• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1996년도 한국자동제어학술회의논문집(국내학술편); 포항공과대학교, 포항; 24-26 Oct. 1996
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• pp.445-448
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• 1996

For a lightly damped system, IIR based filter can have better performance than FIR filter as an adaptive filter in ANC algorithm. IIR based filter which has an infinite impulse response can model a lightly damped acoustic system with a small number of weights compared that of FIR filter and has no nonlinearity and instability problems on weight updating process which are associated to the conventional IIR filters. There are, however, some drawbacks such as design parameters to be determined earlier to get better performance and much increased computational power especially in the presence of error path. In this study, base filter parameters are determined in systematic manner, with the knowledge of the nominal impulse response of the system which should be identified, by Prony's method. Three methods reducing the computational load are proposed and their performance and application limits are discussed. Simulation and experimental results demonstrate the feasibility of the proposed method.

• 제어로봇시스템학회논문지
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• 제16권12호
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• pp.1220-1225
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• 2010

This paper proposes a novel active noise control (ANC) approach that uses an IIR filter and neural network techniques to effectively reduce interior noise. We construct a multiple-channel IIR filter module which is a linearly augmented framework with a generic IIR model to generate a primary control signal. A three-layer perceptron neural network is employed for establishing a secondary-path model to represent air channels among noise fields. Since the IIR module and neural network are connected in series, the output of an IIR filter is transferred forward to the neural model to generate a final ANC signal. A gradient descent optimization based learning algorithm is analytically derived for the optimal selection of the ANC parameter vectors. Moreover, re-estimation of partial parameter vectors in the ANC system is proposed for online learning. Lastly, we present the results of a numerical study to test our ANC methodology with realistic interior noise measurement obtained from Korean railway trains.

• ETRI Journal
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• 제25권5호
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• pp.345-355
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• 2003

In this paper, we propose a method for designing a class of M-channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop-band attenuation, which is also taken care of in the proposed algorithm.

• 한국통신학회논문지
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• 제31권9C호
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• pp.876-881
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• 2006

In this paper, we propose a complex-valued IIR filter for digital VSB signals based on CMA in order to efficiently mitigate multipath distortions, especially the leakage from the quadrature component. The proposed equalizer overcomes the drawback of the conventional real-valued IIR equalizers that it attempts to equalize Hilbert transform of quadrature component. We demonstrate via simulation that the proposed complex IIR filter successfully mitigates the leakages from the quadrature component, while the conventional real IIR filter requires a longer IIR filter to achieve the same performance. We present cost function analysis for a simple two-tap case showing that the proposed IIR equalizer with CMA for VSB signals has a global minimum at the desired location.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제2권2호
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• pp.115-121
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• 2002

This paper presents an evolutionary design of digital IIR filters using the genetic algorithm (GA) with modified genetic operators and real-valued encoding. Conventional digital IIR filter design methods involve algebraic transformations of the transfer function of an analog low-pass filter (LPF) that satisfies prescribed filter specifications. Other types of frequency-selective digital fillers as high-pass (HPF), band-pass (BPF), and band-stop (BSF) filters are obtained by appropriate transformations of a prototype low-pass filter. In the GA-based digital IIR filter design scheme, filter coefficients are represented as a set of real-valued genes in a chromosome. Each chromosome represents the structure and weights of an individual filter. GA directly finds the coefficients of the desired filter transfer function through genetic search fur given filter specifications of minimum filter order. Crossover and mutation operators are selected to ensure the stability of resulting IIR filters. Other types of filters can be found independently from the filter specifications, not from algebraic transformations.

• 제어로봇시스템학회논문지
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• 제19권5호
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• pp.481-487
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• 2013

This paper presents an FIR (Finite Impulse Response) fixed-interval smoothing filter for fast and exact estimating state variables of a discrete nonlinear system with modeling uncertainty. Conventional IIR (Infinite Impulse Response) filter and smoothing filter can estimate state variables of a system with an exact model when the system is observable. When there is an uncertainty in the system model, however, conventional IIR filter and smoothing filter may cause large errors because the filters cannot estimate the state variables corresponding to the uncertain model exactly. To solve this problem, FIR filters that have fast estimation properties and have robustness to the modeling uncertainty have been developed. However, there is time-delay estimation phenomenon in the FIR filter. The FIR smoothing filter proposed in this paper makes up for the drawbacks of the IIR filter, IIR smoothing filter, and FIR filter. Therefore, the FIR smoothing filter has good estimation performance irrespective of modeling uncertainty. The proposed FIR smoothing filter is applied to the integrated navigation system composed of a magnetic compass based DR (Dead Reckoning) and a GPS (Global Positioning System) receiver. Even when the magnetic compass error that changes largely as the surrounding magnetic field is modeled as a random constant, it is shown that the FIR smoothing filter can estimate the varying magnetic compass error fast and exactly with simulation results.

• 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.

• 한국정보통신학회논문지
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• 제13권7호
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• pp.1312-1320
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• 2009

본 논문은 Particle Swarm Optimization(PSO)을 이용하여 2차원 IIR 디지털필터의 설계방법을 제안하였다. 먼저 2차원 디지털필터의 설계문제를 PSO에 적용하기 위하여 최소화 문제로써 형식화 과정이 논의된다. 제안된 PSO 알고리즘을 이용한 설계방법은 기존의 PSO 알고리즘에 IIR 필터설계에서 요구되는 안정성을 보증하는 과정이 검토되어 개선된다. 본 논문에서 제안된 방법의 타당성을 설계예시를 통해 고찰한 결과, 설계된 디지털필터는 동일한 설계사양으로 기존의 설계방법으로 설계된 디지털필터보다 근사오차 면에서 우수한 결과를 얻을 수 있었다. 또한 제안한 설계방법에 의한 2차원 IIR 디지털필터는 설계과정에서 필터의 안정성을 보증할 수 있었다.

• 한국통신학회논문지
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• 제28권6A호
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• pp.341-350
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• 2003

본 논문에서는 시공간 블록 부호 송신 다이버시티 기법(STBC-TD)이 적용된 WCDMA 하향 링크에서 이동 평균(Moving Average) 필터 구조의 채널 추정기에 대해 성능 평가를 하였다. 또한 FIR(Finite Impulse Response) 필터 구조의 채널 추정기에서 발생하는 메모리 요구 문제와 채널 추정 지연 시간 문제를 해결하기 위해 IIR(Infinite Impulse Response) 필터 구조를 제시하고 이에 대한 성능 평가 및 비교를 하였다. 컴퓨터 모의 실험 결과 일반적으로 STBC-TD 기법이 적용된 경우 두 구조 모두 이동국의 속도가 저속일 때 상당한 성능 개선이 발생하였다. 이동평균 필터 구조의 채널 추정기는 STBC-TD 기법을 적용함으로써 성능 개선과 함께 필터의 최적 탭 수가 감소되고 이로 인해 채널 추정 지연 시간과 수신단의 복잡도를 줄일 수 있었다. 또한 IIR 필터 구조의 채널 추정기는 이동평균 필터 구조에 비해 메모리 요구 문제와 추정 지연 시간 문제가 작은 장점을 가지지만 IIR 필터 내부의 계수는 이동국의 속도 변화에 매우 민감하며 따라서 최적의 IIR 필터 계수 설정이 중요한 고려 사항이다.

• 한국콘텐츠학회논문지
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• 제5권6호
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• pp.74-86
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• 2005

본 논문에서는 2 채널 입체음향 시스템을 효율적으로 구현하기 위하여 고차의 FIR 필터로 구현된 머리 전달함수를 저차의 IIR 필터로 근사시키는 알고리즘을 제안한다. 제안하는 알고리즘은 균형화모델감소법의 개념을 바탕으로 한다. 근사화된 머리전달함수를 헤드폰을 통해 바이노럴 사운드로 재생하면 입체음향이 구현된다. 더미 헤더의 머리전달함수를 512차 FIR 필터에서 32차 IIR 필터로 근사화하여 기존의 머리전달함수와 성능을 비교한다. 이를 위해 10명을 대상으로 음상 정위에 대한 실험을 한다. 실험은 컴퓨터 시뮬레이션과 TMS320C32를 이용한 하드웨어 실험을 병행한다. 실험을 통해 기존의 FIR 필터를 사용한 경우와 같이 IIR 필터로 근사화된 머리전달함수를 사용한 경우도 동일하게 인지되는 것을 확인할 수 있다.