• 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.

• ETRI Journal
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• 제32권4호
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• pp.626-629
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• 2010

In this letter, we present an enhanced block-based adaptive loop filter (E-BALF) with multiple filter symmetric structures. The E-BALF adapts various filter symmetric structures in a rate-distortion optimization sense, reflecting the statistical properties of each image in a video sequence. Experimental results show that the proposed method achieves a reduction in the Bj${\phi}$ntegaard delta (BD)-bitrate by an average of 9.60% compared with Joint Model 11.0 of H.264/AVC. Compared to the state-of-the-art BALF, a reduction of up to 1.13% in BD-bitrate is achieved.

• 한국통신학회:학술대회논문집
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• 한국통신학회 1991년도 추계종합학술발표회논문집
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• pp.111-114
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• 1991

This paper proposes an efficient adaptive echo canceller using pilot filter approach to achieve improved convergence speed. The pilot filter is an adaptive filter with only a few filter coefficients to filter the received signal for the purpose of whitening the signal. Thus the convergence speed of the main LMS-TDL filter combined with the pilot filter is improved. In the proposed echo canceller, an adaptive lattice predictor as the pilot filter is used and its inverse filter is used to equalize the distorted near end talker signal. Simulation results for colored signal show that the convergence speed of the proposed echo cancellation algorithm is faster than that of the conventional LMS-TDL echo cancellation algorithm.

• 한국멀티미디어학회논문지
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• 제12권9호
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• pp.1210-1216
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• 2009

본 논문에서는 순환최소순위(RLR) L-필터의 수렴성을 해석하였다. RLR L-필터는 순서통계필터로서 입력의 크기순서에 따른 가중치를 필터계수로 한다. 또한 RLR L-필터는 비선형 적응 필터로서 필터계수의 갱신을 위하여 RLR 알고리즘을 이용한다. RLR 알고리즘은 로버스트 통계학의 순위추정에 기초한 비선형 적응 알고리즘이다. 본 논문에서는 가변적인 스텝 크기를 적용하여 평균 및 평균제곱의 견지에서 수렴성을 해석하였다. RLRL-필터는 잡음의 분포함수가 두꺼운 꼬리 분포인 임펄스 잡음에 가까울수록 메디안 필터의 형태로 적응하며 가우시안 잡음의 경우 평균 필터의 형태로 적응한다.

• 한국통신학회논문지
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• 제10권6호
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• pp.308-316
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• 1985

주파수 영역에서 블록 적응 필터(Block Adaptive Filter),의 새로운 구현 방법을 제시하였다. 블록 적응 필터링은 입력값을 Block 단위로 하였을 때 출력에서도 블록 단위 또는 유한한 값을 갖도록 하는 것이다. 본 논문에서는 Gordard 이론을 이용하여 Block Adaptive Filter의 고속 수렴 알고리즘을 전개하고 Saito의 알고리즘과 BLMS알고리즘의 수렴 결과와 비교하였다. 그 결과 본 논문에서 제시한 주파수 영역블럭 적응 필터의 알고리즘(FBAF)이 BLMS알고리즘의 수렴상태보다 빠르게 수렴하며 Satio의 알고리즘의 수렴 오차보다는 줄어들었다.

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

• 한국자동차공학회논문집
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• 제19권4호
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• pp.84-90
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• 2011

In active noise control filter, LMS algorithms which used for control filter, assure the convergence property, and computational burden of these algorithms are proportionate to the filter taps. The convergence speed of LMS algorithms is mainly determined by value of the convergence coefficient, so optimal selection of the value of convergence coefficient is very important. In this paper, We proposed novel adaptive fuzzy logic LMS algorithms with FIR filter structure which has better convergence speed and less computational burden than conventional LMS algorithms, for single channel active noise control with ill conditioned signal case. Computer simulations were performed to show the effectiveness of a proposed algorithms.

• The Journal of the Acoustical Society of Korea
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• 제27권3E호
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• pp.95-103
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• 2008

In this paper, the new hybrid adaptive Volterra filter was proposed to be applied for compensating the nonlinear distortion of memoryless nonlinear systems with saturation characteristics. Through computer simulations as well as the analytical analysis, it could be shown that it is possible for both conventional Volterra filter and proposed hybrid Volterra filter, to be applied for linearizing the memoryless nonlinear system with nonlinear distortion. Also, the simulations results demonstrated that the proposed hybrid filter may have faster convergence speed and better capability of compensating the nonlinear distortion than the conventional Volterra filter.

• 한국음향학회지
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• 제12권1호
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• pp.37-46
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• 1993

The adaptive algorithm for the Volterra filter is considered. Owing to its simplicity, the LMS algorithm for adaptive Volterra filter(AVF) is widely used as in linear adaptive filters. However, the convergence speed is unsatisfactory. For improving the convergence speed, the frequency domain LMS second order adaptive Volterra filter(FLMS-AVF) is proposed and analyzed. We show that the time and frequency domain LMS AVF's have the same steady state performance under approprate conditons. Moreover, it can be shown that this algorithm can improve the convergence speed significantly by applying self-orthogonalizing method.

• ETRI Journal
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• 제34권1호
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• pp.106-109
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• 2012

In this letter, we propose a new prediction block filter that can reduce errors between the original and prediction blocks. The proposed filter adaptively adjusts filter coefficients by using the previously reconstructed adjacent blocks and their prediction blocks. Then, the filter is selectively applied to the current prediction block according to the rate-distortion optimization. Moreover, since the same filter coefficients can be derived in the decoder, they are not encoded into the bit-stream. The proposed method achieves a 4.65% bitrate saving on average compared with H.264/AVC.