• The Journal of the Acoustical Society of Korea
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• 제16권2E호
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• pp.14-20
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• 1997

The future teleconferencing systems will need an appropriate system which controls properly the acoustic echo for the convenient communication. The conventional acoustic echo cancellation algorithms involve large adaptive filters identifying the impulse response of the echo path. The use of adaptive IIR filters appears to be a reasonable way to reduce computational complexity. Effective cancellation of acoustic echo presented in teleconferencing system requires that adaptive filters have a rapid convergence speed. One of the main problems of acoustic echo cancellation techniques is that the convergence properties degrade for an highly correlated signal input such as speech signals. By the way, the introduction of linear prediction filers onto the structure of the acoustic echo cancellation represents one approach to decorrelate the speech signal. And variable step-size LMS algorithm improves the convergence speed through a little increasing of computational complexity. In this paper, we applied these two methods to the acoustic echo canceller(AEC) and showed that these methods have better performances than the conventional AEC.

• 한국국방경영분석학회지
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• 제13권1호
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• pp.1-18
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• 1987

This paper presents several adaptive linear predictive coding techniques based upon extension of recursive ladder filters. A 2-D recursive ladder filter is extended to a 3-D case which can adaptively track the variation of both spatial and temporal changes of moving images. Using the 2-D/3-D ladder filter and a previous farme predictor, two types of adaptive predictor-control schemes are proposed in which the prediction error at each pel can be obtained at or close to a minimum level. We also investigate several modifications of the basic encoding methods. Performance of the 2-D/3-D ladder filters, their adaptive control schemes, and variations in coding methods are evaluated by computer simulations on a real sequence and compared to the results of motion compensation and frame differential coders. As a validity test of the ladder filters developed, the error signals for the different predictors are compared and evaluated.

• 대한전자공학회논문지SP
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• 제41권6호
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• pp.285-294
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• 2004

신호처리에서 메디안 필터는 임펄스 잡음을 제거하는 비선형 필터 중에서 가장 널리 사용되고, 가장 강력한 효과를 보이고 있다. 본 논문은 잡음 검출에 의한 적응형 메디안 필터를 제안한다. 제안한 필터의 기본 알고리즘은 잡음 여부를 각 판단기준에 의해서 판별한 후, 판별 결과에 따라 조건을 만족하면 메디안 필터를 취하고, 만족하지 않으면 원 영상(No Filter)으로 복원한다. 잡음 판별을 위해서 로컬 분산과 로컬 중간값 분산을 이용한 잡음 검출을 제시했고, 기존의 [5]∼[10] 필터와 특성 및 성능을 비교 분석하였다. 제안한 필터는 기존의 필터를 같은 조건에서 수행한 결과보다 대부분의 경우에서 개선을 보이고, 주관적인 육안으로 판별했을 경우에도, 그 이상의 효과를 보임을 입증하였다. 따라서 로컬 분산과 로컬 중간값 분산을 이용한 적응형 메디안 필터는 메디안 필터의 임펄스 잡음 제거 특성에 강한 장점을 살리고, 에지 보존 능력이 강화되었음을 증명 하였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -2
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• pp.681-684
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• 2000

Gradient-type algorithms for adaptive IIR notch filters are very attractive in terms of both performances and computational requirements. Generally, it is quite difficult to assess their performances analytically. There have been several trials to analyze such adaptive algorithms as the sign and the plain gradient algorithms for some types of adaptive IIR notch filters, but many of them still remain unexplored. Furthermore, analysis techniques used in those trials can not be directly applied to different types of adaptive IIR notch filters. This paper presents a detailed performance analysis of the sign algorithm for a well-known adaptive IIR notch filter with constrained poles and zeros, which can not be done by just applying the related existing analysis techniques, and therefore has not been attempted yet. The steady-state estimation error and mean square error (MSE) of the algorithm are derived in closed forms. Stability bounds of the algorithm are also assessed. extensive simulations are conducted to support the analytical findings.

• 대한의용생체공학회:의공학회지
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• 제22권3호
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• pp.285-291
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• 2001

In this paper, we designed two preprocessing adaptive filter - high pass filter and notch filter - using FPGA. For minimizing the calculation load of multi-channel and high-resolution ECG system, we utilize FPGA rather than digital signal processing chip. To implement the designed filters in FPGA, we utilize FPGA design tool(Altera corporation, MAX-PLUS II) and CSE database as test data. In order to evaluate the performance in terms of processing time, we compared the designed filters with the digital filters implemented by ADSP21061(Analog Devices). As a result, the filters implemented by FPGA showed better performance than the filters based on ADSP21061. As a consequence of examination, we conclude that FPGA is a useful solution in multi-channel and high-resolution signal processing.

• 한국통신학회논문지
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• 제21권4호
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• pp.876-883
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• 1996

Adaptive FIR filters are widely used in the echo canceller. But, most of practical systems have the transfer function composed of poles and zeros. In that case, adaptive IIR filters may be more efficient rather than FIR fiters. In this paper, a recently developed C-HARF algorithm is used to implement an adaptive IIR echo canceller. The proposed convergence of the algorithm make it attractive for this application. Extensive computer simulations show that C-HARF algorithm performs better than the NLMS algorithm after convergence, although C-HARF algorithm converges more slowly.

• 전자공학회논문지B
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• 제29B권1호
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• pp.34-42
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• 1992

This paper presents two fast least-squares lattice algorithms for adaptive nonlinear filters equipped with bilinear system models. The lattice filters perform a Gram-Schmidt orthogonalization of the input data and have very good numerical properties. Furthermore, the computational complexity of the algorithms is an order of magnitude snaller than previously algorithm is an order of magnitude smaller than previously available methods. The first of the two approaches is an equation error algorithm that uses the measured desired response signal directly to comprte the adaptive filter outputs. This method is conceptually very simple`however, it will result in biased system models in the presence of measurement noise. The second approach is an approximate least-squares output error solution. In this case, the past samples of the output of the adaptive system itself are used to produce the filter output at the current time. Results of several experiments that demonstrate and compare the properties of the adaptive bilinear filters are also presented in this paper. These results indicate that the output error algorithm is less sensitive to output measurement noise than the squation error method.

• 전기학회논문지
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• 제62권1호
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• pp.133-139
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• 2013

Convex combination of two adaptive filters is an efficient method to improve adaptive filter performances. In this paper, a subband convex combination method of two adaptive filters for fast convergence rate in the transient state and low steady state error is presented. The cost function of mixing parameter for a subband convex combination is defined, and from this, the coefficient update equation is derived. Steady state analysis is used to prove the stability of the subband convex combination. Some simulation examples in system identification scenario show the validity of the subband convex combination schemes.

• 대한원격탐사학회지
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• 제17권4호
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• pp.319-334
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• 2001

Extracting primitives from imagery plays an important task in visual information processing since the primitives provide useful information about characteristics of the objects and patterns. The human visual system utilizes features without difficulty for image interpretation, scene analysis and object recognition. However, to extract and to analyze feature are difficult processing. The ultimate goal of digital image processing is to extract information and reconstruct objects automatically. The objective of this study is to develop robust method to achieve the goal of the image processing. In this study, an adaptive strategy was developed by implementing Gabor filters in order to extract feature information and to segment images. The Gabor filters are conceived as hypothetical structures of the retinal receptive fields in human vision system. Therefore, to develop a method which resembles the performance of human visual perception is possible using the Gabor filters. A method to compute appropriate parameters of the Gabor filters without human visual inspection is proposed. The entire framework is based on the theory of human visual perception. Digital images were used to evaluate the performance of the proposed strategy. The results show that the proposed adaptive approach improves performance of the Gabor filters for feature extraction and segmentation.

• 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.