pitch-varying transitions similar to the generation of Asian language tonal vowels. The information entropy and plot of the humpback’s complete 65-sub-unit symbol-set’s cumulative probability versus ranked frequency distribution function are nearly identical to the entropy and Zipf power-law profile of the English language phoneme set. The precise specification of units referenced to their constituent sub-unit symbol-codes has enabled the compilation of an “alphabetic-lexicon” of voiced humpback song units.
Quantifying macro-rhythm in English and Spanish: A comparison of tonal rhythm strength in two speech styles
Authors: Prechtel Christine
Published: November 2019

This study quantified macro-rhythm in English and Spanish in two speech styles. Macro-rhythm is defined as phrase-medial tonal rhythm (June 2014), and its strength is determined by the frequency of f0 alternations between peaks and valleys within a phrase, the uniformity of the rise-fall shape, and the regularity of distance intervals between high and low targets. The degree of strength can be predicted based on the number of phrase-level tones in a language’s tonal inventory, the most common type of phrase-medial tone, and the frequency of f0 rise per Prosodic Word. Based on these criteria, Spanish is predicted to have stronger macro-rhythm than English: the most common pitch accent is L+<H* in Spanish and H* in English, and Spanish tends to accent content words with greater regularity than English. Two experiments quantified and compared slope shape variability, the number of f0 alternations per Prosodic Word, and the regularity of distance intervals between tonal targets in read speech (Experiment 1) and newscaster speech (Experiment 2). The results of the variability, frequency, and distance measures support the prediction that Spanish has greater macro-rhythm strength than English.
Vowel descriptions beyond the first and second formant frequencies
Authors: Zhang Hong , Liu Shuang
Published: November 2019

Conclusions about variation and change of vowels based on the first (F1) and second (F2) formant frequencies may be challenged by at least two major drawbacks intrinsic to this simple 2-D representation. First, F1 and F2 measured at one or two representative points can capture only limited amount of vowel dynamics. The numeric values of F1 and F2 are also prone to errors due to aperiodicity in speech signal. In this study, we explore alternative ways for more informative and robust vowel representation. We examine both LPC and MFCC based features using two encoding methods. Frequency domain features are first estimated at each time step for the entire vowel duration. The resulting t feature vectors are then passed sequentially through either a set of t independent autoencoders or a recurrent neural network (RNN), both with k hidden units. In case of the autoencoders, each k-by-t vowel matrix is further projected to a lower dimensional space using PCA. In case of the RNN, the k-dimensional hidden vectors are used for cross-condition comparisons. Compared to the simple F1-F2 measurement, our methods are able to capture further nuanced within-category variations across dialect varieties in TIMIT. Results from LPC and MFCC based representations are also compared.
Frequency hopping by big brown bats is coupled with sonar sound grouping in response to pulse-echo ambiguity
Authors: Tuninetti Amaro , Simmons Andrea , Simmons James A.
Published: November 2019

When flying in clutter, big brown bats actively modify their biosonar broadcasts by successively alternating the lowest frequencies in the first harmonic of their FM broadcasts in a phenomenon called “frequency hopping;” and by grouping broadcasts into “sonar sound groups,” with short time intervals within a group and longer intervals between groups. These modifications can minimize pulse-echo ambiguity. We analyzed the relationship between frequency hopping and emission of sonar sound groups while bats flew through chain corridors with different clutter densities and through circular hoop tunnels. In more difficult flight tasks, bats emit more sounds in groups of triplets and quadruplets, often with frequency shifts of 1–6 kHz between each sound in the group, creating not only sound but frequency triplets and quadruplets. Echoes of successive sounds remain highly correlated and not separable by conventional receivers according to which broadcast causes them. The consistent pairing of frequency shifting with sonar sound grouping may reflect an auditory streaming mechanism whereby coupled shifts in both frequency and timing can link streams of received echoes to their corresponding emitted calls to disambiguate overlapping echo streams for closely spaced broadcasts. [Work supported by ONR.]
Ambisonics and wideband matched-filter approach for bat biosonar study
Authors: Lee Hyeon , Roan Michael J. , Ming Chen , Simmons James A.
Published: November 2019

We previously introduced a method of bat echolocation signal measurement and analysis for target angle estimation using a custom-built high-frequency (20–80 kHz) soundfield microphone and spatial audio techniques. The method provided precise angle estimates of multiple targets. This presentation will provide details and results of expanding the processing of these spatial audio signals using a wideband matched-filter. The custom-built ultrasonic soundfield microphone recorded a static bat’s echolocation signals directly in front of the bat’s head. The measured audio signal in B-format was processed and reproduced for all directions with an increment of 1 deg in azimuth and elevation using Ambisonic audio processing techniques. A matched-filter algorithm was developed and produced wideband ambiguity functions (WAF) of transmitted signals from a high-frequency transmitter or bats. The wideband matched-filter outputs filter responses of the reproduced signals in time domain for all directions. The target estimate accompanying the strongest filter response provides the range, normal velocity, angle, and elevation of the target. This presentation shows the hardware and software development including experimental validation to establish this new method for bat biosonar study.
Context knowledge-aided sparse bayesian learning method for bathymetric sonar
Authors: Shen Jiajun , Zhou Tian
Published: November 2019

Sparse bayesian learning has recently become successful in many compressed sensing problems. However, their performance critically relies on the appropriate tuning of numerous hyperparameters, because they directly control the statistical distribution and have a significant impact on the performance of the model is being trained. Compared with manual tuning, grid search, and random search, bayesian optimization is a better strategy for intelligently optimizing the hyperparameters. Regardless, traditional bayesian optimization treats each problem as an independent and black box. In this study, the hyperparameters tuning process is improved further due to the contextual knowledge can be incorporate into problem domain such that is easier to optimize. For the bathymetric sonar, the adjacent observation data have certain internal correlation. That is, it is generally believed that the underwater acoustic environment changes slowly, which can be draw inspiration from the ability and incorporate it into the bayesian optimization framework as additional prior information. This extension could yield efficient optimization, without requiring too many iteration steps to converge. The tank experiments are conduct, and the results demonstrate that the proposed method has lower computational burden.
Universal adaptive beamforming for moving interferers
Authors: Buck John R.
Published: November 2019

Loud moving interferers challenge adaptive beamformers (ABFs) attempting attenuate these interferers while detecting quiet sources. Specifically, moving interferers create a dynamic tension in choosing the best window length for estimating the sample covariance matrix (SCM) from received snapshots. If the window is too short, the ABF does not accurately steer its nulls toward the interferers, diminishing its effectiveness in attenuating the interferers. If the window is too long, the interferers transit multiple beamwidths, splitting their power across multiple eigenvectors, also diminishing the ABF's effectiveness (Baggeroer and Cox, 1999). Previously proposed approaches to address this challenge include Multirate Adaptive Beamforming (Cox, 2000) and Null-Broadening (Song et al., 2003). This talk proposes a universal adaptive beamformer (UABF) exploiting online learning techniques to adapt the length of the SCM averaging window. The UABF's array weights blend the array weights of several competing ABFs. The recent performance of each fixed-window ABF determines its contribution to the UABF array weights. The UABF's performance provably converges to rival the best fixed-length window ABF for every finite power input sequence, without assuming a specific probability distribution. In nonstationary environments where interferers change speed, the UABF may outperform all of the competing fixed-window ABFs. [Supported by ONR 321US.]
Investigation on measurement method for hydrodynamic noise in centrifugal pumps using reverberation tanks
Authors: Zhang Yiming , Li Qi , Tang Rui , shang dajing
Published: November 2019

Pumps are considered as a primary source of acoustic energy in industrial piping. At present, the measurement of the hydrodynamic noise of the pump is done by installing the sensors in the pipeline attached to the pump. This technique is based on the fact that plane waves are formed inside the pipeline and the pulsating pressure travels in the form of sound waves. However, in actual there are no plane waves in an elastic pipe due to the phenomenon of low frequency cut-off. In fact, this pulsating pressure inside the pipeline will result in further vibration of the pipeline. This paper proposes a method for measuring the hydrodynamic noise of the pump by using the reverberation tank. The vibration of a pipeline attached to a centrifugal pump can be avoided by replacing the elastic pipeline with a flexible hose. The water outlet of the hose is then placed in the reverberation tank fitted with hydrophone arrays. The hydrodynamic noise of the water pump is measured using low frequency expansion technique of the reverberation method. Experiments show that this method can be used effectively to differentiate between the hydrodynamic noise and other associated noises of a pump.
Research on creation and evolution of artificial diffuse sound field
Authors: Sun Jundong , Li Qi , Shang Dajing , Tang Rui
Published: November 2019

In a closed space, it is difficult to create a diffuse sound field by using a single source in the reverberation tank. In this paper, a methodology for construction of diffuse sound field using multiple source excitation is proposed. Proposed methodology is based on space processing technique for a single source sound field and also involves reciprocity of sound field transfer function. Firstly, the analytical method of acoustic excitation reverberation field is studied and subsequently the sound pressure equation for excitation reverberation sound field in case of multi sound is obtained. In addition, the numerical method to observe the influence of sound source distribution upon uniformity of sound field is studied by the acoustic analysis software ACTRAN. Finally, diffuse sound field in a glass tank of 1.5 m × 0.9 m × 0.6 m dimensions, is constructed by using 30 emission transducers of the same type. In the end, the diffuse sound field is scanned using scanning array and is then evaluated by measuring the standard difference of the space distribution of sound field. The results show that the standard deviation of the sound pressure distribution in the diffused sound field in comparison with 30 transmitting transducers simultaneously within the reverberation control area is not greater than 3.9 dB under the narrow band condition. Hence, it clearly illustrates that construction of diffused sound field by this method is definitely a viable option.
A robust and accurate interpolant for tabulated sound speed profiles
Authors: Peterson John C. , Porter Michael B.
Published: November 2019

An new interpolation algorithm that is a hybrid of the cubic spline and monotone piecewise cubic Hermite polynomial interpolants is introduced. The resulting algorithm has several properties that make it an attractive choice for the interpolation of sound speed profiles in acoustic propagation models. It has been shown that when interpolants do not have a continuous first and/or second derivative, the resulting acoustic model predictions can exhibit a variety of artifacts such as false or omitted caustics. The first derivative of our interpolant is continuous everywhere. Our interpolant also automatically respects the monotonicity of the underlying data in the sense that it will be monotone (increasing or decreasing) on any interval where the associated data is also monotone. This property avoids the oscillations between data points that can be a source of frustration when using some interpolants such as cubic splines. Finally, the interpolant will also possess a continuous second derivative everywhere, except at those data points where monotonicity considerations take priority. In some cases, the interpolant will coincide exactly with the cubic spline.
Effects of sound source and its direction on subjects' impression of soundscape in workplace
Authors: Akita Takeshi , Sano Naoko , Yu Hanui , Matsuo Ayako
Published: November 2019

In the ordinary workplace, there exist many kinds of sound. However, workers usually do not intend to listen to them. To keep soundscape in workplace good, it is necessary to know how they feel such sound that they take no notice of. In the present research, one experiment was carried out for the purpose. Twenty subjects participated in the experiment and experienced three kinds of sound source, that is conversation, music, and signal sound, that had sound pressure level of 50 or 60 dBA and presented from four different directions. Results show that sound of 50 dBA has generally better impression than 60 dBA and signal sound of each SPL is recognized as bothering and not calm. On the other hand, 50 dBA of conversation and music that are presented from behind worried subjects little. Besides, 60 dBA of music presented from right and left side of subjects is felt as obstructive as the 50 dBA of signal sound. As a result, it is suggested that soundscape in workplace should be planned with attention to the person's behavior, meaning and volume of sound source, and its spatial layout.
Contextual design of soundscapes: How to
Authors: Elizondo - Garza Fernando J.
Published: November 2019

This paper will be discussed some aspects to be considered in the contextual design of soundscapes in contrast with the classical acoustic design. Aspects as: perception in base of cultural background, perception as a multisensorial integration, attention versus boredom, expectation versus deception, soundscape versus landscape, real versus virtual, deterministic design versus design by compromise, isolation respect other spaces, complying of regulations, etc. Also, the need for an open mind approach in education of soundscape design will be discussed.
Variations in restaurant soundscapes
Authors: Paine Jared A. , Palakapilly Joshua , Wang Lily M.
Published: November 2019

The sound levels and concurrent occupancy at a number of restaurants located in Omaha, Nebraska, have been logged during occupied hours. Information on each restaurant’s room shape, materials, seating layout, and ambient noise sources have also been gathered. The time-logged data are analyzed to understand the acoustic environment better, particularly by looking at statistical levels (for example, L10 and L90) and the percent of time that sound levels exceed certain values. Comparisons of the logged sound level data against Rindel’s predictive equation for restaurant noise levels which includes occupancy and group size (2019) show good correlation. How different sources of ambient noise and their levels in the restaurant may impact the resulting soundscape is additionally highlighted.
Sub-unit-based classifier boosts unit diversity and information entropy estimates of humpback whale songs
Authors: Pines Howard S.
Published: November 2019

The ongoing focus of humpback whale song research is the detection and measurement of periodic patterns of discrete units. The expeditious analysis of high volumes of song data has motivated a holistic approach to unit feature-extraction and classification. These whole-unit-feature-approximation and statistical-classification methods generate diminished unit diversity assessments, resulting in reduced Shannon first-order unit-entropy estimates of 4 to 4.5 bits and second-order estimates in the one-bit range. When the feature-extraction and classification of pitch-varying units is referenced to a new sub-unit-based classification model consisting of an “alphabet” of 65 sub-units, the result is a major expansion in the number of unique units compared to previous studies. This precise specification of sub-unit structure has enabled a marked increase in first-order unit-entropy estimates in the 7-bit range, compared to Shannon’s 11.8 bit first-order entropy estimate of the dictionary of English language words. The units exhibiting the greatest sub-unit diversity are concentrated in three- to five-minute sub-sections of songs. When Shannon second-order entropy analysis is restricted to these time-limited sub-sections, the localized entropy estimates increase significantly. Application of the sub-unit-based classifier to the study of fixed-pitch units and additional song data should lead to a further expansion of the “dictionary” of unique units.
Learning relationships between bat habitats and biosonar behavior
Authors: Goldsworthy Michael , Bradley Aidan , Harrison Xavier , Newsome Walter , Dzikunu Georgina , Mueller Rolf
Published: November 2019

Echolocating bats are known to adapt their biosonar behaviors according to their surroundings. Vespertilionid bats, for example, adapt their pulse duration and bandwidth as a function of their proximity to foliage. However, these relationships have been based on mostly qualitative observations so far. More powerful methods that are capable of handling large quantitative datasets could produce further and more detailed insights. In the current work, aligned multimodal datasets have been gathered to address this challenge: habitat geometry (laser scans), acoustic scene (biomimetic active sonar system), bat flight trajectories (infrared camera array), and biosonar behavior (microphone array). Machine learning approaches can then be used to analyze relationships between the components of these multimodal datasets. To do this, representations of the bat’s surroundings that are suitable for biosonar problems have been developed, for example a probability density function of the angles and distances from the laser scanned points of a habitat model relative to the bat’s position and orientation. These representations will serve as inputs to machine learning algorithms that can predict the quantitative features of the bat’s biosonar behavior. This research could lead to a better understanding of bat pulse adaptation and also aid in developing biomimetic sonar navigation systems.
Listening to biodiversity under extreme weather
Authors: Lee Chia-yun , Lin Tzu-Hao , Huang Chun-Chia , Tuanmu Mao-Ning
Published: November 2019

Climate change is one of the major threats to biodiversity. Besides increasing air temperatures, the magnitude and frequency of extreme weather events have been increasing, and are predicted to increase further in the future. To understand impacts of extreme weather on biodiversity, long-term and frequent data collections on the environment and biodiversity responses are needed. In this study, we used the environmental ultrasounds collected at eight sites across Taiwan to evaluate the usefulness of passive acoustic monitoring for accessing ecological impacts of heavy rains, which are one of the deadliest weather events. We separated acoustic signals of bats, insects and rains using a machine-learning tool we developed. We then investigated temporal changes in the intensity and frequency compositions of bat signals over a heavy rain event. We found significant impacts of the event on the acoustic signals produced by bats. Bat signals took a longer time to recover from the event at the places experiencing heavier and more temporally concentrated rains than at the places with lighter and more spread out rains. This study shows the potential of passive acoustic monitoring for improving our understanding of the responses of biodiversity to extreme weather events under climate change.
Segmentation of overlapping sources in mixtures of bat echolocation calls
Authors: Izadi Mohammad Rasool , Stevenson Robert L. , Kloepper Laura
Published: November 2019

Acoustic recordings of animals in their natural habitat often pose complex analysis challenges. For animals in groups, such as bats echolocating in a large swarm, one challenge is analyzing individual sounds that overlap in frequency and/or time with others. We developed a machine learning method, using deep neural networks, that isolates individual sources in the time-frequency domain using only one network that performs in two separate steps; one of detection and the second one of segmentation. In the detection step, the number of sources along with their bounding boxes are estimated. In the segmentation step, masks of individual sources are extracted. This method allows for parsing of individual bat echolocation calls from recordings containing hundreds of overlapping calls so that individual call characteristics can be studied. We will present the methodology and demonstrate the performance from recordings of variable numbers of bats and SNRs.
Using ambient noise to self-localize and time-synchronize independent sensors for localizing gray whales acoustic activity in Laguna San Ignacio, Mexico
Authors: Tenorio-Hallé Ludovic , Thode Aaron , Swartz Steven , Urbán-Ramírez Jorge
Published: November 2019

Beamforming requires array elements whose data is time-synchronized and spacing is known. This typically involves either fixed sensors running off a central acquisition system, or additional equipment, such as pingers or other calibration sources. It has been previously demonstrated that for a diffuse noise field, the time-averaged ambient noise cross-correlation function between array elements can be used to infer both their spacing and relative clock-offset (Sabra et al., 2005). This allows beamforming across fully independent instruments (i.e., from different acquisition systems), whose relative positions are not precisely known, or may vary. Here, we apply this technique to pairs of bottom-mounted recorders deployed between 5 and 15 meters depth with approximately 15 m spacing. The data was collected in Laguna San Ignacio (Mexico) with the goal of localizing gray whale acoustic activity. Results show that even if the noise field is not strictly diffuse nor azimuthally symmetric, the structure of the time-averaged ambient noise cross-correlation function still allows correcting relative clock-offset and drift throughout the deployment. K. G. Sabra, P. Roux, A. M. Thode, G. L. D'Spain, W. S. Hodgkiss, and W. A. Kuperman, “Using ocean ambient noise for array self-localization and self-synchronization,” IEEE J. Oceanic Eng. 30,338–347 (2005).