• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 1998.04a
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• pp.156-162
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• 1998

The whistle is a very important information source for the safety navigation under foggy weather. But navigator has no concern about this , because it must be achieved by human hearing sense and considered as a vague signal. If the range and relative bearing of signa source can be detected automatically, it would be very useful system for preventing marine casualties making a lot of economic loss and environment pollution. Before the algorithm of 2-dimensinal sound source tracking system was reported . This paper describes the method that can obtain thetime lag between three signas and the theory of cross-correlation analysis and subtraction method for cauculating the time lag by using the digital signal data sequences. And a series of experiments were carried out for various position of sound source in the range from 200m to 530cm.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.2
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• pp.77-82
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• 2011

In this paper, we investigate source separation algorithms for stereo audio mixed using amplitude panning method. This source separation algorithms can be used in various applications such as up-mixing, speech enhancement, and high quality sound source separation. The methods in this paper estimate the panning angles of individual signals using the principal component analysis being applied in time-frequency tiles of the input signal and independently extract each signal through directional filtering. Performances of the methods were evaluated through computer simulations.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.5
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• pp.807-812
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• 2020

This paper proposes a blind speech source separation algorithm using a microphone to separate only the target speech source signal in an environment in which various speech source signals are mixed. The proposed algorithm is a model of frequency domain representation based on independent component analysis method. Accordingly, for the purpose of verifying the validity of independent component analysis in the frequency domain for two speech sources, the proposed algorithm is executed by changing the type of speech sources to perform speech sources separation to verify the improvement effect. It was clarified from the experimental results by the waveform of this experiment that the two-channel speech source signals can be clearly separated compared to the original waveform. In addition, in this experiments, the proposed algorithm improves the speech source separation performance compared to the existing algorithms, from the experimental results using the target signal to interference energy ratio.

• International Journal of Naval Architecture and Ocean Engineering
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• v.7 no.1
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• pp.128-141
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• 2015

Independent Component Analysis (ICA), one of the blind source separation methods, can be applied for extracting unknown source signals only from received signals. This is accomplished by finding statistical independence of signal mixtures and has been successfully applied to myriad fields such as medical science, image processing, and numerous others. Nevertheless, there are inherent problems that have been reported when using this technique: instability and invalid ordering of separated signals, particularly when using a conventional ICA technique in vibratory source signal identification of complex structures. In this study, a simple iterative algorithm of the conventional ICA has been proposed to mitigate these problems. The proposed method to extract more stable source signals having valid order includes an iterative and reordering process of extracted mixing matrix to reconstruct finally converged source signals, referring to the magnitudes of correlation coefficients between the intermediately separated signals and the signals measured on or nearby sources. In order to review the problems of the conventional ICA technique and to validate the proposed method, numerical analyses have been carried out for a virtual response model and a 30 m class submarine model. Moreover, in order to investigate applicability of the proposed method to real problem of complex structure, an experiment has been carried out for a scaled submarine mockup. The results show that the proposed method could resolve the inherent problems of a conventional ICA technique.

• Journal of Biomedical Engineering Research
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• v.31 no.6
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• pp.456-463
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• 2010

In the present study, we proposed a new subspace scanning algorithm to enhance the spatial resolution of electroencephalography (EEG) and magnetoencephalography(MEG) source localization. Subspace scanning algorithms, represented by the multiple signal classification (MUSIC) algorithm and the first principal vector (FINE) algorithm, have been widely used to localize asynchronous multiple dipolar sources in human cerebral cortex. The conventional MUSIC algorithm used principal component analysis (PCA) to extract the noise vector subspace, thereby having difficulty in discriminating two or more closely-spaced cortical sources. The FINE algorithm addressed the problem by using only a part of the noise vector subspace, but there was no golden rule to determine the number of noise vectors. In the present work, we estimated a non-orthogonal signal vector set using independent component analysis (ICA) instead of using PCA and performed the source scanning process in the signal vector subspace, not in the noise vector subspace. Realistic 2D and 3D computer simulations, which compared the spatial resolutions of various algorithms under different noise levels, showed that the proposed ICA-MUSIC algorithm has the highest spatial resolution, suggesting that it can be a useful tool for practical EEG/MEG source localization.

• ETRI Journal
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• v.36 no.5
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• pp.772-782
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• 2014

Two-microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time-frequency (T-F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T-F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T-F units with hit rates above 85%. It outperforms previous solutions in terms of signal-to-noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.13 no.2
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• pp.107-114
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• 2002

The problem of blind signal separation of independent sources consist in retrieving the source from the observation of unknown mixtures of unknown sources. In this paper, we propose a technique for blind signal separation that can extract original signals from their non-stationary mixtures observed in a ordinary room. The proposed method implements blind signal separation by minimizing a non-negative cost function that achieves the minimum when the second-order cross-correlation value of the observed signals becomes zero. The validity of the proposed method has been verified by a computer simulation and experiment that extracts two source signals from their mixtures observed in a normal room.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.30 no.4
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• pp.290-299
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• 2019

Direction-finding equipment with multiple antennas are used to estimate the direction of a signal emitted by a source; they can be used to rescue a victim or locate a specified source. During direction finding, reflection waves are present and signal distortion is observed depending on the external shape and material of a system that incorporates the direction-finding equipment and multiple antennas. Therefore, to accurately estimate the azimuth of the signal source and develop the direction-finding equipment, a calibration should be performed to reflect the influence of the antenna arrangement(layout) and system contour. In this paper, we describe an in-flight calibration method to develop direction-finding equipment to locate communication signals using an aviation system, and we analyze the direction-finding performance when applying phase calibration data obtained through the in-flight calibration.

• Journal of the Korean Institute of Navigation
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• v.24 no.5
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• pp.443-450
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• 2000

Before some experiments were carried out with analog bandpass filter which used for filtering the noise included in sound source signal. And this filter was constituted by condenser, register and operational amplifier. Hut these elements made the phase characteristics to differentiate in each sensing channel and cause a little of measurement error. We made new measurement system that was substituted digital filter for the analog filter in order to develop the optimal system which could find the time delay between each sensors with high accuracy. This paper describes the new system's constitution and the function of each parts. Specially three digital filters were designed and applied to the digital signal processing Part. And a series of experiments were carried out with the source＇s distance 9.53meters and the random bearing interval within the limits of $0^{\circ}$ ~ $180^{\circ}$. As a result, we have recognized that the accuracy of measurements were differentiated by the methods what kind of digital filter were adopted. And we have confirmed the facts that IIR LPF was suitable for sound source's bearing measurement and FIR LPF reduced the range measurement error.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.21 no.2
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• pp.129-136
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• 2011

The aim of this paper is to present the method of identifying the impact location on the plate. This basic research has the future purpose to achieve the human-interaction technology based on the signal processing, piezoelectric materials, and wave propagation. The present work concerning the location identification of a single impact on the plate simulated the waveform numerically generated by impact force and applied the SWFOM(sliced Wigner higher fourth order moment) to the waveform to get the arrival time differences due to impact force between three sensors attached to the plate. The simulated signal is useful to get the information for time interval for the only direct wave. This information is used the source localization by using experimental work. The measured signal is also used for source localization of a single impact based on the higher order time frequency as a novel work.