bluuebunny/crossref_metadata_2025 huggingface dataset:
Auditory selective attention in cochlear implant users
Authors: Choi Inyong , Kim Subong , Gander Phillip E.
Published: November 2016

Selective attention is a fundamental process for our communication in noisy everyday environments. Previous electrophysiologic studies have shown that selective attention modulates the neural representation of the auditory scene, enhancing responses to a target sound while suppressing the background. Given that most cochlear implant (CI) users complain about difficulty understanding speech within background noise, we investigated whether CI users' selective attention performance and underlying neural processes are 1) degraded compared to normal-hearing (NH) listeners and 2) variable among individuals. Using speech stimuli recorded from two competing speakers (male and female), we measured 12 CI and 17 NH listeners' selective attention performance while their cortical neural processes were simultaneously recorded with high-density electroencephalography. While both groups showed above-chance level performance and significantly larger evoked responses to attended stimuli in auditory cortex, the CI group's selective attention performance and neural gain control were poorer than those of NH listeners. Furthermore, a positive correlation was found between the attentional modulation of evoked cortical responses and behavioral performance in the CI group. This result suggests that degraded selective attention processes contribute to an explanation of CI users' difficulty in real-world communication.
Unidirectional wave transmission in a diatomic acoustic metamaterial
Authors: Tan Kwek Tze , Li Bing
Published: November 2016

In this study, we propose the use of dual resonators to design a linear diatomic metamaterial, consisting of several small-sized unit cells, to realize large unidirectional wave transmission in low frequency domain (below 1 kHz). The asymmetric transmission mechanism is theoretically derived and systematically investigated. Numerical results are verified using both lattice structures and continuum models. This passive system does not require external energy supply or result in any frequency conversion. Various wave transmission band gaps, including passing band, stopping band gap, and asymmetric transmission band, can be theoretically predicted and mathematically controlled, which extends the design concept of unidirectional transmission devices. The asymmetric transmission band can be easily altered by careful and deliberate design of the unit size geometrical parameters and material properties. This proposed idea offers a new design concept in wave energy control through unidirectional systems, and may be applied to practical applications such as ultrasound imaging, directional transducer, and noise control.
Nonlinear wave propagation and rotational dynamics In microscale granular media
Authors: Wallen Samuel , Boechler Nicholas
Published: November 2016

Granular media have been shown to exhibit complex dynamics stemming from dispersion and nonlinear particulate interactions. Studies of wave propagation in granular materials composed of ordered arrays of spherical particles interacting via Hertzian contact, known as “granular crystals,” have demonstrated rich nonlinear phenomena, including amplitude-dependent and tunable sound speeds, solitary waves, and discrete breathers. However, most of the past work on granular crystals has considered macroscale spheres, with diameters on the order of centimeters or millimeters. More recent works have explored wave phenomena in microscale granular crystals, having particle diameters of a few microns, which can be fabricated using self-assembly methods, and characterized experimentally using laser ultrasonic techniques. The physics of microscale granular media have been shown to vary drastically from their macroscale counterparts; for example, interparticle adhesion, which is negligible at macroscale, creates an intrinsic static load, which linearizes the contact forces at low amplitude, and can increase the importance of shear interactions and particle rotations. In this work, we present theoretical and computational analyses of nonlinear wave propagation in microscale granular crystals, examining the interplay between nonlinearity and rotational dynamics. In particular, we focus on amplitude-dependent distribution of energy into the translational and rotational modes of a hexagonally close-packed granular crystal.
Effects of prosodic focus on discourse processing by native speakers versus second language learners
Authors: Schafer Amy J.
Published: November 2016

The acquisition of sentence-level prosody and intonation is often challenging for second language learners (L2ers). A speaker’s native and second language can differ at several levels of representation involved in the use of prosody, and so multiple new mappings must be acquired for proficiency in the second language’s prosodic system. This talk examines prosodic focus in English, Japanese, and Korean, and its effects on discourse processing by native speakers and L2ers. Although all three languages make use of prosodic focus, and draw on some of the same acoustic cues to realize it, there are also substantial differences across the languages in how prosodic focus interfaces with other grammatical components. Results from a series of experiments indicate context-sensitivity in how L2ers respond to prosodic focus, showing native-like responses in one discourse context but quite different responses to the same tonal realization in another context. Such findings point to the importance of testing the perception of prosody across multiple grammatical configurations, particularly when investigating language acquisition.
Phonological traits of the Japanese and English spoken by Nikkei speakers in Hawaii
Authors: Long Daniel
Published: November 2016

The Japanese and English language varieties spoken by Nikkei speakers in Hawaii show phonological traits which result from language contact and are uncommon in other varieties. Characteristics of the Japanese spoken by Nikkei speakers which differ from native varieties of the language include the following. Non-geminate and geminate consonants are found in variation ([kappamaki] vs. [tekkamaki], [napa]) and speakers do not hear germination as being phonemically significant. The English labiodental fricative [f] is substituted for the Japanese bilabial fricative [ϕ] ([furo], [furikake]). Vowel lengthening is in inconsistent and not phonemically significant ([obatʃan] contrasts with [oba:tʃan], but [zori] has shortened vowel). Characteristics of Hawaiian English spoken by Nikkei speakers (as well as other ethnicities) include the following. The alveolar flap /ɾ/ ([kaɾai] ‘spicy’) is a phoneme contrasted with /l/ and /r/. Phonotactic rules allow for the /ts/ sequence even in the word-initial position (Hawaiian English [tsunami] vs. General American English [sunami]). In this paper, I will use PRAAT acoustical analyses to illustrate these traits.
The utility of binary-sequence-based signals
Authors: Martin James S.
Published: November 2016

It is frequently implied that there are advantages to the use of signals based on binary sequences for the characterization of linear time invariant (LTI) systems. However, this appears to neglect the role of the interpolation (or reconstruction) filter which is necessary in order to translate discrete sequences into analog signals and to discount the utility of analog signals that are generated by other means. Whereas a discrete sequence may offer desirable or optimal characteristics in comparison with other discrete sequences, these may easily be destroyed by the imposition of the interpolation filter. Similarly, even when these characteristics are substantially preserved in the interrogation signal, they appear consistently to be either mediocre or sub-optimal in comparison to properties of signals that are generated without consideration of binary sequences (e.g., chirps or multi-sine signals). The relative superiority of any signal with respect to the alternatives for interrogation of an LTI system depends on the goals of the characterization and relies on the assumption that some form of nonlinearity will ultimately limit signal amplitude. There does not appear to be any set of goals and limiting nonlinearity that are uniquely suited to binary-sequence-based signals.
Compressive beamforming on a cylindrical array
Authors: Gaumond Charles F. , Botts Jonathan , Summers Jason E.
Published: November 2016

Previously, we presented subarray compressive beamforming for Doppler sensitive midfrequency active sonar on a cylindrical array, focusing on practical application for mitigating Doppler spreading caused by sidelobes and the width of the main lobe [J. Acoust. Soc. Am. 139, 2050 (2016)]. Here, we describe ongoing work focusing on theoretical challenges of adapting compressive beamforming to a cylindrical array. Unlike linear arrays, for example, the optimal sampling for the sound field incident on a cylinder array is not evident. Formal design guidelines for compressive beamforming on a cylindrical array are discussed, guided by the statistical restricted isometry property (StRIP). Prior work has demonstrated subarray compressive beamforming by combining sparse estimates from arbitrarily chosen subarrays. Here, a formal approach is outlined in which StRIP is used to specify minimal subarrays that can individually reconstruct a sound field of a given sparsity, using a new strategy for combining individual estimates. Finally, differences between the fundamental assumptions and resulting limitations of compressive, conventional, and adaptive beamforming techniques are discussed for the cylindrical array geometry and continuous-wave (CW) waveforms of interest, as demonstrated with high-fidelity simulated data. [Work supported by a NAVSEA Phase I SBIR award.]
Sparse tomographic acoustic imaging for environmental noise-source mapping
Authors: Tuna Cagdas , Jones Douglas L. , Zhao Shengkui , Nguyen Thi Ngoc Tho
Published: November 2016

Environmental noise has become a major problem in large cities, increasing health risks for the urban population. Current methods are overly expensive for city-wide noise-monitoring as they generally require dense deployment of fixed microphones for noise-level measurements. We present here alternative sparse tomographic acoustic imaging techniques using arrays with relatively few number of microphones for large-region acoustic noise mapping. We first demonstrate that the locations and sound pressure levels of fixed noise sources sparsely located in a large field are recovered by collecting acoustic data at multiple locations with a portable microphone array for tomographic reconstruction. We then introduce a nonstationary tomographic imaging approach using fixed microphone arrays, which can also capture the intermittent changes in the acoustic field due to transient and/or moving noise sources. We test both the sparse static and dynamic imaging models with acoustic measurements collected with a circular microphone array mounted on an electric vehicle and two prepositioned circular arrays, respectively. The experimental results show the promise of the proposed methodology for low-cost and time-efficient large-scale noise-monitoring.
Study on the prevention of Sinkhole in cities through sound analysis
Authors: Ann Ik-Soo , Bae Myung-Jin
Published: November 2016

Sinkhole is a symptom where the surface area collapses due to lack of supporting power and this is caused by erosion of rocks under the ground or washing away of soil. It is very difficult to say that this sinkhole happening in the downtown is natural but rather resulted from development of underground water, water leakage from the old water pipes buried, and subway construction. As this symptom cannot be predicted visually by looking at the surface it happens unexpectedly and thus has a high possibility of causing a lot of damage. However, active research on developing prevention system by using a special laser video or sound has been carried out. In regards to this, this study suggests methods of preventing downtown sinkhole by analyzing sound. The special feature that classifies sound made from tapping the normal surface and surface of the potential sinkhole is the resonance. If we can have a focused control over these surfaces with high potential of sinkhole we will be able to prevent any accidents taking from this.
Surf infrasound recorded with smartphones during the 2016 Eddie Aikau
Authors: Asmar Karina A. , Garces Milton , Schnurr Julie , Christe Anthony
Published: November 2016

Standardization of signal characteristics is a recurring problem in infrasound detection. We are developing a suite of scalable multiresolution algorithms for infrasound data processing that may include unevenly sampled smartphone barometer data. The algorithms include fundamental methods for Lomb-Scargle periodograms and coherent energy estimators. The implementation of these algorithms will facilitate reproducible and transportable characterization of diverse signatures captured by traditional and ubiquitous global infrasound networks. We discuss the application of these algorithms to surf infrasound signals recorded with smartphones during the 25 February 2016 Eddie Aikau Surf Contest held at Waimea Bay, Oahu.
A basic study of acoustic communications between an autonomous surface vehicle and an autonomous underwater vehicle
Authors: Deguchi Mitsuyasu , Kida Yukihiro , Meguro Koji , Watanabe Yoshitaka , Shimura Takuya , Ochi Hiroshi , Akamine Yukinori
Published: November 2016

Recently, there has been an increasing interest in submarine resources and demand to survey them. For that purpose, Japan Agency for Marine-Earth Science and Technology (JAMSTEC) has developed autonomous underwater vehicles (AUVs). Currently, only one AUV is operated by a research vessel because a mother ship must always follow an AUV. However, to make survey more efficient, the project of multiple AUVs operation, paired with autonomous surface vehicles (ASVs), is being progressed. In this plan, one ASV always monitor status and position of its paired AUV with acoustic communication and localization, and follow it autonomously. The size of an ASV is desired to be as small as possible in terms of operation. Thus, it causes two difficulties. First, such a small ASV rolls and pitches up to a large angle easily. Second, receivers are close to the noise sources, such as thrusters or sea surface. As a basic study, at-sea experiments were demonstrated with various types of modulation using a prototype of an ASV. The results show that OFDM signals are more sensitive to rolling and pitching movement than single carrier modulation (SCM) and that it is possible to achieve demodulation with SCM in spite of noise as mentioned above.
Separation of individual instrumental sounds in monaural music signals applying a modified Wiener filter on the time-frequency plane
Authors: Fukayama Yukio , Watanabe Masafumi
Published: November 2016

This research seeks a signal processor for random processes which is suitable for distilling melodies played by a particular instrument from music ensemble. The proposed processor sequentially evaluates statistics of the mixture assuming a model of music sound and employs a modified Wiener filter so as to separate the mixture of time-variant and correlated random processes such as amplitude changing tones and same pitch tones by different instruments. The model of music sounds is assumed to be a weighted mixture of each standard tone, which is a typical tone of every pitch played by a particular instrument with invariant power, where the weight is called the amplitude factor corresponding to the standard tone. Thus, the proposed processor, at first, evaluates the current amplitude factors which are the the coordinates of the mixture where the standard tones form oblique basis. Then, according to evaluated coordinates of the mixture, its current autocorrelation and cross-correlation to each standard tone are calculated for obtaining the modified Wiener filter. Finally, the filter separates the mixture into individual tone, which is a component of a particular pitch played by an instrument with variant power, and combination of which provides a melody.
Shouted speech detection using hidden markov model with rahmonic and mel-frequency cepstrum coefficients
Authors: Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu , Nanjo Hiroaki
Published: November 2016

In recent years, crime prevention systems have been developed to detect various hazardous situations. In general, the systems utilize the image information recorded by a camera to monitor the situations. It is however difficult to detect them in the blind area. To address the problem, it is required to utilize not only image information but also acoustic information occurred in such situations. Our previous study showed that two acoustic features including rahmonic and mel-frequency cepstrum coefficients (MFCCs) are effective for detecting the shouted speech. Rahmonic shows a subharmonic of fundamental frequency in the cepstrum domain, and MFCCs represent coefficients that collectively make up mel-frequency cepstrum. In this method, a shouted speech model is constructed from these features by using a gaussian mixture model (GMM). However, the previous method with GMM has difficulty in representing temporal changes of the speech features. In this study, we further expand the previous method using hidden Markov model (HMM) which has state transition to represent the temporal changes. Through objective experiments, the proposed method using HMM could achieve higher detection performance of the shouted speech than the conventional method using GMM.
Vandermonde transform for complex residual based on complex speech analysis
Authors: Shimabukuro Taishi , Funaki Keiichi
Published: November 2016

Vandermonde transform using the orthogonal basis is a time-frequency transform method that can describe a speech and audio signal to be uncorrelated and sparse, and can be considered to be also effective in speech processing. In particular, when applied to the residual considered to be effective in a pitch estimation of speech and speaker recognition. On the other hand, we have already proposed Time-Varying Complex AR (TV-CAR) speech analysis for an analytic signal. The TV-CAR analysis can improve spectral estimation accuracy in the low frequencies and it can estimate the complex AR coefficients for each sample. As a result, in the estimated complex residual signal, formant components are more subtracted from speech signal. It can perform better on pitch estimation, or so on. In this paper, we evaluated the time-frequency analysis using the Vandermonde transform based on an Orthogonal Matching Pursuit (OMP) method of the literature using the complex AR residual. The performance is evaluated for several orthogonal basis such as Fourier basis, Cos basis, K-L transform, and Gabor basis. Furthermore, the Ramanujan-sum is also introduced as the basis function to realize time-frequency transform.
Sound source separation using image signal processing based on sparsity of sound field
Authors: Ito Masaaki , Ozawa Kenji , Morise Masanori , Shimizu Genya , Sakamoto Shuichi
Published: November 2016

Microphone arrays have been used to separate sound sources to improve speech recognition in a noisy environment. We propose a method using image signal processing to achieve highly accurate sound source separation. The microphone array is 1-m long and consists of eight microphones. Temporal sequences of the sound pressures obtained from the eight microphones are converted into sequences of luminance. These are arranged in parallel in an image, which is referred to as aspatio-temporal sound pressure distribution image. Sparse modeling using L1 regularization (Lasso) is applied to the image for restoring a high-resolution image. The spatial spectrum of the restored image is obtained using a two-dimensional fast Fourier transform (FFT) algorithm. In this spectrum, the angle of a line through the center denotes the arrival direction of sound, and the distance from the center indicates its frequency. By extracting a line from the spectrum, the sound source can be separated. A computational experiment revealed that high-resolution sound field obtained by a 512-microphone array could be restored using the proposed method. Moreover, SNR was improved by 32.1 dB when two sounds arrived 45° apart, indicating sufficient performance to extract a desired sound. [Work supported by JSPS KAKENHI Grant (16K06384).]
Development of an underwater sound monitoring system using two hydrophones for a remotely operated underwater vehicle
Authors: Kim Sea-Moon , Byun Sung-Hoon , Choi Hyun-Taek , Lee Chong-Moo
Published: November 2016

Recently, a sound monitoring system for hearing fishery sound using three hydrophones installed on an ROV (Rountree and Juanes, 2010) has been developed. However, the operation of the system was not successful because of the high level of the self-noise generated by the ROV. Its thrusters had to be turned off to reduce the self-noise and listen to fishery sound clearly. Currently, Korea Research Institute of Ships and Ocean Engineering (KRISO) is developing a light-work class ROV and a similar sound monitoring system was developed for the ROV. The design of the system was focused on the following two strategies: (1) minimization of self-noise effect; (2) reproducing sound for operators to identify source direction. To minimize the self-noise effect, a digital filter with seven filter banks with octave bandwidth was implemented. A time-domain beamforming technique has been applied to analyze the source location. Reproducing sound signals for the two speakers in the control room were generated according to the estimated source location. The performance of the algorithm was verified by using a receiver array signal simulator. Its detail characteristics will be presented in the talk. [This work was financially supported by the research project # 20130197 funded by KIMST.]
Stereophonic reproduction based on multi-beam-forming with curved-type parametric array loudspeaker
Authors: Komori Shinya , Hasegawa Nana , Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu
Published: November 2016

A parametric array loudspeaker (PAL) has a flat surface arrangement of ultrasonic transducers and achieves a sharper directivity by utilizing an ultrasound wave. It can form a narrow audible space called an “audio spot” that can provide a target sound to a particular listener. In addition, the PAL can achieve a stereophonic reproduction by designing the audio spots for the positions of ears on the basis of delay-and-sum beamforming. However, designing wider audio spots for several listeners is difficult for the conventional PAL because it has overly sharp directivity. To solve this problem, we propose a new stereophonic reproduction with a curved-type PAL that can expand the audio spots. The curved-type PAL consists of ultrasonic transducers arranged on a concave surface. It can form a wider directivity compared with the conventional flat-type PAL. The proposed method steers the acoustic beam by designing the focal point to the positions of ears, and the radius of the curved-type PAL varies the width of the acoustic-beams. In this study, we carried out an objective evaluation experiment to confirm the effectiveness of the proposed method. As a result, we confirmed that the proposed method can design two wider acoustic beams for the stereophonic reproduction.
Visualization of one source in a two sources sound field by estimating source positions
Authors: Maureen  , Haneda Yoichi
Published: November 2016

Sound field visualization is important in many applications. Acoustic holography (AH) uses a linear microphone array to visualize sound fields. However, in general, all existing sound sources will appear when AH is used. In this paper, we propose a method that visualizes a single source from two sources in a sound field by estimating source positions and separating them. In the proposed method, sound source positions are estimated based on the microphone input signals’ phase components for each frequency. The source positions of all frequencies are placed in a histogram that expresses each position’s quantity, which facilitates acquisition of accurate source positions. We determine one source and select frequency bins that indicate the obtained source position. We can also virtually extrapolate the microphone array length by solving the wave propagation model using the least-squares method. The results of sound field visualization experiments conducted considering the depth direction and using the one-dimensional near-field acoustic holography inverse propagation method show that the proposed method can separate a single source by selecting the histogram peak even when the wavefronts of two sources appear simultaneously. Future work will attempt to restore only one source or both two sources to time signal.
Segmental conditional random fields audio-to-score alignment distinguishing percussion sounds from other instruments
Authors: Noguchi Ayako , Sako Shinji , Kitamura Tadashi
Published: November 2016

Audio-to-score alignment is useful technique because it can be widely applied to many practical applications for musical performance. However, it is still open problem due to the complexity of audio signal especially in the polyphonic music. Additionally, performing in real-time is also important in practical situation. In this study, we propose a new alignment method based on segmental conditional random fields (SCRFs). The attractive feature of this method is utilizing to distinguish percussion sounds from the other instruments. In general, percussion sounds have a role in managing whole music. Moreover, performers can pick up the percussion sounds from the others by hearing whole sound thanks to their unique features of the sound. In the field of score alignment, hidden Markov models (HMMs) or CRFs was used in previous studies including our previous one. In summary, these methods were formulated as a matching problem of the state sequence of mixed notes with the audio feature sequence. In this study, we extend our previous method by combining an additional state which represents percussion sounds. Furthermore, we introduce the balancing factor to control the importance of classifying feature functions. We confirmed the effectiveness of our method by conducting experiments using RWC music database.
Directivity control of a compact circular loudspeaker array based on selected orders of circular harmonic expansion
Authors: Sato Koya , Haneda Yoichi
Published: November 2016

We investigated directivity control of a compact circular loudspeaker array that can be attached to the arm. The loudspeaker array can be used for various wearable systems, such as personal audio reproduction systems. A personal audio system often requires that sound leakage outside the listening area in three-dimensional space be minimized. Digital filters that control directivity are usually designed using the multi-point control method. However, the multi-point method causes large filter gains at low frequencies, owing to unstable inverse matrix calculation, which results in sound distortion during practical use. Many methods that use a regularization parameter to overcome the unstable inverse matrix have been proposed; however, they are difficult to apply directly. Thus, after studying the relationship between orders of circular harmonic expansion and filter gain, we propose a stable digital filter design based on appropriately selected orders of circular harmonic expansion for each frequency band. Directivity simulations conducted using the measured circular loudspeaker array response based on the proposed method indicate that the proposed method achieves a more stable filter gain than the multi-point control method. Further, it controls three-dimensional directivity by sacrificing a narrow directivity at low frequencies.
The largest Cramer-Rao lower bound for variance estimation of acoustic signals
Authors: Takahashi Hiroki , Murakami Takahiro
Published: November 2016

The Cramer-Rao Lower Bounds (CRLBs) for variance estimation of acoustic signals is discussed. The CRLB expresses the lower limit of the variance of estimation errors of an unbiased estimator. It is empirically known that the accuracy of variance estimation is improved as the signal length is long because the CRLB for variance estimation is inverse proportion to the length of a signal. Therefore, when the variance of estimation errors is desired to be lower than a certain value, the signal length must be longer than the length that is determined by the CRLB. If the distribution of a signal is known, the CRLB can be easily formulated. In practice, however, acoustic signals follow various distributions and they are unknown in advance. In our research, we focus on the distribution that gives the largest CRLB for variance estimation. In order to find such a distribution, we compare the CRLBs for variance estimation using various signal models. Our results conclude that the Gaussian distribution gives the largest CRLB for variance estimation.
Novel sound logger system for sound source localization using global navigation satellite system
Authors: Takasugi Chikao , Takeyama Keisuke , Fukaishi Takuya , Koike Yoshikazu , Hojo Harimasa , Matsuno Tatsuo
Published: November 2016

Accurate sound localization is a very important technology to solve noise problem in the association with air plane, railroad, or wind-power generation and so on. In order to localize the sound source, direction of arrival (DOA) estimations in a couple of positions at a distance of several hundred meters are required. Generally, the measurement start timing is synchronized in each estimation position using 1PPS time signal from the global navigation satellite systems (GNSSs). However, the slight deviation in each measuring instrument occurs at the start time. Because the sampling clock of ADC (analog digital converter) is not synchronized. Therefore, the authors propose the application of time synchronization signal obtained from GNSS module to a sampling clock on an acoustic measurement in a wide range. Current some GNSS modules provide a pulse multiplied by 1PPS, which we call a time-synchronization signal. For the purpose, the authors have developed a logger apparatus which consists of a microphone amplifier, a GNSS, and an ADC module. The developed logger apparatuses are employed to perform sound source localization experiment in the field. As a result, the precision of sound localization within 2 meters is achieved successfully when the microphone sets at a distance of 10 m.
Iterative spectral subtraction based on variable frame processing for musical noise reduction
Authors: Wakabayashi Yukoh , Nagano Yuki , Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu
Published: November 2016

In recent years, speech communication and speech applications like speech recognition are widely used with the development of information equipment. But a speech corrupted by noises makes fluent communication difficult and causes the poor performance of speech applications. Thus, an accurate noise reduction method is necessary to solve these problems. Spectral subtraction (SS) and iterative spectral subtraction (I-SS) are well-known noise reduction methods that subtract estimated noise from noisy speech in the power spectral domain. However, SS and I-SS generate an artificial noise called musical noise. The noise is non-stationary and has an offensive sound. Miyazaki et al. have proposed a musical-noise-free SS that determines subtraction parameters such that musical noise is not generated. The method effectively reduces the musical noise but requires estimation of the statistical characteristics of the observed noise. Meanwhile, we tackle the musical noise problem by proposing a new I-SS based on variable frame processing. In this processing, variable frame analysis is carried out in each iteration to reduce the musical noise. Evaluation experiments show that the proposed I-SS achieved a high SDR and resulted in less musical noise compared with constant frame processing.
Projection of a virtual speaker into a vehicle using sound field control
Authors: Yamamura Toshiki , Ishiguro Yoshio , Nishino Takanori , Takeda Kazuya
Published: November 2016

In this study, we propose a system which projects a virtual speaker into a vehicle by controlling the sound field of the interior of the vehicle. The system uses a sound field reproduction technique to reproduce a point source in the vehicle representing a virtual speaker, enabling drivers to communicate with virtual speakers as if they were riding in the same vehicle. The system is based on a sound field reproduction technique known as Wave Field Synthesis (WFS), which can represent the depth of a sound source by reproducing its wavefront precisely. However, since WFS is generally implemented in an ideal environment, the influences of reflection and reverberation are not generally taken into consideration. Therefore, in order to apply WFS to an in-vehicle system, it is necessary to investigate the effects of reflection and reverberation on WFS within a vehicle. To do so, we compare the accuracy of the perception of locations of sound point sources reproduced using WFS with a 32 channel loudspeaker array in both a vehicle and a low reverberation room. We also compare perception accuracy of the proposed system with perception accuracy when a loudspeaker is placed in the same position as the virtual speaker.
The mathematical paradigm of “copula” to model a faded signal received by multiple sensors
Authors: Huang Shih-Hao , Huang Mong-Na L. , Wong Kainam T.
Published: November 2016

In sonar, radar, or wireless communication that deploys multiple sensors in the receiver for a fading channel, the received signal (at each time instant) may be modeled as a stochastic vector whose elements correspond to the various sensors of the receiver. This random signal’s spatial statistics may be described (at each time instant) by a multivariate probability distribution. For this multivariate distribution to have ARBITRARILY different univariate marginal distributions and/or an ARBITRARY spatial correlation function—the mathematical paradigm of “copula” has recently been advanced in Huang, Huang, Wong, and Tseng, “Copula—to model multi-channel fading by correlated but arbitrary Weibull marginal, giving a closed-form outage-probability of selection-combining reception,” IET Microwaves, Antennas & Propagation, December 2015. That seminal work shows how the mathematical paradigm of copula can allow unprecedented flexibility in modeling and can offer unparalleled mathematical simplicity in expressing the receiver’s outage probability in a closed and simple mathematical form explicitly in terms of the model parameters. That reference focuses on a particular copula called the survival-Gumbel copula. This paper will explore other copulas.
Effect of internal waves on acoustic energy structure in the northern East China Sea
Authors: Park Jungyong , An Sangkyum , Choo Youngmin , Seong Woojae
Published: November 2016

Relation between depth direction acoustic energy and sound speed fluctuation caused by internal wave activity observed during SAVEX15 experiment is investigated. In the SAVEX15 region (northern East China Sea with water depths around 100 m), underwater sound channel was observed during May 2015 along with extensive internal wave activities. When the source is near the sound channel axis, most of the acoustic energy is trapped in the channel. However, internal waves change the sound speed profile, which results in spreading the acoustic energy over depth. Measured data during the experiment is used to investigate the relation between acoustic energy structure and internal wave activity. Parameters reflecting their variabilities are defined. Two main parameters are standard deviation of the acoustic energy integrated over depth and the sound speed gradient at the channel axis, which reflects the internal wave activity. The correlation of two parameters is used to show the effect of internal waves on the changes of acoustic energy distribution.
Measurements of high frequency acoustic transmission loss during SAVEX15
Authors: Son Su-Uk , Choi Jee Woong , Lee Seung Woo , Nam SungHyun , Cho Sungho
Published: November 2016

Transmission loss measurements in high frequency range along with the ocean environmental measurements were made during the Shallow-water Acoustic Variability EXperiment 2015 (SAVEX15) in the south of Jeju Island, Korea, where the water depth is about 100 m. Two vertical line arrays (VLAs) were moored with a distance of 5.5 km, covering the water column between 24 and 80 m. Continuous waves in a frequency band of 10-25 kHz were transmitted from a vertical source array at depths of 30-45 m. Sound speed and temperature profiles were measured using a standard/stationary conductivity-temperature-depth (CTD), underway CTD, and temperature loggers attached to the two VLAs. The measured sound speed profiles consistently reveal a sound speed minimum layer (SSML) at depths between 30 to 50 m. During the SAVEX15 period, the perturbations in the SSML derived by various kinds of internal waves were observed. In this talk, the temporal fluctuations of the measured transmission loss are presented, and the results are discussed in comparison with the internal wave field in the region. [Work supported by ONR, KIOST, SNU, and ADD.]
Geoacoustic inversions in the East China Sea for a 3D sediment model
Authors: Potty Gopu R. , Miller James H. , Dosso Stan E. , Dettmer Jan , Bonnel Julien
Published: November 2016

Geoacoustic inversions using wide-band acoustic sources (WBS) deployed during the Asian Seas International Acoustic Experiment (ASIAEX) along a circular path of radius 30 km centered on a vertical hydrophone array will be used to construct a pseudo 3D model of the seabed sediments. The acoustic data used in the inversions consist of modal dispersion as estimated using a warping transform technique. Existing information about the bottom includes historic data, seismic records, and gravity and piston cores which indicate significant spatial variability in the thickness of sediment layers and types of sediment. The geoacoustic inversion approach is based on trans-dimensional Bayesian methodology in which the number of sediment layers is included as unknown in addition to the layer parameters. A sediment-acoustic model, such as viscous grain shearing, will be used to estimate the wave parameters corresponding to the sediment parameters. One-dimensional (depth-dependent) inversions will be applied along the various acoustic propagation paths to construct a pseudo 3D sediment model using interpolation. [Work supported by the Office of Naval research.]
Regional variations of Saccharina japonica forests distribution around the Tsugaru Strait by acoustic method
Authors: Shao Huamei , Minami Kenji , Fujikawa Yoshikazu , Yusa Takashi , Yotsukura Norishige , Nakaoka Masahiro , Miyashita Kazushi
Published: November 2016

As the global warming, commercially important species Saccharina japonica disappeared, what is called sea desertification occurred in some coastal waters around the Tsugaru Strait recently. For the sustainability of coastal ecosystem and resources use, understanding the distribution and variation of S. japonica forests is necessary. Surveys for the distribution of S. japonica forests by acoustic method were conducted in four coastal waters: ① Osatsube, Hakodate, the Pacific Ocean side of Hokkaido; ② Ishizaki, Hakodate in Tsugaru Strait of Hokkaido; ③ Okoppe, Oma, in Tsugaru Strait of Aomori Prefecture; ④ Shiriyazaki, Higashidori, the Pacific Ocean side of Aomori Prefecture. Thickness and spatial distributions of seaweed were obtained by acoustic method and density of sea urchins by visual observations. In Osatsube coastal water, main species was S. japonica and which forests were most among survey regions. In Ishizaki and Okkope coastal waters, Sargassum spp. was also luxuriant except S. japonica. In Shiriyazaki coastal water, S. japonica dominated and a greater number of sea urchins were observed than other regions. In regions where little benefit from Oyashio Current, Sargassum spp. grew thickly when sea urchins were rare whereas sea urchin barren occurred when sea urchins with high density.
Annual change in the distribution of Sargassum horneri using a quantitative echo sounder in Yamada Bay, Iwate, Japan
Authors: Minami Kenji , Kita Chihomi , Sawai Masayuki , Shirakawa Hokuto , Shao Huamei , Tomiyasu Makoto , Kobayashi Motoki , Miyashita Kazushi
Published: November 2016

Sargassum horneri, a type of seaweed, has attracted attention as a new fisheries resource in Japan because it contains large amounts of functional ingredients. Yamada Bay (16 km2), in Iwate, Tohoku, Japan, was seriously damaged by the 2011 tsunami, and a sustainable new fisheries resources would aid in the reconstruction of local industries. In this study, we estimated annual change in the distribution of S. horneri using a quantitative echo sounder for sustainability of S. horneri in Yamada Bay. Surveys were conducted in spring (during the growth season) 2013, 2014, and 2015. The acoustic data were obtained using a 120 kHz quantitative echo sounder (KCE300, Sonic Co.). S. horneri were detected on the rope of aquaculture facilities and the rock reef in the coastal sea area. Growth conditions were better in 2013 than those in 2014 and 2015. In 2013, water temperature in Yamada Bay was lower than in 2014 and 2015 because of the influence of cold water from the strong Oyashio Current. It was concluded that primary production of S. horneri in Yamada Bay is influenced by the Oyashio Current, causing their abundance to fluctuate annually.
Accuracy verification of tilt-angle estimation by split-beam echosounder: Comparison with acceleration data-logger data
Authors: Shirakawa Hokuto , Minami Kenji , Kawauchi Yohei , Tomiyasu Makoto , Tsuda Yuichi , Umetsu Takeru , Shao Huamei , Kobayashi Motoki , Iino Yuki , Ogawa Masahiro , Miyashita Kazushi
Published: November 2016

Fish tilt-angle is one of the most important factors for fish target strength (TS) estimation, because the little change varies TS value dramatically. Split-beam echosounder collects a fish position with 3-D coordinates. Hence, if fish swim within acoustic beam continuously, the tilt-angle can be estimated from the gap between two positions easily. However, the accuracy of the estimation depends on performance of the echosounder. We verified accuracy of the estimated tilt-angle by comparing high-resolution angle from an acceleration data logger. Experiment was conducted at a large experimental tank (W:10m x D:5m x H:6m) in HRCFO. A 2-axis acceleration data logger (φ15 mm, L: 53 mm, M190L-D2GT, Little Leonardo, Inc.) was tethered into the tank, and towed at respective angle from -50 to 50 degree (1 degree pitch). Positions of the logger echos (from KCE300, Sonic Co., 120kHz) were exported by algorithms of Echoview7.1. After this process, the tilt-angles were calculated. 28 logger echos were identified from the acoustic data, and the estimated tilt-angles was from -45.59° to 32.87 °. Significant correlation of tilt-angles (logger vs. acoustic data) was found (p > 0.01, r2 = 0.90). This result shows high precision of the estimated tile angle from the split-beam echosounder data.
Distributional factors of juvenile Walleye Pollock in the Musashi Bank area, northern Japan Sea
Authors: Umetsu Takeru , Tsuda Yuichi , Ito Yasushi , Sirakawa Hokuto , Minami Kenji , Miyashita Kazushi
Published: November 2016

Most of juvenile walleye pollock (Gadus chalcogrammus) in the northern Japan Sea migrate to the bottom layer with 150-200 m depth of Musashi bank area in summer. Clarification of distribution factors of nursery area after the migration is important for resource management. We examined the factors of juvenile pollock after migration to bottom in summer 2014 and 2015. The distribution of density of juvenile pollock was obtained by quantitative echo sounder. The distribution of bottom sediment was discriminated by the side scan sonar. The bottom backscattering strength and sediment data were jointed to analyze bottom sediment of quantitative echo sounder survey transect line based on the Roxann method. In the result, bottom sediment was separated into clay, sand, and boulders. Distribution density of juvenile pollock was high in boulders area. Additionally, the density of juvenile pollock was higher at northward upslope than gentle landform and else way upslope in all boulders area. It is considered that northward ocean flow collided with the northward upslope of boulders make upward flowing and stagnate water, and the water forms the retention space of prey plankton that is favorable for juvenile pollock.
An approach to the design of drama theaters for speech clarity
Authors: Miller Gregory A. , Strong John , Giegold Carl , Brill Laura C.
Published: November 2016

In drama theaters, the unamplified human voice is used to convey both information and nuanced emotional cues to large groups of listeners. Supportive sound reflective surfaces are relatively distant from the actors, and the inherent (and desirable) reverberance of a larger space run counter to many accepted practices for achieving speech clarity in smaller spaces. Some theater configurations, such as thrust and in-the-round, are even arranged so that actor faces away from large segments of the audience at any given time, offering further challenges to the connection between actor and audience. This paper will present an understanding of the way that people listen to speech in large rooms, and approaches to room design to support the clarity of unamplified speech. A case study will be presented describing the development of theater design from initial concept through computer modeling and finally with three dimensional measurements and visualizations of the acoustic environment in the completed room.
Controlled minimum background sound, the missing link in acoustic design
Authors: Doedens Ric
Published: November 2016

Current industry standards related to controlling background sound in occupied spaces are all based on “not to exceed” maximum levels. At the other end of the scale, it is assumed that we cannot predict with any degree of certainty how low the background sound level may descend, what its spectral composition is, and when or how it may vary. Given that the “signal to noise ratio” is fundamental to all communication based acoustic design, the absence of this knowledge presents significant challenges. As a consequence, we adapt our acoustic designs to compensate for our inability to control minimum noise levels by over-specifying STC, CAC, and NRC values in order to obtain the desired results. And yet, precise generation of controlled minimum background sound levels is readily available. This session explores the negative impact of traditional compensatory acoustic design practises, the significant value of incorporating predictable/consistent minimum background sound levels into acoustic design planning, and makes a case for establishing minimum background sound level industry standards.
Self-fitting hearing aids reveal strong individual preferences in noisy rooms
Authors: Nelson Peggy B. , VanTasell Dianne
Published: November 2016

Noisy restaurants are among the most challenging listening situations for listeners with hearing loss, but little data are available to optimize hearing aids for noisy situations. In addition, evidence suggests that listeners are less satisfied with hearing devices in noisy listening environments. New technology from EarMachine(c) allows for individuals with hearing loss to self-adjust hearing aid amplification parameters. Self-adjustment technology can provide data to empirically answer questions about listener preferences in noise. Results from 30 listeners demonstrated self-selected gain profiles that reveal very large individual preferences that are not well predicted by age, gender, or hearing loss. These results indicate that there are strong preferences for listening in noisy rooms that are not yet well understood. [Funding provided by NIDCD R01 DC013267.]