• Title/Summary/Keyword: DFT Filter

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Advanced Frequency Estimation Technique using Gain Compensation (이득 보상에 의한 개선된 주파수 추정 알고리즘)

  • Park, Chul-Won
    • The Transactions of the Korean Institute of Electrical Engineers P
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    • v.59 no.2
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    • pp.173-178
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    • 2010
  • Frequency is an important operating parameter of a power system. Due to the sudden change in generation and loads or faults in power system, the frequency is supposed to deviate from its nominal value. It is essential that the frequency of a power system be maintained very close to its nominal frequency. And monitoring and an accurate estimation of the power frequency by timing synchronized signal provided by FDR is essential to optimum operation and prevention for wide area blackout. As most conventional frequency estimation schemes are based on DFT filter, it has been pointed out that the gain error by change in magnitude could cause the defects when the power frequency is deviated from nominal value. In this paper, an advanced frequency estimation scheme using gain compensation for fault disturbance recorders (FDR) is presented. The proposed scheme can reduce the gain error caused when the power frequency is deviated from nominal value. Various simulation using both the data from EMTP package and user's defined arbitrary signals are performed to demonstrate the effectiveness of the proposed scheme. The simulation results show that the proposed scheme can provide better accuracy and higher robustness to harmonics and noise under both steady state tests and dynamic conditions.

A design of adaptive equalizer using the transversal walsh filter and the optimal LMS algorithm (횡단형 월쉬필터와 최적 LMS 기법을 이용한 적응 등화기의 설계)

  • 김종부
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.33B no.3
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    • pp.1-8
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    • 1996
  • This paper proposes a novel transversal filter and an optimal LMS algorithm, and show how these can be realized as an adaptive equalizer. The transversal filter consists of a walsh and block pulse functions. in the LMS algorithm with equalizers, the convergence factor is an improtant design parameter because it governs stability and convergence speed. The conventional adaptation techniques use a fixed time constant convergence factor by the trial and error method. In this paper, an optimal method in the choice of the convergence factor is proposed. The proposed algorithm is obtrained that is tailored for each filter tap and is updated at each iteration. The performance of the proposed algorithm is compared iwth those of the conventional TDL and DFT equalizers by computer simulations.

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VHDL Design of High Performance FIR Filter for Digital Protection Relay Using Least Square Algorithm (최소자승 알고리즘을 이용한 디지털 보호 계전기용 고성능 FIR 필터의 VHDL 모델 설계)

  • Shin, Jae-Shin;Kim, Jong-Tae;Park, Jong-Kang;Seo, Jong-Wan;Shin, Myung-Cheol
    • Proceedings of the KIEE Conference
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    • 2003.07a
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    • pp.345-347
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    • 2003
  • 본 논문에서는 디지털 보호 계전기에 쓰이는 필터 중에서 최소 자승 알고리즘을 이용한 고성능 FIR 필터를 설계하였다. 기존의 DFT필터와 MATLAB 시뮬레이션을 이용하여 비교하였으며 FIR 필터의 VHDL모델 및 합성에 중점을 두었다. FIR 필터는 기본적으로 유한개의 임펄스 응답이 이루어지기 때문에 기타 다른 필터에 비하여 안정도가 높으며 선형적인 위상을 가지기 때문에 차단 주파수 대역의 왜곡현상을 없앨 수 있는 장점을 가지고 있다. 여러 가지 알고리즘으로 구현한 FIR 필터를 시뮬레이션 한 결과 최소 자승 알고리즘이 가장 우수한 결과를 나타내었다. 기본적으로 디지털 보호 계전기에서 디지털 필터의 기능은 사고 전압, 전류로부터 60Hz의 기본파 추출 CT, PT 왜곡 및 DC offset을 제거하는데 있다. 본 논문에서는 이러한 기능을 가지면서 샘플링 주파수와 차수를 같게 하여 FIR 필터와 DFT 필터의 주파수 응답과 연 산 속도를 비교 하였다. 본 논문에서 설계된 최소 자승 알고리즘을 이용한 FIR 필터는 같은 조건의 DFT필터에 비해 1고조파와 2고조파의 차이가 10db 이상 더 우수 하였으며 연산 속도 또한 2배 이상 좋은 결과를 보였다.

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Efficient Channel Delay Estimation for OFDM Systems over Doubly-Selective Fading Channels

  • Heo, Seo Weon;Lim, Jongtae
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.6 no.9
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    • pp.2218-2230
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    • 2012
  • In this paper, we propose an efficient channel delay estimation method for orthogonal frequency-division multiplexing (OFDM) systems, especially over doubly-selective fading channels which are selective in both the symbol time domain and subcarrier frequency domain. For the doubly-selective fading channels in single frequency network (SFN), long and strong echoes exist and thus the conventional discrete Fourier Transform (DFT) based channel delay estimation system often fails to produce the exact channel delay profile. Based on the analysis of the discrete-time frequency response of the channel impulse response (CIR) coefficients in the DFT-based channel delay estimation system, we develop a method to effectively extract the true CIR from the aliased signals by employing a simple narrow-band low-pass filter (NB-LPF). The performance of the proposed system is verified using the COST207 TU6 SFN channel model.

Video Stabilization using Phase Correlation and Kalman Filter-Based Motion Prediction (위상상관과 칼만 필터 움직임 예측을 이용한 동영상 안정화)

  • Han, Hag-Yong;Jeong, Hyo-Won;Kang, Bong-Soon;Hur, Kang-In
    • Journal of the Institute of Convergence Signal Processing
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    • v.10 no.2
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    • pp.106-111
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    • 2009
  • Real-time video stabilization technology is used in correction for the camera vibrations of the hand-held camera by hand or fixed camera by external condition. This paper is about the counterplan to cope with the vibration of the movie generated by the large external cause relatively. we use the movie stabilization parameters with the phase correlation method based the DFT to get the displacements of the current frame to the reference frame. we use the kalman filter for the efficient and stable searching works on the phase correlation map and present the proper conditions for the real-time processing through the experiments. We propose the measure to evaluate the capability of the video stabilizer which is the standard deviation of the brightness of the center block. and compare the capability for the video sequences randomly shifted and the jittered video sequences obtained from camera.

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Noise Cancellation System Based on Frequency Domain Adaptive Filter Using Modified DFT Pair

  • Nakanishi, Isao;Nakamura, Youichi;Itoh, Yoshio;Fukui, Yutaka
    • Proceedings of the IEEK Conference
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    • 2000.07a
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    • pp.225-228
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    • 2000
  • It is well known that a Frequency Domain Adaptive Filter (FDAF) converges faster than a Time Domain Adaptive Filter (TDAF) even when the input signal is colored such as a speech signal. We have proposed the FDAF using the Modified Discrete Fourier Transform Pair (MDFTP) and its realization and effectiveness has been confirmed through the computer simulations. In this paper, we apply the FDAF using the MDFTP to the noise cancellation system. The proposed system is based on the Adaptive Line Enhancer (ALE) and utilizes single microphone; therefore it is suitable for the portable electronic equipment. Moreover, we propose to utilize the MDFT for detecting of the pitch in the speech because the number of data points in the MDFT must be equal to the pitch to confirmed that the noise can be removed to near the level of SNR.

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A Study on the Extraction of Fundamental Frequency Components in the Transient Wave Signals Using Artificial neural networks (신경회로망을 이용한 과도파형의 기본파성분 추출에 관한 연구)

  • 신명철;이복구
    • The Transactions of the Korean Institute of Electrical Engineers
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    • v.43 no.4
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    • pp.553-563
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    • 1994
  • This paper presents a filtering method using neural networks to extract fundamental frequency components of the transient wave signals in power systems. Based on the ability of multilayer feedforward neural networks to approximate any continuous function, a neural networks mapping filter is proposed for the protective distance relaying systems to extract the effective components efficiently. A characteristic feature of this mapping filter is composed of the multilayer perceptron neural networks which are trained by using random signals and those are mapped to the DFT filtering computational structure by GDR(Generalized Delta Rule). The advantage of this approach is demonstrated by the random waves and the fault transient wave signals of EMTP(electromagnetic transients program) in power systems fault conditions. The proposed method is compared with the conventional method and the simulation results show the efficiency of the neural networks.

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The Structure and the Convergence Characteristics Analysis on the Generalized Subband Decomposition FIR Adaptive Filter in Wavelet Transform Domain (웨이블릿 변환을 이용한 일반화된 서브밴드 분해 FIR 적응 필터의 구조와 수렴특성 해석)

  • Park, Sun-Kyu;Park, Nam-Chun
    • Journal of the Institute of Convergence Signal Processing
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    • v.9 no.4
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    • pp.295-303
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    • 2008
  • In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

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The Study of Fast Frequency Measurement Technique for Protection Relay (계전기를 위한 고속의 주교수 계측 알고리즘)

  • Kim, Beung-Jin;Lee, Chang-Ho;Park, Jong-Chan
    • Proceedings of the KIEE Conference
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    • 2005.07a
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    • pp.211-213
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    • 2005
  • In this paper, the frequency compensator for digital protective relay is introduced. Due to sudden appearance of generation-load or fault in power system, the frequency can deviate from its nominal value. The Orthogonal filter adopted to digital protective relay is exposed to the problem. Therefore, the frequency variation makes a difficult to protective function and power measurement in digital protective relay. The essential property of the proposed algorithm presented is its outstanding immunity to both signal orthogonal component magnitudes and DFT filter gain variations which ensures below 0.05Hz accuracy of estimation. Additionally, the proposed method has excellent characteristics with low signal sampling rate. Based on the frequency estimation, author design under frequency protection relay for generator and verifies its performance with several experimental tests.

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Transmission waveform design for compressive sensing active sonar using the matrix projection from Gram matrix to identity matrix and a constraint for bandwidth (대역폭 제한 조건과 Gram 행렬의 단위행렬로의 사영을 이용한 압축센싱 능동소나 송신파형 설계)

  • Lee, Sehyun;Lee, Keunhwa;Lim, Jun-Seok;Cheong, Myoung-Jun
    • The Journal of the Acoustical Society of Korea
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    • v.38 no.5
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    • pp.522-533
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    • 2019
  • The compressive sensing model for range-Doppler estimation can be expressed as an under-determined linear system y = Ax. To find the solution of the linear system with the compressive sensing method, matrix A should be sufficiently incoherent and x to be sparse. In this paper, we propose a transmission waveform design method that maintains the bandwidth required by the sonar system while lowering the mutual coherence of the matrix A so that the matrix A is incoherent. The proposed method combines two methods of optimizing the sensing matrix with the alternating projection and suppressing unwanted frequency bands using the DFT (Discrete Fourier Transform) matrix. We compare range-Doppler estimation performance of existing waveform LFM(Linear Frequency Modulated) and designed waveform using the matched filter and the compressive sensing method. Simulation shows that the designed transmission waveform has better detection performance than the existing waveform LFM.