• 제목/요약/키워드: infinite impulse response

검색결과 65건 처리시간 0.03초

Indentification of continuous systems in the presence of input-output measurement noises

  • Yang, Zi-Jiang;Sagara, Setsuo;Wada, Kiyoshi
    • 제어로봇시스템학회:학술대회논문집
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    • 제어로봇시스템학회 1990년도 한국자동제어학술회의논문집(국제학술편); KOEX, Seoul; 26-27 Oct. 1990
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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.

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Blind MMSE Equalization of FIR/IIR Channels Using Oversampling and Multichannel Linear Prediction

  • Chen, Fangjiong;Kwong, Sam;Kok, Chi-Wah
    • ETRI Journal
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    • 제31권2호
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    • pp.162-172
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    • 2009
  • A linear-prediction-based blind equalization algorithm for single-input single-output (SISO) finite impulse response/infinite impulse response (FIR/IIR) channels is proposed. The new algorithm is based on second-order statistics, and it does not require channel order estimation. By oversampling the channel output, the SISO channel model is converted to a special single-input multiple-output (SIMO) model. Two forward linear predictors with consecutive prediction delays are applied to the subchannel outputs of the SIMO model. It is demonstrated that the partial parameters of the SIMO model can be estimated from the difference between the prediction errors when the length of the predictors is sufficiently large. The sufficient filter length for achieving the optimal prediction is also derived. Based on the estimated parameters, both batch and adaptive minimum-mean-square-error equalizers are developed. The performance of the proposed equalizers is evaluated by computer simulations and compared with existing algorithms.

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Window 함수를 이용한 FIR 디지털필터에 관한 연구 (A Study on FIR Digital filter using Window Function)

  • 구본석;배상범;김남호
    • 한국정보통신학회:학술대회논문집
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    • 한국해양정보통신학회 2004년도 춘계종합학술대회
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    • pp.68-74
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    • 2004
  • 현대산업사회의 발전에 따라 신호처리 분야에서 디지털필터의 사용이 급격히 증가하고 있으며, 이러한 디지털필터에는 일반적으로 IIR과 FIR 필터가 있다. 그리고, FIR 디지털필터는 구현이 용이하고 선형위상특성을 나타내며, 설계에 있어서 통과대역의 차단주파수 부근에서 깁스 현상에 의해 발생하는 리플을 완화하기 위해 window 함수를 사용한다. 본 논문에서는 새로운 window 함수를 설계하여 기존의 Manning, Hamming, Blackman window와 비교하였으며, 판단기준으로 peak side-lobe와 천이특성 둥을 사용하였다.

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위전도 신호처리를 위한 웨이브렌 필터와 디지털 필터의 비교 (Comparison of Digital Filters with Wavelet Multiresolution Filter for Electrogastrogram)

  • 유창용;남기창;김수찬;김덕원
    • 대한의용생체공학회:의공학회지
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    • 제23권2호
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    • pp.109-117
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    • 2002
  • 위전도(EGG electrogastrography)는 비관혈적으로 복부에 전극을 부착하여 위에서 발생하는 위 근육의 전기적인 활동성(gastric electrical activity)을 측정하는 방법이다. 위전도 신호는 주파수가 매우 낮으며(0.0083~0.15 Hz) 진폭이 매우 작기(10~100 uV) 때문에 잡음의 영향을 많이 받게 된다. 이로 인해 FIR(finite impulse response) 필터나 IIR(infinite impulse response) 필터에서는 위전도 신호와 같이 0.1417 Hz의 좁은 대역폭을 가지는 신호를 필터링하기 위해서는 높은 차수로 인해 불안정해지거나 신호가 왜곡되는 경우가 발생한다. 그래서 본 연구에서는 Daubechies 모웨이브렛을 7단계로 확장한 웨이브렛 다단계 분해 필터를 사용하였으며 기존에 많이 사용되는 2종류의 FIR 필터, 4종류의 IIR 필터들을 신호대 잡음비(SNR : signal to noise ratio)과 재생신호 자승오차(RSE : reconstruction squared error) 등의 파라미터를 이용하여 시뮬레이션 위전도 신호를 이용하여 성능을 평가하였다. 정규분포임의잡음(normal distribution random noise)이 합성된 시뮬레이션 위전도 신호를 사용한 웨이브렛 다단계 분해 필터의 SNR은 비교대상이 된 필터들 중, 최고의 SNR에 비해 잡음의 레벨에 따라 각각 9.5, 6.9, 4.7 dB 더 좋은 성능을 보여주었다. RSE에서도 $1.22{\times}10^6, 1.16{\times}10^6, 1.02{\times}10^6$이 더 적은 에러를 보여주었다.

비선형 무한 응답 최소자승형 적응 여파기의 수렴속도에 관한 연구 (On the Convergence Speed of Nonlinear Least-Squares IIR Adaptive Filter)

  • 김화종
    • 한국음향학회:학술대회논문집
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    • 한국음향학회 1987년도 학술발표회 논문집
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    • pp.58-60
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    • 1987
  • In this paper, we investigate an infinite impulse response adaptive digital filter based on the nonlinear least-squares algorithm, and compare its convergence speed to that of a self-orthogonalizing IIR ADF which is known to have fastest convergence. By simulation, it is shown that the NLS IIR ADF converges faster than other known IIR ADF's, especially for a low-order case.

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A Fast Algorithm for Real-time Adaptive Notch Filtering

  • Kim, Haeng-Gihl
    • Journal of information and communication convergence engineering
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    • 제1권4호
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    • pp.189-193
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    • 2003
  • A new algorithm is presented for adaptive notch filtering of narrow band or sine signals for embedded among broad band noise. The notch filter is implemented as a constrained infinite impulse response filter with a minimal number of parameters, Based on the recursive prediction error (RPE) method, the algorithm has the advantages of the fast convergence, accurate results and initial estimate of filter coefficient and its covariance is revealed. A convergence criterion is also developed. By using the information of the noise-to-signal power, the algorithm can self-adjust its initial filter coefficient estimate and its covariance to ensure convergence.

젖은 곱슬머리를 안정적으로 표현하기 위한 헤어 다이나믹스 (Hair Dynamics for Stable Expression of Wet and Curly Hair)

  • 안장훈;김종현
    • 한국컴퓨터정보학회:학술대회논문집
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    • 한국컴퓨터정보학회 2023년도 제67차 동계학술대회논문집 31권1호
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    • pp.389-390
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    • 2023
  • 본 논문에서는 헤어가 젖었을 때 표현되는 다양한 물리적 효과를 사실적으로 표현할 수 있는 알고리즘에 대해 제안한다. 젖은 헤어에서 나타는 곱슬 형태를 헬릭스 기반으로 모델링하여 젖었을 때 더욱더 과하게 표현되도록 하였으며, 큰 계산양 없이 효율적으로 젖은 곱슬 헤어를 효율적으로 표현하였다.

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IIR LDM 디지탈필터의 구현 (Realization of IIR LDM Digital Filters)

  • 계영철;은종관
    • 한국음향학회지
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    • 제6권3호
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    • pp.52-59
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    • 1987
  • 본 논문에서 선형 델타 변조방식을 간단한 아날로그/디지탈 변환기로 이용하여 무한 응답 디지탈 여파기를 구현하는 방법을 제시하였다. 이 방법은 하드웨어 승산기나 펄스 부호변환 아날로그/디지탈 변환기를 필요로 하지 않으므로 종래의 무한응답 디지탈 여파기의 구현방법보다 매우 간단하다. Lee와 Un의 유한응답LDMDF에 비해서 이 무한응답LDMDF는 매우 적은 계산시간이 요구된다.

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Application of Digital Signal Analysis Technique to Enhance the Quality of Tracer Gas Measurements in IAQ Model Tests

  • Lee, Hee-Kwan;Awbi, Hazim B.
    • Journal of Korean Society for Atmospheric Environment
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    • 제23권E2호
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    • pp.66-73
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    • 2007
  • The introduction of tracer gas techniques to ventilation studies in indoor environments provides valuable information that used to be unattainable from conventional testing environments. Data acquisition systems (DASs) containing analogue-to-digital (A/D) converters are usually used to function the key role that records signals to storage in digital format. In the testing process, there exist a number of components in the measuring equipment which may produce system-based inference to the monitored results. These unwanted fluctuations may cause significant error in data analysis, especially when non-linear algorithms are involved. In this study, a pre-processor is developed and applied to separate the unwanted fluctuations (noise or interference) in raw measurements and to reduce the uncertainty in the measurement. Moving average, notch filter, FIR (Finite Impulse Response) filters, and IIR (Infinite Impulse Response) filters are designed and applied to collect the desired information from the raw measurements. Tracer gas concentrations are monitored during leakage and ventilation tests in the model test room. The signal analysis functions are introduced to carry out the digital signal processing (DSP) work. Overall the FIR filters process the $CO_2$ measurement properly for ventilation rate and mean age of air calculations. It is found that, the Kaiser filter was the most applicable digital filter for pre-processing the tracer gas measurements. Although the IIR filters help to reduce the random noise in the data, they cause considerable changes to the filtered data, which is not desirable.

최대 평탄특성을 위한 2-D FIR Circular 필터 설계에 관한 연구 (A Study on Design of Maximally Flat 2-D FIR Circular Filter)

  • 서현수;배상범;김남호
    • 융합신호처리학회 학술대회논문집
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    • 한국신호처리시스템학회 2005년도 추계학술대회 논문집
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    • pp.159-162
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    • 2005
  • Recently, due to rapid developments of wireless communication and digital TV, modern society needs to process of aquisition, storage and transmission of much information. So the importance of signal processing is increasing and various digital filters are used in the two-dimensional signal such as image. And kinds of these digital filters are IIR(infinite impulse response) filter and FIR(finite impulse response) filter. And FIR filter which has the phase linearity, the easiness of creation and stability is applied to many fields. In design of this FIR filter, flatness property is a important factor in pass-band and stop-band. In this paper, we designed a 2-D Circular FIR filter using the Bernstein polynomial, it is presented flatness property in pass-band and stop-band. And we simulated the designed filter with noisy test image and compared the results with existing methods.

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