• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.524-527
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• 2005

A society has progressed rapidly toward the highly advanced digital information age. However, noise is generated by several causes, when signal is processed. Therefore, methods for eliminating those noises have researched. There were the existing FFT(fast fourier transform) and STFT(short time fourier transform) for removing noise but it's impossible to know information about time and time-frequency localization capabilities have conflictive relationship. Therefore, for overcoming these limits, wavelet-based denoising methods that are capable of multiresolution analysis are applied to the signal processing field. However, existing threshold- and correlation-based denoising methods consider only statistical characteristics for noise, accordingly a lot of noise is acceptable as an edge and are impossible to remove AWGN and impulse noise, at the same time. Hence, in this paper we proposed wavelet-based new denoising algorithm and compared existing methods with it.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.653-660
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• 2009

This paper proposes a robust audio watermarking algorithm for copyright protection and improvement of sound quality after embedding a watermark into an original sound. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and divides the spectrum into a subbands. Then, it is necessary to calculate the energy of each subband and sort n subbands in descending order corresponding to its power. After calculating the energy we choose k subbands in sorted order and find p peaks in each selected subbands, and then embed a length m watermark around the p peaks. When the listeners hear the watermarked sound, they do not recognize any distortions. Furthermore, the proposed method is robust as much as Cox's method to MP3 compression, cropping, FFT echo attacks. In addition to this, the experimental results show that the proposed method is generally 10 dB higher than Cox's method in SNR (signal-to-noise ratio) aspect.

• The Transactions of the Korean Institute of Electrical Engineers A
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• v.54 no.11
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• pp.525-532
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• 2005

This paper presents a new two-terminal numerical algorithm for fault location estimation and for faults recognition using the synchronized phaser in time-domain. The proposed algorithm is also based on the synchronized voltage and current phasor measured from the assumed PMUs(Phasor Measurement Units) installed at both ends of the transmission lines. Also the arc voltage wave shape is modeled numerically on the basis of a great number of arc voltage records obtained by transient recorder. From the calculated arc voltage amplitude it can make a decision whether the fault is permanent or transient. In this paper the algorithm is given and estimated using DFT(discrete Fourier Transform) and the LES(Least Error Squares Method). The algorithm uses a very short data window and enables fast fault detection and classification for real-time transmission line protection. To test the validity of the proposed algorithm, the Electro-Magnetic Transient Program(EMTP/ATP) is used.

• IEMEK Journal of Embedded Systems and Applications
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• v.12 no.2
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• pp.71-77
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• 2017

This paper proposes a low complexity frequency modulated continuous wave (FMCW) surveillance radar algorithm. In the conventional surveillance radar systems, the two dimensional (2D) fast Fourier transform (FFT) method is usually employed in order to detect the distance and velocity of the targets. However, in a surveillance radar systems, it is more important to immediately detect the presence or absence of the targets, rather than accurately detecting the distance or speed information of the target. In the proposed algorithm, in order to immediately detect the presence or absence of targets, 1D FFT is performed on the first and M-th bit signals among a total of M beat signals and then a phase change between two FFT outputs is observed. The range of target is estimated only when the phase change occurs. By doing so, the proposed algorithm achieves a significantly lower complexity compared to the conventional surveillance scheme using 2D FFT. In addition, show in order to verify the performance of the proposed algorithm, the simulation and the experiment results are performed using 24GHz FMCW radar module.

• Proceedings of the KIEE Conference
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• 2005.11b
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• pp.283-285
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• 2005

This paper presents a new two-terminal numerical algorithm for fault location estimation and for faults recognition using the synchronized phasor in time-domain. The proposed algorithm is also based on the synchronized voltage and current phasor measured from the PMUs(Phasor Measurement Units) installed at both ends of the transmission lines. Also the arc voltage wave shape is modeled numerically on the basis of a great number of arc voltage records obtained by transient recorder. From the calculated arc voltage amplitude it can make a decision whether the fault is permanent or transient. In this paper the algorithm is given and estimated using DFT(Discrete Fourier Transform) and the LES(Least Error Squares Method). The algorithm uses a very short data window and enables fast fault detection and classification for real-time transmission line protection. To test the validity of the proposed algorithm, the Electro-Magnetic Transient Program(EMTP/ATP) and MATLAB is used.

• Proceedings of the KIEE Conference
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• 2015.07a
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• pp.579-580
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• 2015

Power frequency is one of important operational parameters evaluating reliability, stability, and measuring efficiency of power. Therefore, an accurate and fast estimate of the power frequency is required. The magnitude gains of cosine and sine filters become different when the power frequency is deviated from the nominal value. The proposed algorithm estimates the power frequency based on this. To demonstrate the performance of the proposed algorithm, RTDS and DSP are used. The simulation results show that the algorithm has not only a high level of robustness but also high measurement accuracy over a wide range of frequency changes. In addition, the algorithm was immune to harmonics and noise.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.

• Journal of Power Electronics
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• v.10 no.4
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• pp.412-418
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• 2010

This paper proposes a new algorithm to generate a reference signal for an active power filter using a sliding-window FFT operation to improve the steady-state performance of the active power filter. In the proposed algorithm the sliding-window FFT operation is applied to the load current to generate the reference value for the compensating current. The magnitude and phase-angle for each order of harmonics are respectively averaged for 14 periods. Furthermore, the phase-angle delay for each order of harmonics passing through the controller is corrected in advance to improve the compensation performance. The steady-state and transient performance of the proposed algorithm was verified through computer simulations and experimental work with a hardware prototype. A single-phase active power filter with the proposed algorithm can offer a reduction in THD from 75% to 4% when it is applied to a non-linear load composed of a diode bridge and a RC circuit. The active power filter with the proposed reference generation method shows accurate harmonic compensation performance compared with previously developed methods, in which the THD of source current is higher than 5%.

• ETRI Journal
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• v.36 no.1
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• pp.1-11
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• 2014

In this paper, we propose a distributed synchronization algorithm for wireless mesh networks based on orthogonal frequency division multiple access. For time and frequency synchronization, a node requests its neighbor nodes for a change of fast Fourier transform starting points, transmission times, and carrier frequencies needed for synchronization. The node also updates its own time and frequency elements through simple formulas based on request messages received from neighbor nodes using a guard interval and a cyclic prefix. This process with the cooperation of neighbor nodes leads to a gradual synchronization of all nodes in the network. Through a performance comparison with a conventional scheme, we obtain simulation results indicating that the proposed scheme outperforms the conventional scheme in random topologies and a grid topology.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04c
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• pp.328-330
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• 2003

홍채 인식 시스템의 성능향상을 위해서는 전처리 단계에서 위조된 데이터나 잡음이 섞인 데이터를 걸러내는 과정이 매우 중요하다. 이 논문에서는, 강인한 홍채 인식 시스템을 위해서 두 단계로 이루어진 눈 영상 검사 알고리즘을 제안한다. 알고리즘의 첫 번째 단계에서는 동공 반지름과 눈꺼풀 움직임 변화량의 상관계수(coefficient)와 2차원 고속 퓨리에 변환(Fast Fourier Transform) 스펙트럼을 이용해 위조된 데이터를 찾아낸다. 두 번째 단계에서는 눈 영상에 눈 깜박임이나 속눈썹의 홍채영역 침범, 홍채 영상이 부분적으로 잘렸을 경우 등의 시스템의 성능저하의 원인이 되는 부적절한 데이터를 찾아낸다. 111영의 1734개의 눈 영상을 대상으로 실험한 결과, 제안된 알고리즘의 성공률은 96.5%였고 이러한 전처리 과정을 통해서 전체적인 시스템의 정확률을 향상시킬 수 있다.