• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 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.

• ETRI Journal
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• 제4권3호
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• pp.18-29
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• 1982

In a block adaptive filtering procedure, the filter coefficients are adjusted once per each output block while maintaining performance comparable to that of widely used LMS adaptive filtering in which the filter coefficients are adjusted once per each output data sample. An efficient implementation of block adaptive filter is possible by means of discrete transform technique which has cyclic convolution property and fast algorithms. In this paper, the block adaptive filtering using Fermat Number Transform (FNT) is investigated to exploit the computational efficiency and less quantization effect on the performance compared with finite precision FFT realization. And this has been verified by computer simulation for several applications including adaptive channel equalizer and system identification.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -1
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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.

• 한국통신학회논문지
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• 제22권4호
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• pp.706-713
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• 1997

In this paper, we propose a band stop filter(BSF) for reducing drag-like effect of the low pass filter(LPF), a block by block adaptive filtering method, and a motion adaptive filtering method, which show better results in terms of PSNR or human visual perception compared to the conventional method using LPF. The BSF improves the draglike effects of the low pass filter by passing temporal high frequency components of video sequences which correspond to objects with large motion. The proposed adaptive methods also improve the conventional adaptive filtering by modifying the conventional algorithm and applying the algorithms for small blocks. The simulation results show that the proposed filtering methods show better results in terms of PSNR and subjective tests in most cases. Also in case of block by block adaptive filtering, it is verified that the application of the algorithm for smaller block gives better results.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.838-841
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• 2009

This paper proposes a novel lossless coding scheme for Bayer color filter array (CFA) images which are generally used as internal data of color digital cameras having a single image sensor. The scheme employs a block-adaptive prediction method to exploit spatial and spectral correlations in local areas containing different color signals. In order to allow adaptive prediction suitable for the respective color signals, four kinds of linear predictors which correspond to 2 ${\times}$ 2 samples of Bayer CFA are simultaneously switched block-by-block. Experimental results show that the proposed scheme outperforms other state-of-the-art lossless coding schemes in terms of coding efficiency for Bayer CFA images.

• 방송공학회논문지
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• 제17권6호
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• pp.1048-1060
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• 2012

본 논문에서는 영상에 포함되어 있는 잡음을 효율적으로 제거하기 위해 원본 영상과 잡음이 포함된 영상 사이의 mean square error (MSE) 추정에 기반한 적응적인 시공간 디노이징 필터(Adaptive Temporal and Spatial De-noising Filter : ATSF)를 제안하였다. 제안하는 디노이징 필터는 잡음이 포함되어 있는 영상에 블록 단위로 적용되며, 시간적 필터인 Multi-Hypothesis Motion Compensated Filter (MHMCF)를 사용하고, 공간적 필터로는 bilateral filter를 사용하였다. 각각의 블록에 대해 시간적 필터와 공간적 필터 중에서 최적의 필터를 선택하기 위해서 잡음이 포함되지 않은 원본 영상과 잡음이 포함된 입력 영상 사이의 MSE를 추정하는 기법을 제안하였다. 디노이징 단계에서 원본 영상이 주어지지 않기 때문에 MSE를 추정하기 위해서, 본 논문에서는 MHMCF가 적용된 블록의 MSE를 수학적으로 예측하고, bilateral filter가 적용된 블록의 MSE를 통계적 선형 모델을 통해 예측하였다. 이렇게 예측된 MSE를 비교하여 더 작은 MSE를 갖는 필터를 선택적으로 매 단위 블록마다 적용하게 된다. 제안된 방법은 시간적 필터와 공간적 필터를 적응적으로 적용함으로써 기존의 디노이징 방법에 비해 객관적 화질 뿐만 아니라 주관적인 화질에서 우수한 성능을 보여준다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.689-692
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• 2003

In this paper, we propose an adaptive beamforming algorithm for array antenna. The proposed beamforming algorithm, based on Block LMS (Block - Least Mean Squares) algorithm, has a variable step size from coefficient update. This method shows some advantages that the convergence speed is fast and the calculation time can reduced using a block LMS algorithm from frequency domain. As the adaptive parameter approaches a stationary state, it could reduce the number of filter coefficient update with the help of various step size. In this paper we compared the efficiency of the proposed algorithm with a standard LMS algorithm which is a representative method of adaptive beamforming.

• 대한전자공학회논문지
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• 제24권6호
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• pp.1005-1012
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• 1987

An adaptive scalar DCT-Wiener filtering method that can be applied to the problem of restoring an image degraded by white Gaussian noise is presented. In this method, the power spectrum needed to Winer filering is adaptively estimated in block-wise according to the lical properties in transform domain. In addition, overlapping method for reducing the block artifact is considered. Experimental results show that the adaptive Wiener filter by the proposed method yields performance improvement and better image quality over the nonadaptive one and the spatial Lee filter.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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• pp.240-240
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• 2000

This paper presents two algorithms based on the concept of the frequency domain adaptive filter(FDAF). First the frequency domain recursive least squares(FRLS) algorithm with the overlap-save filtering technique is introduced. This minimizes the sum of exponentially weighted square errors in the frequency domain. To eliminate discrepancies between the linear convolution and the circular convolution, the overlap-save method is utilized. Second, the sliding method of data blocks is studied Co overcome processing delays and complexity roads of the FRLS algorithm. The size of the extended data block is twice as long as the filter tap length. It is possible to slide the data block variously by the adjustable hopping index. By selecting the hopping index appropriately, we can take a trade-off between the convergence rate and the computational complexity. When the input signal is highly correlated and the length of the target FIR filter is huge, the FRLS algorithm based on the block processing technique has good performances in the convergence rate and the computational complexity.