• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제14권1호
• /
• pp.53-76
• /
• 2020

The synchronous phasor measurement algorithm is the core content of the phasor measurement unit. This manuscript proposes a dynamic synchronous phasor measurement algorithm based on compressed sensing theory. First, a dynamic signal model based on the Taylor series was established. The dynamic power signal was preprocessed using a least mean square error adaptive filter to eliminate interference from noise and harmonic components. A Chirplet overcomplete dictionary was then designed to realize a sparse representation. A reduction of the signal dimension was next achieved using a Gaussian observation matrix. Finally, the improved orthogonal matching pursuit algorithm was used to realize the sparse decomposition of the signal to be detected, the amplitude and phase of the original power signal were estimated according to the best matching atomic parameters, and the total vector error index was used for an error evaluation. Chroma 61511 was used for the output of various signals, the simulation results of which show that the proposed algorithm cannot only effectively filter out interference signals, it also achieves a better dynamic response performance and stability compared with a traditional DFT algorithm and the improved DFT synchronous phasor measurement algorithm, and the phasor measurement accuracy of the signal is greatly improved. In practical applications, the hardware costs of the system can be further reduced.

• 전기학회논문지P
• /
• 제59권2호
• /
• pp.173-178
• /
• 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.

• 전자공학회논문지B
• /
• 제33B권3호
• /
• pp.1-8
• /
• 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.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2003년도 하계학술대회 논문집 A
• /
• pp.345-347
• /
• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제6권9호
• /
• pp.2218-2230
• /
• 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.

• 융합신호처리학회논문지
• /
• 제10권2호
• /
• pp.106-111
• /
• 2009

실시간 영상 안정화 기술은 손떨림에 의한 휴대용 카메라 혹은 외부적 조건에 의한 고정 카메라의 흔들림 보상에 이용된다. 본 논문은 비교적 큰 외부적 요인으로 인하여 발생하는 동영상의 흔들림에 대한 대책에 관한 것이다. 동영상 안정화 파라메터로 이용되는 기준 프레임에 대한 현재 프레임의 변위를 얻기 위하여 DFT에 기반한 위상 상관법을 이용한다. 그리고 위상 상관지도에서의 효율적이고 안정적인 탐색을 위하여 칼만 필터를 이용하여 탐색 범위를 추정하는 방법과 안정적인 성능과 실시간 처리에 필요한 조건을 실험적으로 찾아내고 그 조건을 제시한다. 중심 지점에 대한 평균밝기의 표준편차 값을 동영상 안정화의 성능 평가 척도로 제안하고 가상 흔들림 동영상과 실제 흔들림 동영상에 대하여 성능을 서로 비교하였다.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2000년도 ITC-CSCC -1
• /
• pp.225-228
• /
• 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.

• 대한전기학회논문지
• /
• 제43권4호
• /
• pp.553-563
• /
• 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.

• 융합신호처리학회논문지
• /
• 제9권4호
• /
• pp.295-303
• /
• 2008

변환영역 적응필터는 시간영역 적응필터보다 일반직으로 수렴속도가 빠르지만 필터의 차수가 증가함에 따라 계산량이 크게 증가한다. 이러한 문제점은 변환영역 적응필터를 서브밴드 분해구조로 변경함으로써 해결할 수 있다. 이 논문에서는 일반화된 서브밴드 분해 FIR 적응 필터의 수렴속도 향상을 위해 웨이블릿 변환영역에서 다이아딕 희소인자 서브필터를 가지는 일반화된 서브밴드 분해 FTR 적응 필터의 구조를 설계하였다. 그리고 이 적응필터에서 변환영역의 일반화된 등가입력을 유도하고 이 입력을 이용하여 LMS 일고리듬에 대한 수렴특성을 해석 및 평가하였다. 이 서브밴드 FIR 적응필터를 이용하여 역 모델링 시스템과 주기성 잡음제거기를 구성하고 LMS 알고리듬 대한 이 시스템들의 수련속도를 이산푸리에 변환을 이용한 서브밴드 적응필터의 것과 컴퓨터 모의실험으로 비교하였다.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2005년도 제36회 하계학술대회 논문집 A
• /
• pp.211-213
• /
• 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.