• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.230-237
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• 2005

The angle-tracking of maneuvering targets is required to the state estimation and classification of targets in underwater acoustic systems. The Problem of angle-tracking multiple closed and crossing targets has been studied by various authors. Sword et al. Proposed a multiple target an91e-tracking algorithm using angular innovations of the targets during a sampling Period are estimated in the least square sense using the most recent estimate of the sensor output covariance matrix. This algorithm has attractive features of simple structure and avoidance of data association problem. Ryu et al. recently Proposed an effective multiple target angle-tracking algorithm which can obtain the angular innovations of the targets from a signal subspace instead of the sensor output covariance matrix. Hwang et al. improved the computational performance of a multiple target angle-tracking algorithm based on the fact that the steering vector and the noise subspace are orthogonal. These algorithms. however. are ineffective when a subset of the incident sources are coherent. In this Paper, we proposed a new multiple target angle-tracking algorithm for coherent and incoherent sources. The proposed algorithm uses the relationship between source steering vectors and the signal eigenvectors which are multiplied noise covariance matrix. The computer simulation results demonstrate the improved Performance of the Proposed algorithm.

• Phonetics and Speech Sciences
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• v.14 no.4
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• pp.45-56
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• 2022

This paper proposes an optimal methodology for automatically diagnosing and predicting the severity of the chronic kidney disease (CKD) using patients' utterances. In patients with CKD, the voice changes due to the weakening of respiratory and laryngeal muscles and vocal fold edema. Previous studies have phonetically analyzed the voices of patients with CKD, but no studies have been conducted to classify the voices of patients. In this paper, the utterances of patients with CKD were classified using the variety of utterance types (sustained vowel, sentence, general sentence), the feature sets [handcrafted features, extended Geneva Minimalistic Acoustic Parameter Set (eGeMAPS), CNN extracted features], and the classifiers (SVM, XGBoost). Total of 1,523 utterances which are 3 hours, 26 minutes, and 25 seconds long, are used. F1-score of 0.93 for automatically diagnosing a disease, 0.89 for a 3-classes problem, and 0.84 for a 5-classes problem were achieved. The highest performance was obtained when the combination of general sentence utterances, handcrafted feature set, and XGBoost was used. The result suggests that a general sentence utterance that can reflect all speakers' speech characteristics and an appropriate feature set extracted from there are adequate for the automatic classification of CKD patients' utterances.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.8-16
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• 2004

An inversion method is presented for the determination of the compressional wave speed, compressional wave attenuation, thickness of the sediment layer and density as a function of depth for a horizontally stratified ocean bottom. An experiment for estimating those properties was conducted in the shallow water of South Sea in Korea. In the experiment, a light bulb implosion and the propagating sound were measured using a VLA (vertical line array). As a method for estimating the geoacoustic properties, a coherent broadband matched field processing combined with Genetic Algorithm was employed. When a time-dependent signal is very short, the Fourier transform results are not accurate, since the frequency components are not locatable in time and the windowed Fourier transform is limited by the length of the window. However, it is possible to do this using the wavelet transform a transform that yields a time-frequency representation of a signal. In this study, this transform is used to identify and extract the acoustic components from multipath time series. The inversion is formulated as an optimization problem which maximizes the cost function defined as a normalized correlation between the measured and modeled signals in the wavelet transform coefficient vector. The experiments and procedures for deploying the light bulbs and the coherent broadband inversion method are described, and the estimated geoacoustic profile in the vicinity of the VLA site is presented.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8A
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• pp.777-784
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• 2010

Recently, the necessity of underwater communication and demand for transmitting and receiving various data such as voice or high resolution image data are increasing as well. The performance of underwater acoustic communication system is influenced by characteristics of the underwater communication channels. Especially, ISI(inter symbol interference) occurs because of delay spread according to multi-path and communication performance is degraded. In this paper, we study the OFDM technique to overcome the delay spread in underwater channel and by using CP, we compensate for delay spread. But PAPR which OFDM system has problem is very high. Therefore, we use DFT-spread OFDM method to avoid nonlinear distortion by high PAPR and to improve efficiency of amplifier. DFT-spread OFDM technique obtains high PAPR reduction effect because of each parallel data loads to all subcarrier by DFT spread processing before IFFT. In this paper, we show performance about delay spread through OFDM system and verify method that DFT spread OFDM is more suitable than OFDM for underwater communication. And we analyze performance according to two subcarrier mapping methods(Interleaved, Localized). Through the simulation results, performance of DFT spread OFDM is better about 5~6dB at $10^{-4}$ than OFDM. When compared to BER according to subcarrier mapping, Interleaved method is better about 3.5dB at $10^{-4}$ than Localized method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.8
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• pp.1631-1636
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• 2015

With the advance of wireless communication technology in terrestrial area, underwater communication is also evolving very fast from a simple point-to-point transmission to an elaborate networked communications. Underwater acoustic channel has quite different features comparing with the terrestrial radio channel in terms of propagation delay, Doppler shift, multipath, and path loss. Thus, existing technologies developed for terrestrial communication might not work properly in underwater channel. Especially medium access control (MAC) protocols which highly depend on propagation phenomenon should be newly designed for underwater network. CSMA/CA has drawn lots of attention as a candidate of underwater MAC protocol, since it is able to resolve a packet collision and the hidden node problem. However, a received signal could be degraded by the interferences from the nodes locating outside the receiver's propagation radius. In this paper, we study the effects of interference on the CSMA/CA based underwater network. We derived the SNR with the interference using the sonar equation and analyzed the degradation of the RTS/CTS effects. These results are compared with the terrestrial results to understand the differences. Finally we summarized the design considerations in CSMA/CA based underwater network.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2E
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• pp.91-97
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• 1995

This paper describes the implementation of a multi-channel audio codec for HETV. This codec has the features of the 3/2-stereo plus low frequency enhancement, downward compatibility with the smaller number of channels, backward compatibility with the existing 2/0-stereo system(MPEG-1 audio), and multilingual capability. The encoder of this codec consists of 6-channel analog audio input part with the sampling rate of 48 kHz, 4-channel digital audio input part and three TMS320C40 /DSPs. The encoder implements multi-channel audio compression using a human perceptual psychoacoustic model, and has the bit rate reduction to 384 kbit/s without impairment of subjective quality. The decoder consists of 6-channel analog audio output part, 4-channel digital audio output part, and two TMS320C40 DSPs for a decoding procedure. The decoder analyzes the bit stream received with bit rate of 384 kbit/s from the encoder and reproduces the multi-channel audio signals for analog and digital outputs. The multi-processing of this audio codec using multiple DSPs is ensured by high speed transfer of date between DSPs through coordinating communication port activities with DMA coprocessors. Finally, some technical considerations are suggested to realize the problem of real-time operation, which are found out through the implementation of this codec using the MPEG-2 layer II sudio coding algorithm and the use of the hardware architecture with commercial multiple DSPs.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.87-93
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• 2014

If one identifies the detailed distribution of pressure and axial velocity at a source plane, the position and strength of major noise sources can be known, and the propagation characteristics in axial direction can be well understood to be used for the low noise design. Conventional techniques are usually limited in considering the constant source characteristics specified on the whole source surface; then, the source activity cannot be known in detail. In this work, a method to estimate the pressure and velocity field distribution on the source surface with high spatial resolution is studied. The matrix formulation including the evanescent modes is given, and the nearfield measurement method is proposed. Validation experiment is conducted on a wide duct system, at which a part of the source plane is excited by an acoustic driver in the absence of airflow. Increasing the number of evanescent modes, the prediction of pressure spectrum becomes further precise, and it has less than -25 dB error with 26 converged evanescent modes within the Helmholtz number range of interest. By using the converged modal amplitudes, the source parameter distribution is restored, and the position of the driver is clearly identified at kR = 1. By applying the regularization technique to the restored result, the unphysical minor peaks at the source plane can be effectively suppressed with the filtering of the over-estimated pure radial modes.

• Journal of the Korean Society for Nondestructive Testing
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• v.24 no.1
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• pp.15-21
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• 2004

It is possible to detect flaws in pipelines without interruption using all EMAT transducer because it is a non-contact transducer which can transmit ultrasonic waves into specimens without couplant. And it ran easily generate guided waves desired in each specific problem by altering the design of coil and magnet. In the present work, EMAT systems have been fabricated to generate surface waves, and selectively the plate wave of $A_1\;or\;S_1$ mode. The surface wave of 1.5MHz showed a good signal-to-noise ratio without distortion in its propagation along a pipeline, while the $S_1$ mode of 800kHz and the $A_1$ mode of 940kHz were distorted according to their dispersive properties. The wider the excitation pulse becomes, the better the mode selectivity of the plate waves becomes. A pipe of 256mm inner diameter and 5.5m thickness with 5 flaws was used for comparing the flaw detectability among the modes under consideration.

• The Journal of the Acoustical Society of Korea
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• v.40 no.4
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• pp.320-329
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• 2021

In this paper, an empirical model of air bubble size to be applied to an air masker for reduction of underwater radiation noise is presented. The proposed model improves the divergence problem under the low-speed flow condition of the existing model derived using Rayleigh's jet instability model and simple continuity condition by introducing a jet flow velocity of air. The jet flow velocity of air is estimated using the bubble size where the liquid is quiescent. In a medium without flow, the size of the bubble is estimated by an empirical method where bubble formation regime is divided into a laminar-flow range, a transition range, and a turbulent-flow range based on the Reynolds number of the injected air. The proposed bubble size model is confirmed to be in good agreement with the Computational Fluid Dynamics (CFD) analysis result and the experimental results of the existing literature. Using the acoustic inversion method, the air bubble population is estimated from the insertion loss measured during the air injection experiment of the air- masker model in a large cavitation tunnel. The results of the experiments and the bubble size model are compared in the paper.

• KIPS Transactions on Computer and Communication Systems
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• v.11 no.7
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• pp.217-224
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• 2022

Today, with the development of technology and industry, fire accidents in special buildings are increasing as special buildings increase. However, despite the rapid development of information and communication technology, human casualties are steadily occurring due to the underdeveloped and ineffective indoor fire alarm system. In this study, we confirmed that the existing indoor fire alarm system using acoustic alarm could not deliver a sufficiently large alarm to the in-room personnel. To improve this, we designed and implemented a fire alarm system using edge computing and beacons. The proposed improved fire alarm system consists of terminal sensor nodes, edge nodes, a user application, and a server. The terminal sensor nodes collect indoor environment data and send it to the edge node, and the edge node monitors whether a fire occurs through the transmitted sensor value. In addition, the edge node continuously generate beacon signals to collect information of smart devices with user applications installed within the signal range, store them in a server database, and send application push-type fire alarms to all in-room personnel based on the collected user information. As a result of conducting a signal valid range measurement experiment in a university building with dense lecture rooms, it was confirmed that device information was normally collected within the beacon signal range of the edge node and a fire alarm was quickly sent to specific users. Through this, it was confirmed that the "blind spot problem of the alarm" was solved by flexibly collecting information of visitors that changes time to time and sending the alarm to a smart device very adjacent to the people. In addition, through the analysis of the experimental results, a plan to effectively apply the proposed fire alarm system according to the characteristics of the indoor space was proposed.