• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2078-2080
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• 2001

본 논문은 기존의 가중 메디안 필터에 음의 가중치를 갖는 recursive weighted median(RWM) 필터 구조를 소개하고자 한다. 이는 연산과정에 퍼지 클러스터를 적용하여 또 다른 구조의 RWM 필터를 제안하였다. 제안된 RWM 필터는 infinte impulse response(IIR) 선형 필터의 부류로 상사되는 것으로서 기존의 선형 IIR 필터와는 달리 강건한 대역 또는 고역통과 필터의 특성 뿐만 아니라 피드백 되어진 계수 값에 상관없이 항상 BIBO 시스템에서의 안정성과 잡음에 강건한 특성을 나타낸다. 특히 퍼지 클러스터를 적용하여 중앙값 주변의 값을 적절히 취함으로서 신뢰성과 보다 빠른 연산속도의 성능을 여러 다른 구조의 필터들과 성능 비교 실험을 통해 입증하였다.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1758-1763
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• 2013

This paper is a study on the performance of the stable cascading BPF. There are generally two methods of designing IIR filter, which are a direct method and an indirect one. The indirect design method that is transformed into the BPF by frequency transformation using the prototype analog LPF which is satisfied for designing specifications is applied to this study. As typical prototype analog LPFs, there are the Butterworth filter, the Chebyshev filter and the elliptic filter. In this study, we connect the frequency transformed BPFs (to the cascade form) which have been converted from the stable Butterworth filter and Chebyshev filter. Three classified simulations are conducted in this study, which are the cascading Butterworth BPF Only, the cascading Chebyshev BPF Only and the cascading Butterworth and Chebyshev BPFs. As a result of the simulation, this study shows that a ripple size of the cascading Chebyshev BPF Only is about 1[dB] smaller than that of the cascading Butterworth and Chebyshev BPFs and also the skirt characteristic of the cascading Chebyshev BPF in the transition band is most outstanding and has the widest bandwidth. The result of performance comparison shows the validity of specifications required in the workplace.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.5
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• pp.42-49
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• 1994

In this paper we consider efficient 1-D and 2-D linear-phased half-band filter designs. We first introduce a new derivation of the existing Vaidynathan-Nguyen 's half-band filter design method, which verifies that the design provides optimal half-band filters. We then propose an approximately-linear-phased IIR half-band filter design method, which is based on the all-pass equalizer design with the linear phase -$\omega$/2. Finally, we propose an efficient method to design optimal 2-D half-band filters, for which we utilize a 2-D all pass prototype filter of half the order of the desired 2-D half-band filters.

• 제어로봇시스템학회:학술대회논문집
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• 1990.10b
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• pp.1222-1227
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• 1990

The problem of identification of continuous systems is considered when both the discrete input and output measurements are contaminated by white noises. Using a predesigned digital low-pass filter, a discrete-time estimation model is constructed easily without direct approximations of system signal derivatives from sampled data. If the pass-band of the filter is designed so that it includes the main frequencies of both the system input and output signals in some range, the noise effects are sufficiently reduced, accurate estimates can be obtained by least squares(LS) algorithm in the presence of low measurement noises. Two classes of filters(infinite impulse response(IIR) filter and finite impulse response(FIR) filter) are employed. The former requires less computational burden and memory than the latter while the latter is suitable for the bias compensated least squares(BCLS) method, which compensates the bias of the LS estimate by the estimates of the input-output noise variances and thus yields unbiased estimates in the presence of high noises.

• Proceedings of the KOSOMBE Conference
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• v.1998 no.11
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• pp.68-69
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• 1998

Discriminating the clutter signal from Doppler signal is the main function of CDP(color Doppler processor). Up to now, a general method of eliminating clutter signal is using IIR high pass filter. There are many filters that were introduced in other paper. In this paper, we propose the new method of filtering clutter signal. To the new method, we adopt an appropriate filter that can eliminate clutter filter most effectively.

• Journal of Navigation and Port Research
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• v.28 no.2
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• pp.135-140
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• 2004

A purpose of developing a sound source tracking system in this paper is to reduce the noise efficiently from the received signal by microphone array and measure the signal's time delay between the microphones. I have applied the wavelet analysis algorithm to the system and calculated the sound source's relative position For the performance evaluation, I have compared with the results of utilizing the digital filtering methods based on the FIR LPF using Kaiser window function and the inverse Chebyshev IIR LPF. As a result, I have confirmed the fact that 'time-scale' filter using inverse discrete wavelet transform was suitable for this system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.68-74
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• 2004

The use of digital filters in the signal processing field is increasing rapidly with development of the modem industrial society, and generally digital filter is classified into FIR (infinite impulse response) fillers and FIR (finite impulse response) filters. The FIR digital filter has the phase linearity and the easiness of creation. In the design of the FIR digital filter, the window function is used to alleviate ripples caused by Gibbs Phenomenon around the cut-off frequency of the passband. In this paper, we designed a new window function and compared with existing Manning, Hamming and Blackman window functions. And we used peak side-lobe and transient characteristics as standard of judgement.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.377-380
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• 2002

When it calculates the optimum price of filter coefficient, the many operation quantity is necessary. Is like that the real-time control is difficult and the hardware embodiment expense is big. The case which does not know advance information of input signal or the case where the statistical nature changes with change of surroundings environment is necessary the adaptive filter. Every hour to change a coefficient automatically and system in order to reach to the condition of optimum oneself, the fact that is the adaptive filter. When it does not the quality of input signal or it does not know the environment of surroundings every hour changing, it does not emit not to be, in order to collect, the fact that is the adaptive filter. The case of the Acoustic Echo Canceler does thousands filter coefficients in necessity. It reduces a many calculation quantity to respect, it uses the IIR filter from hour territory. Also it uses the block adaptive filter which has a block input signal and a block output signal. The former there is a weak point where the stability discrimination is always demanded. Consequently, The block adaptive filter is researched plentifully. This dissertation planned the block MADF adaptive filter used to MADf algorithm.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.66 no.4
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• pp.200-205
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• 2017

In this paper, we propose a technique to determine the normal and arc of an alternating current using a mixed filter composed of an average filter and a band-pass filter and a frequency analysis. The proposed method uses the moving average filter of the FIR filter structure for noise removal and the band-pass filter of the IIR filter structure for detecting only specific frequency components after normalizing the measured current signal based on the maximum value. After performing Fast Fourier Transform (FFT) using the band-pass filtered signal, the total energy is calculated using the magnitude component of the frequency, and the arc is detected using the magnitude of the calculated energy. In order to show the validity of the proposed method, we experimented with various data and found that arc and steady state can be easily discriminated by calculating spectral energy. Therefore, it is considered that the proposed method can be applied to arc diagnosis of low voltage electric wire.

• International Journal of Aeronautical and Space Sciences
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• v.13 no.4
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• pp.491-498
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• 2012

A Ground Based Augmentation System (GBAS) is an enabling technology for an aircraft's precision approach based on a Global Navigation Satellite System (GNSS). However, GBAS is vulnerable to interference, so effective GNSS interference detection and mitigation methods need to be employed. In this paper, an intentional GNSS interference detection and characterization algorithm is proposed. The algorithm uses Automatic Gain Control (AGC) gain and adaptive notch filter parameters to classify types of incoming interference and to characterize them. The AGC gain and adaptive lattice IIR notch filter parameter values in GNSS receivers are examined according to interference types and power levels. Based on those data, the interference detection and characterization algorithm is developed and Monte Carlo simulations are carried out for performance analysis of the proposed method. Here, the proposed algorithm is used to detect and characterize single-tone continuous wave interference, swept continuous wave interference, and band-limited white Gaussian noise. The algorithm can be used for GNSS interference monitoring in an excessive Radio Frequency Interference environment which causes loss of receiver tracking. This interference detection and characterization algorithm will be used to enhance the interference mitigation algorithm.