bluuebunny/crossref_metadata_2025 huggingface dataset:
Numerical study for the function of moving pad on tone hole acoustics
Authors: Kobayashi Taizo , Matsuda Hidetaka , Tatami Toshiya , Takahashi Kin'ya
Published: November 2016

In this poster, we will discuss numerical methods for calculating a tone hole model with a moving pad of woodwind instrument by using compressible Large Eddy Simulation (LES), i.e., how to treat moving boundary problems. There are three stages of numerical methods to treat a moving pad on a tone hole. The 1st stage is the simplest method in which a dynamical mesh technique is not employed. It is so called “Stop Motion.” The numerical conditions of the distance between the pad and the top of the tone hole are set to adequate points to discuss the effect of the distance, e.g., we set it to 0 mm (i.e., the tone hole is closed by a pad), 0.5 mm, 1 mm, 2 mm, 3 mm, and 5 mm. In the second stage, while a dynamical mesh technique is employed, on the other hand, it does not treat the condition with topological change of the boundary, i.e., in this situation, the distance changes from/into 0 mm. The third stage is a full dynamical mesh technique which provides a method for treating moving boundary problems with topological changes. As the results of calculation, we will also report how the pitch is changed with fingering, and how the fluid field and acoustical field behave around the tone hole when it is changed, comparing with experimental study on the function of tone holes reported by Keefe [1]. Keefe, D.H. ``Experiments on the single woodwind tone hole'', J. Acoust. Soc. Am., 72 688-699 (1982).
Electronic cymbals using spectral modeling synthesis with variable timbre by hitting strength and beater hardness
Authors: Wakatsuki Naoto , Mizutani Koichi
Published: November 2016

Sound source for cymbals in electronic drums based on spectral model synthesis were proposed instead of wavetable based sound sources. In the sound source, the timbre of the synthesized sound can be changed by hitting strength. In the real acoustical cymbals, the timbre change is caused by the coupling among the eigenmodes due to geometrical nonlinearities. In the sound source, it was reproduced using a simple model. As well as the hitting strength, the hardness of beaters hitting a sensor pad are also reflected to the timbre. It was achieved by measuring pulse width in the output analog signal from a commercial sensor pad. The proposed sound source was implemented as a Steinberg's Virtual Studio Technology (VST) plugin. In the talk, the implemented sound source will be simply demonstrated.
Transcribing piano music in the time domain
Authors: Cogliati Andrea , Duan Zhiyao , Wohlberg Brendt
Published: November 2016

Automatic music transcription is the process of automatically inferring a high-level symbolic representation, such as music notation or piano-roll, from a music performance. It has many applications in music education, content-based music search, musicological analysis of non-notated music, and music enjoyment. Existing approaches often perform in the frequency domain, where the fundamental time-frequency resolution tradeoff prevents them from obtaining satisfactory transcription accuracies. In this project, we develop a novel approach in the time domain for piano transcription using convolutional sparse coding. It models the music waveform as a summation of piano note waveforms (dictionary elements) convolved with their temporal activations (onsets). The piano note waveforms are pre-recorded in a context-dependent way, i.e., for the specific piano to be transcribed in the specific environment. During transcription, the note waveforms are fixed, and their temporal activations are estimated and post-processed to obtain the pitch and onset transcription. This approach models temporal evolution of piano notes, and estimates pitches and onsets simultaneously in the same framework. Experiments show that it significantly outperforms a state-of-the-art frequency-domain music transcription method trained in the same context-dependent setting, in both transcription accuracy and time precision, in various scenarios including synthetic, anechoic, noisy, and reverberant environments.
Multiple-timbre note tracking using linear dynamical systems
Authors: Benetos Emmanouil
Published: November 2016

This work addresses the problem of multi-pitch detection and note tracking in multiple-instrument polyphonic music recordings. A system is developed that extends probabilistic latent component analysis (PLCA) and supports the use of a five-dimensional dictionary of spectral templates per instrument, pitch, deviation from ideal tuning, and sound state (e.g., attack, sustain, decay). A method based on linear dynamical systems (LDS) is introduced for note tracking, which assumes that the output of the PLCA model is the (noisy) observation in an LDS, with the latent states corresponding to the ideal multi-pitch activation output. The LDS-based process supports the tracking of multiple concurrent pitches and can also be integrated within the PLCA-based model, thus guiding the convergence of the multi-pitch detection process. Experiments performed on several datasets of multiple-instrument polyphonic music demonstrate that the LDS-based method leads to significant improvements in multi-pitch detection as compared to using the frame-based PLCA model alone.
Acoustics upstaged: Challenging conundrums at the Starlight Bowl
Authors: Conant David A.
Published: November 2016

Since its debut in 1935, the 3600-seat Starlight Bowl in San Diego’s renowned Balboa Park has been increasingly impacted by environmental noise, principally from aircraft landing at nearby Lindbergh Field. Traditionally, performers would temporarily “freeze” on cue from the director as a jet approached, then resume as though nothing had happened. Although amusing initially, audiences soon were just annoyed. MCH was retained to identify and rank prospective mitigation measures as the principal tenant was losing audience. In addition to collecting the usual ambient and fly-by noise level data, MCH examined the efficacy of partial and full enclosure options, and associated implications on audio system planning and operation. Ultimately, the myriad of acoustical/audio noise-mitigating options developed could not be reconciled with the historic venue’s name, as the visual impact alone, of large jets flying within 600 ft would forever be up-staging any on-stage performance. We will report on our process, observations, and recommendations.
“Shhh.” and other methods for controlling crowd noise
Authors: Kanter Shane J. , Laney Jonathan
Published: November 2016

As designers of venues for performance, we spend a considerable amount of time and effort on maintaining a low background noise level while the acoustically critical spaces are in use. The typical noise sources, such as mechanical systems and light fixtures, are predictable and controllable. Just as background noise is inferential, audience noise within the common classical music venue is generally understood and under control. Effects of outdoor background noise are explored—such as city noise, road noise, and cicadas at two Chicago outdoor venues: Jay Pritzker Pavilion and Ravinia Festival. Second, because both of these venues have a fixed section of seating near the stage and a large section of general admission law seating, the lawn seating typically fosters a relaxed atmosphere where patrons feel free to converse with one another during performances. Ravinia and the Pritzker Pavilion differ when it comes to how they deal with chatting patrons during performances. Special attention will be paid to the audience noise in both the fixed seating and lawn seating at each venue and through the use of patron interviews, and exploration the effect of audience noise on the performance experience.
Overcoming challenges to future outdoor entertainment performances due to disruptive commercial vendor noise intensity and the resulting community outrage
Authors: Hale Marlund E.
Published: November 2016

Outdoor entertainment and concerts are commonplace historically. Many of these public attractions also include commercial, concessions, and other vendor booths. Typically, such events are regulated by local government. During the inaugural performance of a successful music festival, whose sponsors had contractually agreed to the community permitting requirements, a commercial vendor proceeded to generate such loud and intense sounds that the entire festival and closest residential community were bombarded with object vibrating sounds. The public outcry from the local residential community was immediate and so was the reaction of the festival organizers. The commercial vendor was ordered to vacate the venue. However, the local community was so outraged that the festival organizers were put on notice that the festival had violated its contract and would not be permitted again. After many hours of community outreach and public hearings, the festival organizers were given a highly conditional permit to hold the festival again the following year. Strict A-weighted and C-weighted noise limits and heavy violation fines were imposed, as well as a requirement for continuous noise monitoring at multiple sites at the venue and in the residential community. This paper reports the concert performance results of the past few years.
Analysis of nozzle geometry effect on supersonic jet noise using Schlieren
Authors: Ozawa Yuta , Oyama Akira , Anyoji Masayuki , Oyama Akira , Mamori Hiroya , Fukushima Naoya , Yamamoto Makoto , Fujii Kozo
Published: November 2016

Strong acoustic waves emitted from rocket plume might damage to rocket payloads because rocket payloads consist of fragile structure. Therefore, it is important to predict acoustic directivity and reduce its intensity level. In this study, we conduct experiments of supersonic jet flows and investigate an influence of the nozzle geometry on acoustic waves by means of Schlieren method and microphone measurement. Three different nozzles are examined: a conical nozzle, a convergent-divergent nozzle (referred as C-D nozzle), and a tab-C-D nozzle. Tabs are equipped in the nozzle inside and turbulence is generated in the tab-C-D nozzle case. The Schlieren visualization shows that the strong shock trains are observed in the potential core of the jet for the conical nozzle case, while the shock waves are relatively weak since the nozzles are in the nearly ideal expanded condition in the C-D nozzle and tab-C-D cases. The distribution of near field OASPL (over all sound pressure level) obtained by microphone measurement shows strong directivity in the downstream direction for all the cases. This directivity seems to be the Mach wave radiation. Moreover, conical nozzle cases have strong acoustic intensity level caused by shock associated noise.
Comparison of spatial correlation parameters between full and model scale launch vehicles
Authors: Giacomoni Clothilde , Kenny R. Jeremy
Published: November 2016

The current vibro-acoustic analysis tools require specific spatial correlation parameters as input to define the liftoff acoustic environment experienced by the launch vehicle. Until recently, these parameters have not been very well defined. A comprehensive set of spatial correlation data were obtained during a scale model acoustic test conducted in 2014. From these spatial correlation data, several parameters were calculated: the decay coefficient, the diffuse to propagating ratio, and the angle of incidence. Spatial correlation data were also collected on the EFT-1 flight of the Delta IV vehicle which launched on December 5, 2014. A comparison of the spatial correlation parameters from full scale and model scale data will be presented.
Spatial variation in similarity spectra decompositions of a Mach 1.8 laboratory-scale jet
Authors: Vaughn Aaron B. , Neilsen Tracianne B. , Gee Kent L. , Okamoto Koji , Akamine Masahito
Published: November 2016

The primary source of noise from supersonic jets is turbulent mixing noise. Tam et al. [AIAA Paper 96-1716 (1996)] proposed similarity spectra for a two-source model of turbulent mixing noise corresponding to noise from omnidirectional fine-scale turbulence structures and directional large-scale turbulent structures. These empirical similarity spectra have been compared with reasonable success to the spectra of both military and laboratory-scale jets. Most application have looked at the variation in angle: fine-scale similarity spectra matches sideline of the jet nozzle, large-scale in the maximum radiation lobe, and a combination needed in between. A similarity spectra decomposition of from an ideally expanded, Mach 1.8 laboratory-scale jet allows for a spatial comparison between the near and far-field spectra. The sound from the convergent-divergent nozzle was collected at the Hypersonic and High-Enthalpy Wind Tunnel at Kashiwa Campus of the University of Tokyo at a variety of near, mid, and far field locations. Comparison of similarity spectra decompositions over distance yield insights into the sound propagation from this supersonic jet. In addition, results from a preliminary wavenumber eduction technique to build a wavepacket-based equivalent source model of the large-scale turbulent mixing noise. [Work supported by Office of Naval Research grant.]
Relating binaural psychophysics to human physiology with functional magnetic resonance imaging: Opportunities and challenges
Authors: Stecker G. C. , Higgins Nathan C. , Da Costa Sandra , McLaughlin Susan A.
Published: November 2016

Functional magnetic resonance imaging (fMRI) has revolutionized the study of human perception in all sensory modalities. While early studies focused on identifying the brain regions involved with various processes (“where”), recent approaches aim to quantify relationships between brain activity and the physical or perceptual features of sensory stimuli (“how”). These modern approaches thus mirror the psychophysical goal of relating sensation to its underlying physical parameters, providing a clear means to connect human psychophysics and brain physiology. This presentation will describe a series of studies investigating the sensitivity of human auditory cortex (AC) to binaural features of sounds and their perceptual correlates. Specifically, three questions will be addressed: (1) Does AC activity respond to the physical features or to the perception of binaural sound? (2) Do competing sounds sharpen or broaden the binaural selectivity of AC responses? (3) Does binaural sensitivity in human AC change across spatial, non-spatial, and non-auditory tasks? Questions (1) and (2) assess cortical correlates of psychophysical performance, elucidating the AC processes involved in spatial hearing. Question (3), however, goes a step further by addressing the potential effects of the psychophysical task itself. [Work supported by NIH R01 DC011548.
Effects of Auditory Attention on Otoacoustic Emissions
Authors: Hernandez Perez Heivet , McMahon Catherine , Dhar Sumitrajit , Boothalingam Sriram , Poeppel David , Monaghan Jessica J.
Published: November 2016

In humans, auditory efferent control of the cochlea has been implicated in sharpening of frequency tuning, improved detection of signals in noise, and the ability to switch the focus of auditory attention. However, it remains unknown whether the modulation of efferent activity during an auditory attention task depends on the degree of task difficulty. This study aimed to compare the suppression of otoacoustic emissions (OAEs), an objective measure of auditory efferent activation, during a lexical decision task with varying degrees of difficulty, compared with passive auditory listening. Ten normal-hearing 18-35 year-olds were assessed using monosyllabic words and non-words presented in a natural or noise-vocoded (less intelligible) condition. The participants were instructed to press a button every time they heard a non-word. Simultaneously, click evoked-OAEs were obtained from the contralateral ear to the speech stimuli with a probe in the external ear canal. Preliminary results showed that OAEs amplitudes were suppressed during the lexical decision tasks relative to passive listening. In addition, an effect of task difficulty was found, whereby the less intelligible condition showed stronger suppression. These data suggest that the auditory efferent system is recruited through auditory attention, and that this may play a role in speech perception.
Frequency characteristics of neuromagnetic auditory steady-state responses at the supra-threshold level
Authors: Otsuka Asuka , Yumoto Masato , Kuriki Shinya , Nakagawa Seiji
Published: November 2016

Auditory steady-state response (ASSR) is a neuronal electrical component recorded as a continuous sinusoidal signal phase-locked to a modulation frequency of sounds. Since ASSR amplitude shows high correlation with perceptual characteristics at hearing threshold level, the ASSR has been examined as a candidate for an objective index of the human hearing ability per frequency. However, the relationship between ASSR and perception at the supra-threshold level has not been clarified. In this study, characteristics of ASSR magnitude relative to loudness at the supra-threshold level were investigated. Neuromagnetic 40-Hz ASSR was recorded in response to sinusoidally amplitude-modulated sweep tones with carrier frequency covering the frequency range of 0.1-12.5 kHz. Sound intensity was equalized at 50-, 60-, and 70-dB SPL with an accuracy of ±0.5-dB SPL at the phasic peak of the modulation frequency. Corresponding loudness characteristics were modeled by substituting the detected individual hearing thresholds into a standard formula (ISO226:2003(E)). The strength of the ASSR component was maximum at 0.5 kHz, and it decreased linearly on logarithmic scale toward lower and higher frequencies, whereas loudness model was plateaued between 0.5 and 4 kHz. The results indicated that frequency characteristics of the ASSR were not equivalent to those of SPL and loudness model.
Effects of noise on a person's psychological state under task-loaded condition from the viewpoint of brain function analysis
Authors: Akita Takeshi , Sano Naoko , Matsuo Ayako
Published: November 2016

Measurement and analysis of brain function by near infrared spectroscopy (NIRS) obtained from the frontal area of scalp can reveal some aspects of psychological state of human. In the present research, investigation of effects of noise on task-loaded person is attempted using analysis of NIRS. Two experiments are conducted. In the first experiment, brain function of frontal regions is measured under resting and task-loaded condition with back ground noise to confirm that task-loaded person uses their frontal area of brain. The results show that subjects use their frontal region of brain sufficiently while performing their tasks. In the second experiment, subjects carry out their calculating tasks under several noise conditions. Analyses of the data show that activity of frontal brain area diminishes. It is suggested that noise interrupt subjects' thinking activity for carrying out their task. These results suggest that measurement and analysis of NIRS is available for clarifying the negative effects of noise on task-loaded person. Using these methods can lead to create an index of distracting aspects of noise.
Acoustic topological insulators
Authors: Fleury Romain
Published: November 2016

The recent discovery of topological states in condensed matter physics has spawned a quest for similar effects in classical physics. Topological insulators are bulk insulators supporting bandgap crossing, chiral edge states that are immune to backscattering caused by defects or disorder. Here, we present the acoustic equivalent of topological insulators, obtaining the analog of the Quantum Hall Effect (QHE) and the Quantum Spin-Hall Effect (QSHE) in acoustic metamaterials with, respectively, broken or preserved time-reversal symmetry. We demonstrate how our recent results in the field of non-reciprocal acoustics [cf. Science 343, 516 (2013); Ac. Today. 11, 14 (2015)] can be used to induce an acoustic version of the QHE in active metamaterials with broken time-reversal symmetry. We also put forward resonant and non-resonant schemes to emulate the Kane-Mele Hamiltonian in completely passive acoustic metamaterials, obtaining the acoustic analog of the QSHE. Our work reveals that topological acoustic metamaterials enable fundamentally new ways of manipulating sound, including one-way, chiral propagation of acoustic waves over broad bandwidth and topological protection against geometry imperfections, impedance mismatches, crystal defects, and local disorder.
Loss-induced asymmetric transmission through gradient-index metasurfaces
Authors: Shen Chen , Li Yong , Jing Yun
Published: November 2016

Gradient-index metasurfaces have shown a great potential for wavefront modulation and transmission/reflection control. However, the effect of loss is rarely discussed in acoustic metasurfaces. Here, we study the effect of loss in gradient-index metasurfaces and suggest its application in asymmetric transmission. For a gradient-index metasurface composed of six types of unit cells, theoretical analysis reveals that, when loss is considered, the transmission coefficients for different diffraction modes will have different responses for acoustic waves with oblique incidence. The gradient-index metasurface thus can be utilized to achieve asymmetric transmission. Numerical simulations based on effective medium and real structure are performed to validate the theoretical findings using a single layer of gradient-index metasurface. The transmission contrast can be greater than 20 times in simulations within a certain range of incident angles. This design may be useful in ultrasonic and noise control applications.
A transparent broadband metallic lens for focusing underwater sound
Authors: Su Xiaoshi , Norris Andrew N.
Published: November 2016

Diffractive lenses usually suffer from spherical aberration. Even though it is possible to minimize aberration by design, the quality of focused sound is still limited owing to the side lobes caused by aperture diffraction. In recent years, hyperbolic secant index profiles have been widely used in gradient index (GRIN) metamaterials design to reduce aberration and suppress side lobes. Here, we introduce a 1D transformation to modify the secant profile for further aberration reduction, and present a GRIN lens comprised of hexagonal unit cells obeying the modified profile for focusing underwater sound in the near field. The transversely isotropic unit cells are designed with water-like acoustic properties using foam mechanics and optimized using a homogenization technique based on finite element method. The in-plane shear modulus of the unit cells are minimized to suppress the shear modes in the GRIN lens. Moreover, the impedances of the unit cells are tuned close to water, such that most of the energy carried by the normally incident plane wave propagate across the interface between the lens and water. This lens can focus underwater sound with high efficiency over a broadband (10 kHz—40 kHz).
Enhanced energy dissipation with nonlinear snapping acoustic metamaterial inclusions
Authors: Konarski Stephanie G. , Haberman Michael R. , Hamilton Mark F.
Published: November 2016

Through the design of subwavelength structures, nonlinear acoustic metamaterials can be engineered to exhibit regimes of positive and negative incremental stiffness. Negative incremental stiffness leads to large, rapid deformation, commonly called snapping, between local stable configurations. When perturbed by a small external pressure, snapping elements can be exploited to enhance energy absorption of acoustic waves. The present research focuses on a low volume fraction of snapping inclusions, dispersed within a nearly incompressible viscoelastic matrix to create a multiscale nonlinear material. The forced dynamic response of the heterogeneous material is modeled using a generalized Rayleigh-Plesset equation, which couples the dynamic behavior at the micro- and macroscale. We investigate the nonlinear dynamic behavior of the heterogeneous medium for small harmonic perturbations about several pre-strains of the snapping inclusions to demonstrate the influence of the microscale dynamic response on the macroscale response. Of primary interest are energy dissipation capabilities and characteristic acoustic impedance, which change with inclusion pre-strain. The behavior is also compared to heterogeneous media created from conventional inclusions, such as air voids and steel inclusions, to demonstrate the wide range of material behavior obtained using snapping inclusions. [This work was supported by the Office of Naval Research.]
Underwater acoustic absorption using a periodically voided soft elastic medium
Authors: Sharma Gyani Shankar , Skvortsov Alex , MacGillivray Ian , Kessissoglou Nicole
Published: November 2016

A coating comprising periodic cavities in a soft rubber medium can significantly reduce the reflection of acoustic waves from a steel plate. In this work, analytical and numerical models are developed to study acoustic reflection and transmission through a single layer of periodic cylindrical voids embedded in a soft elastic medium with steel backing. The analytical model is based on effective medium approximation theory which allows modeling of a composite material as a homogeneous material using effective material properties. The numerical model is based on the finite element method and developed using commercial finite element code COMSOL Multiphysics. Minimal reflection and transmission and maximal absorption of sound waves in a broad frequency range is attributed to the monopole type resonance of voids. The monopole resonance frequency of voids in an array is shown to be significantly higher compared to the monopole resonance of a single void in an infinite medium due to resonance coupling. An analytical framework to predict the resonance frequency of voids in an array is presented. The effect of strong and weak coupling of void resonances on reflection and transmission characteristics is investigated. Non-dimensionalized results obtained from the analytical and numerical models are compared.
Variation in the second formant: Its role in a creole's coarse-refined contrast
Authors: Lee Nala H.
Published: November 2016

Baba Malay encodes an interesting coarse-refined contrast. Speakers overtly recognize words ending with -al, -aɾ, and -as (e.g., bakaɾ “burn") as kasar “coarse,” while their counterparts ending with -ɛ (e.g., bakɛ “burn") are halus “refined.” The contrast is neither phonetic nor is it constrained by phonology or morphology. The coarse-refined system may be based on sound symbolism. The frontness of -ɛ is associated with a smaller articulatory space in the oral cavity, and hence refinedness, while the more backwards al, -aɾ, and -as forms, are associated with a larger articulatory space and hence coarseness. A perception experiment, mainly a within-subjects matched guise task, is conducted. Coarse and refined pairs are elicited from speakers. The relevant F2 of both variants are adjusted twice upwards and twice downwards in steps of 100 Hz. Listeners rate these forms on a scale of 1 to 5, 5 being most associated with values that are “refined.” Results show that the lower F2 is, the more likely listeners are to associate the coarse guise with “coarse” values, and the higher F2 is, the more likely listeners are to associate the refined guise with “refined” values. The kasar-haluscontrast is thus mitigated by sound symbolism operationalized by F2.
Substrate effects in the phonetic realization of Pidgin vowels
Authors: Grama James
Published: November 2016

Hawai'i is home to an English-lexified creole, known locally as Pidgin. While much research has focused on the structural development of Pidgin (e.g., Bickerton & Odo 1976), little work has investigated acoustic phonetic variation in the Pidgin vowel system. This talk presents results from an analysis of 854 tokens of /a/ produced by 32 speakers of Pidgin from two corpora—one from the 1970s and one from the 2000s—and investigates factors that contribute to differences in realizations of this vowel. Results indicate that /a/ in Pidgin exhibits a raised nucleus over real time for female speakers (p<0.001) but not male speakers. Additionally, analysis reveals that the youngest group of Pidgin speakers in the 2000s corpus realize /a/ in words of Hawaiian origin as higher in the vowel space than words of English origin (p<0.0001). Possible reasons for the emergence of this fine-grained phonetic difference in the youngest speaker group are discussed, including the role played by Hawaiian language education and the apparent valuation of Hawaiian. Taken together, results suggest that a combination of factors influences the realization of /a/ in Pidgin, including age, gender, and whether a word derives from the superstrate language ( = English) or a substrate language ( = Hawaiian).
Dictionary learning of acoustic sound speed profiles
Authors: Bianco Michael J. , Gerstoft Peter
Published: November 2016

Inversion for true ocean acoustic sound speed profiles (SSPs) is generally a non-linear and underdetermined problem. Traditional regularization methods model SSPs as a summation of leading-order empirical orthogonal functions (EOFs), with a minimum-energy constraint on coefficients. However, this often provides low resolution estimates of ocean SSPs. Sparse processing methods (e.g., compressive sensing) yield high resolution reconstruction of signals from few observations, provided few coefficients (of many) explain the observations. Using dictionary learning techniques, an overcomplete basis or dictionary of shape functions, which represent ocean SSPs using a minimum number of coefficients, is learned from a training set of SSPs. Learned dictionaries, which are not constrained to be orthogonal, optimally fit the distribution of possible ocean SSPs. Thus, each dictionary entry is informative to ocean SSP variability whereas EOFs become less informative for increasing order. It has been found that these learned dictionaries largely explain the variability of ocean SSPs using as few as one coefficient. Thus, in addition to improving the resolution of SSP estimates, learned dictionaries can improve the robustness of SSP inversion by significantly reducing the parameter search space.
Individual sound image generation by null-space based sound field control under few loudspeakers in real environment
Authors: Horii Shinpei , Tatekura Yosuke
Published: November 2016

To achieve an individual sound image generation with fewer loudspeakers based on NBSFC (Null-space Based Sound Field Control) filtering, we attempt to expand NBSFC under underdetermined condition, which means the number of loudspeakers is smaller than the number of control points. NBSFC filtering can suppress the sound pressure only on optional control points. In the conventional NBSFC design, the number of loudspeakers is required to be greater than that of control points. On the other hand, many control points are indispensable to achieve fine individual sound image generation. Thus, NBSFC filters are designed with the maximum control points limited by the conventional method from the transfer functions between the control points and the loudspeakers in various combinations. Also, the expanded NBSFC filters can be designed corresponding to the underdetermined condition by superimposing all conventional NBSFC filters. As the result of numerical simulation with a real environmental data, the sound pressure difference between inside and outside the control area was averaged by 7 dB. This reveals that the control area where sound pressure is suppressed can be realized by locating many control points closely.
Performance analysis of passive time reversal communication technique for multi-path interference with synthetic data
Authors: Kida Yukihiro , Deguchi Mitsuyasu , Watanabe Yoshitaka , Shimura Takuya , Ochi Hiroshi , Meguro Koji
Published: November 2016

The interference by time-varying and intense underwater acoustic multi-path is one of the most obstructive phenomena in underwater acoustic communication. Time-Reversal (TR) communication techniques utilize the reciprocity of acoustic channels to converge the spread signals to the original one, and are considered one of the most effective methods to overcome multi-path interference as shown in many past studies. It is recognized empirically that the large number of multi-path arrival could generally raise the demodulation result of TR. However, its relationship is rarely evaluated quantitatively. In this paper, we investigate the efficiency of the TR acoustic communication techniques for the multi-path interference cancelation by applying our passive TR method to the synthetic datasets of horizontal acoustic communication in shallow water environment. Mainly, we focused on the relation between the Signal to Interference Noise Ratio (SINR) and the output SNR of demodulation result. The result shows: 1) SINR and output SNR has a nearly linear relation up to a limitation, and 2) above the limitation, output SNR reaches its ceiling or slightly decreases.
Voice livness detection based on pop-noise detector with phoneme information for speaker verification
Authors: Mochizuki Shihono , Shiota Sayaka , Kiya Hitoshi
Published: November 2016

This paper proposes a pop-noise detector using phoneme information for a voice liveness detection (VLD) framework. In recent years, spoofing attacks (e.g., reply, speech synthesis, and voice conversion) have become a serious problem against speaker verification systems. Some techniques have been proposed to protect the speaker verification systems from these spoofing attacks. The VLD framework has been proposed as one of fundamental solutions. The VLD framework identifies that an input sample is uttered by an actual human or played by a loudspeaker. To realize the VLD framework, pop-noise detection methods have been proposed and these methods perform well as the VLD module. However, since pop-noise is a common distortion in speech that occurs when a speaker’s breath reaches a microphone, the phenomenon of pop-noise is able to be occurred by winds or attackers arbitrary. It is one problem of the pop-noise detection methods. In order to improve the robustness of the pop-noise detection methods, this paper proposes a pop-noise detector using phoneme information as an evidence of an actual human. From the experimental results, the proposed method increases the robustness of the VLD against spoofing attacks.
An improved filter employing measured characteristics of diaphragm displacement for compensating loudspeaker nonlinearity
Authors: Omura Manabu , Iwai Kenta , Kajikawa Yoshinobu
Published: November 2016

In this paper, we propose an improved filter employing measured characteristics of diaphragm displacement for compensating nonlinear distortions in loudspeaker systems. Loudspeaker systems generate nonlinear distortions that degrade the sound quality because of complicated structure of loudspeaker systems. Therefore, we set out to improve the sound quality by employing Mirror filter which compensates nonlinear distortions in electro-dynamic transducer. Mirror filter calculates linear instantaneous displacement of diaphragm from the linear differential equation of the electro-dynamic transducer and generates compensation signals based on the nonlinear differential equation of the electro-dynamic transducer and the calculated displacement of diaphragm. Hence, the compensation performance with Mirror filter is determined by the accuracy of the calculated linear and nonlinear motion based on the differential equations. However the calculated motion of the diaphragm does not agree with the actual motion perfectly because of the errors which come from parameters in the differential equations and modeled error. Therefore, an improved Mirror filter is proposed to exclude these errors. In detail, the improved Mirror filter employs the impulse response, which represents measured characteristics of diaphragm displacement, to calculate instantaneous displacement of diaphragm. In this paper, we demonstrate the validity of the proposed filter through experimental results.
Amplitude limiters based on phase optimization
Authors: Kakitani Akira , Saito Daisuke , Kosugi Yasuhiro , Minematsu Nobuaki
Published: November 2016

In order to reduce the peak value of source waveforms without quality degradation, a novel method is proposed. In this method, by exploiting the characteristics of human auditory system, which is often said to be insensitive to phase changes, peak value reduction is made possible. The proposed method is an improved version of our previous study, which manipulated the waveforms mainly based on phase control. In the conventional algorithm, however, amplitude characteristics have to be altered to some degree. The proposed method attempts to minimize the change of amplitude characteristics. Evaluation experiments show that the proposed method outperforms the conventional one in a Perceptual Evaluation of Audio Quality (PEAQ) test but the superiority is not found in a MUltiple Stimuli with Hidden Reference and Anchor (MUSHRA) test.
An integration method of multiple search results for spoken term detection
Authors: Shimizu Yoshino , Iwasaki Eitaro , Lee Shi-Wook , Tanaka Kazuyo , Kojima Kazunori , Itoh Yoshiaki
Published: November 2016

We propose a new integration method of multiple search results for improving search accuracy of Spoken Term Detection (STD). A usual STD system prepares two types of recognition results of spoken documents. If a query consists of in-vocabulary (IV) terms, the results using word-based recognizer are used, and if a query includes out-of-vocabulary (OOV) terms, the results using subword-based recognizer are used. The paper proposes an integration method of these two search results. Each utterance has a similarity score included in the search results. The scores of two results for an utterance has been integrated linearly using a constant weighting factor so far. Our preliminary experiments showed the search accuracy using the subword-based results was higher for some IV queries. In the same way, that using the word-based results was higher for some OOV queries. In the proposed method, the similarity scores of the two search results are compared for the same utterance and a higher weighing factor is given to the results that showed a higher similarity score. The proposed method is evaluated using open test sets, and experimental results demonstrated the search accuracy improved for all test sets.
Tracking whales in the Indian Ocean using the comprehensive test ban treaty hydrophone data
Authors: Lechner David A. , Guan Shane , Vignola Joseph F.
Published: November 2016

This paper will present preliminary results and status in development of an automated tracking algorithm for whale signals using the Comprehensive Test Ban Treaty (CTBT) sensor network. The data from the CTBT stations is being made available for academic work, and the stations are also detecting bioacoustics sounds as well as seismic activity. We present an approach used to process several channels of the acoustic data collected off Cape Leeuwin, Australia, and automatically search for biologic activity using cross-correlation and Time-Difference-of-Arrival methods. After developing a graph of locations for peak correlation amplitude, the locations are mapped and grouped using location and proximity, in order to develop motion tracks and compare coherence of the signals within a grouping and between groupings, thus identify individual whales based on the differences in location and coherence levels.
Basic study based on diffuse sound field model on relationships among architectural conditions, intelligibility, and speech privacy
Authors: Yamakawa Takashi
Published: November 2016

Rates of both speech transmission and speech privacy are usually estimated on the basis of intelligibility score. There are many physical indexes proposed to stand for intelligibility, like STI, AI, U50, SNRuni32, and so on. It will be reported that the results of calculations of intelligibility indexes with various room conditions based on classical diffuse sound field model, which shows relationships between degrees of absorption, sound blocking, background noise, distance between a target source and listeners, and intelligibility indexes. Some graphs of intelligibility versus each architectural condition will be provided. For example, architects could know how much each countermeasure, which includes adjusting absorption, set a barrier, adding masking noise, acquiring space, works for speech privacy to see the graphs. It is expected that the results will give us the basic guideline to develop architecturally oral communication and speech privacy.
Experimental studies on speech level and room acoustic condition focusing on speech privacy
Authors: Lee Hyojin , Sakamoto Shinichi
Published: November 2016

Recently, concerns about speech privacy to avoid oral information leakage at medical facilities are increasing continuously in Japan. In order to achieve the requirements of the speech privacy, standardization has also been discussed. On discussing the speech privacy, speech level is one of the most important factors. This study investigated the speech levels of Japanese talkers and the relationships between the speech levels and the room acoustic conditions (reverberation time, ratio of direct and reflected sound, and loudness of background noise). Subjective experiments were conducted under three conversation conditions; normal voice level with a partner sitting 1 m away, raised voice level with the partner sitting 3 m away, and voice level considering overhearing about the contents by a third party. Acoustic environments were reproduced in an anechoic room with a binaural sound field simulation system. As a result, the speech levels of the three conversation conditions were varied. In the simulated test conditions, it was revealed that the room acoustic conditions affected on the speech levels, but the changes of the levels were small.
Characterizing bounded transient sound fields in terms of potential effects on communication in aquatic animals
Authors: Hastings Mardi C.
Published: November 2016

Characterization of underwater transient sound fields in laboratory test chambers and shallow water field locations often used for bioacoustic studies is challenging. For example, no closed form solution exists for transient excitation of a water-filled waveguide with elastic walls. Key issues affecting interpretation of effects on animal communication in these situations are spatially dependent initial conditions that create subsequent sound propagation immediately following termination of the source signal; dissipative effects created by interaction of sound with boundaries; and possible excitation of modal resonances within the confined fluid. Therefore, evaluation and interpretation of data require time domain analysis of sound wave propagation and accurate knowledge of the locations of animal receivers within the sound field. Discrepancies in observed effects reported in the literature can often be explained by proper spatial evaluation of the transient sound field. Several examples will be presented along with results of an experimental evaluation of a water-filled steel waveguide excited by pulsed tones.
Passive travel time tomography using ships as acoustic sources of opportunity
Authors: Cornuelle Bruce , Kuperman William A. , Hodgkiss William S. , Tippmann Jeff , Sarkar Jit , Verlinden Chris , Sabra Karim
Published: November 2016

Active ocean acoustic tomography transmits known source signals and often includes precise source and receiver positioning. Passive estimation of channel Greens functions has been widely demonstrated using both diffuse and concentrated noise sources, including the estimation of ocean sound speeds and currents. The uncertainty of surface ship positions determined by the Automatic Identification System (AIS) translates to travel time variability that is larger than expected from ocean structure at ranges of a few kilometers, but listening to the same source from spatially-distributed receivers adds more data while maintaining the same source position unknowns and may allow useful estimates of bottom depths from bottom-interacting rays and perhaps ocean structure. A fall 2016 experiment will test these ideas in the Santa Barbara Basin using 4 vertical line receiver arrays recording ship-radiated sound for 10 days. Estimates of the expected performance of the estimates of bottom depth and ocean sound speed have been a part of the design process. A general circulation model will be used to increase the efficiency of information use by parameterizing the four-dimensional ocean sound speed structure in terms of initial conditions, boundary conditions, and surface forcing. We present the design process and some proof-of-concept calculations from experimental observations.
Autonomous underwater vehicle navigation using a single tomographic node
Authors: Huang Sheng-Wei , Taniguchi Naokazu , Huang Chen-Fen , Guo Jenhwa
Published: November 2016

This study investigates AUV navigation using the one-way travel-time from an ocean tomographic sensor node. For the study of ocean acoustic tomography, several tomographic nodes are moored in the study area and broadcast m-sequence signals periodically. Each moored node consists of an acoustic transceiver, a GPS module, and a processing unit. The pulse-per-second signal from the GPS module is used to ensure the time synchronization of all the moored nodes. A tomographic sensor is installed on an AUV, serving as a moving node. To obtain highly accurate one-way travel-time between moored nodes and AUV, a chip scale atomic clock is operated on the AUV. Based on dead-reckoning and travel-time measurement, Extended Kalman Filter (EKF) is employed to improve the AUV localization. An experiment was conducted using an AUV and one moored node deployed in a shallow water environment southwest of Taiwan. The measurements from a Doppler velocity logger and a compass are used to calculate the current position of vehicle. The AUV was operated near the surface to obtain the GPS position as the ground truth for the performance evaluation. Preliminary results show the RMS position error between the EKF prediction and ground truth is about 80 m.
Applicability and feasibility studies of coastal acoustic tomography for long-term monitoring of the Indonesian throughflow transport variability
Authors: Syamsudin Fadli , Adityawarman Yudi , Sulistyowati Reni , Sutedjo Bayu , Kaneko Arata , Goda Noriaki
Published: November 2016

We have been trying to monitor the Indonesian Throughflow (ITF), especially to understand how prominent tides and tidal mixing along the ITF main pathways in the Makassar and Lombok straits could transform the Pacific waters into Indian Oceans in the shorter scale and how the ITF variability in longer time scales affect the climate regimes over the region, using a modern and innovative technology of Coastal Acoustic Tomography (CAT). The first trial will be done in the Lombok sill where two 5-kHz CATs and three tide gauge sensors are put in the sill of depth range 200-300 m, lying between Lombok and Nusa Penida islands in the southern Lombok strait by the end of July 2016. The experimental design is a real-time land-cable CAT system, equipped with a 3G mobile antenna for telemetering CAT data to our Maritime Continent Center of Excellence (MCCOE) monitoring center in Jakarta every 10 minutes interval. Results are expected to verify CAT aplicability and feasibility as a cost effective oceanographic instrument to monitor the ITF transport variability along major straits in the Indonesian seas. This talk will also present the possibility of CAT to study mixing processes as important factors to improve model parametrizations.