• Journal of Korea Multimedia Society
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• v.4 no.3
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• pp.230-237
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• 2001

Adaptive filters are widely used for acoustic echo canceler, adaptive equalizer and adaptive noise canceler. Coefficients of adaptive filters are updated by NLMS algorithm. However, Coefficients are misaligned by ambient noises when they are adapted by NLMS algorithm. In this Paper, a method determined the adaptation constant by low-pass filtered instantaneous gradient vector of LMS algorithm using orthognality principles of optimal filter is proposed. At initial states, instantaneous gradient vector, that is the cross-correlation of input signals and estimation error signals, has large value because input signals are remained in estimation error signals. When an adaptive filter is conversed, the cross-correlation will be close to zero. It isn's affected by ambient noises because ambient noises are uncorrelated with input signals. Determining adaptation constant with the cross-correlation, adaptive filters can be robust to ambient noises and the convergence rate doesn't slower As results of computer simulations, it is shown that the performance of proposed algorithm is betted than that of conventional algorithms.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.4
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• pp.173-179
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• 2002

This paper describes an equalizer using the DFE (Decision Feedback Equalizer) structure, CMA (Constant Modulus Algorithm) and LMS (Least Mean Square) algorithms. The DFE structure has better channel adaptive performance and lower BER than the transversal structure. The proposed equalizer can be used for 16/64 QAM modems. We employ high speed multipliers, square logics and many CSAs (Carry Save Adder) for high speed operations. We have developed floating-point models and fixed-point models using the COSSAP$\^$TM/ CAD tool and developed VHDL filter. The proposed equalizer shows low BER in multipath fading channel. We have performed models. From the simulation results, we employ a 12 tap feedback filter and a 8 tap feedforward logic synthesis using the SYNOPSYS$\^$TM/ CAD tool and the SAMSUNG 0.5$\mu\textrm{m}$ standard cell library (STD80) and verified function and timing simulations. The total number of gates is about 130,000.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2004.11a
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• pp.197-200
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• 2004

디지털 TV 전송 방식중의 하나인 ATSC 8-VSB 시스템의 등화기는 훈련신호가 존재하는 구간에서 LMS 알고리즘을 사용하는 DFE 적옹 등화기가 사용된다. 그러나 LMS 알고리즘은 그 수렴속도가 느리고 수렴 후 오차 수준이 다른 적응 알고리즘에 비해 높다는 단점이 있다. 본 논문에서는 LMS 알고리즘을 사용하는 DFE의 오차 수준을 낮추기 위한 선형 등화기 구조의 전 처리부(pre-processor)를 사용하여 필터 수렴 후의 DFE의 오차수준을 기존의 DFE보다 낮추었으며 제안된, DFE 구조의 성능을 컴퓨터 모의 실험을 통해 분석하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1329-1338
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• 1999

In this paper, an efficient signal interference control technique to improve the convergence speed of LMS algorithm is introduced. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. According as the step-size parameter $\mu$ is increased, the rate of convergence of the algorithm is controlled. Also, a increase in the step-size parameter $\mu$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the mean squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• Proceedings of the KIEE Conference
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• 1999.11c
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• pp.760-762
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• 1999

본 논문에서는 기존의 적응필터인 LMS(Least Mean Square)와 RLS(Recursive Least Square)의 수렴속도의 향상과 안정성을 개선하기 위한 방안을 제안하였다. 제안된 알고리즘은 기존의 시간영역 LMS 알고리즘보다 상당히 빠른 수렴속도를 보일 수 있도록 설계하였다. RLS 알고리즘는 역행렬연산으로 인한 연산량이 많고 자기상관행렬이 positive definite 특성을 잃어버릴 경우 시스템이 수치적으로 불안정하게 되어 발산하는 단점이 있다. 이런한 단점을 보완하기 위해 제안된 알고리즘을 사용하였다. 기존의 알고리즘은 전력 정규화 과정에서 입력신호의 변환이 백색화가 완전히 이루어지지 않게 되어 자기상관행렬이 순수한 대각행렬이 되지 않는 단점을 지니고 있으나, 본 연구에서는 이러한 대각화 과정에서 좀더 많은 정보를 포함하도록 설계하였다. 아울러 제안된 알고리즘을 적응 등화기에 적용하여 수렴속도가 개선됨을 검증하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1998.11a
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• pp.398-402
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• 1998

본 논문에서는 유ㆍ무선 통신 채널을 통한 데이터 전송시 발생하는 비선형 왜곡을 적응 필터를 이용하여 보상하였다. 특히 통신채널에서는 심볼간 간섭(ISI)이 발생하는데 이러한 간섭을 비선형 필터를 이용하여 제거하였다. 비선형 채널을 모델링하는 방법에는 볼테라급수를 이용하는 방법과 쌍선형 방법이 있다. 쌍선형 방법은 볼테라 방법에 비하여 계산량이 적은 장점을 지니고 있다. 따라서 쌍선형 필터에 적응 알고리듬을 적용하여 신호의 왜곡을 보상하였다. 적응 알고리듬에는 LMS 계열과 LS 계열 알고리듬이 있으나 통신 채널에서는 알고리듬의 안정도가 중요하므로 LMS 계열 알고리듬을 적용하였다. 또한 적응 알고리듬은 입력 데이터의 상관성과 데이터 수에 의존하여 수렴속도와 안정도가 결정된다. 알고리듬의 수렴속도를 증가시키기 위하여 입력신호를 신호파형으로부터 다량의 데이터를 검출하는 방법을 적용하였다. 이러한 방법을 입증하기 위하여 입력신호는 2진 랜덤 가우시안 데이터를 이용하였고, 통신채널에서 채널간 간섭을 발생시켰으며 화이트 가우시안 잡음을 부가 시켰다. 이러한 신호를 수신한 수신기에 적응 등화기를 설계하여 대량의 데이터를 생성시키고, 적응 알고리듬을 적용하여 채널의 왜곡을 빠른 속도로 보상하였다.

• The Journal of the Acoustical Society of Korea
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• v.12 no.1E
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• pp.5-11
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• 1993

본 논문에서는 페이딩 채널에서 백색 가우시안 잡음과 임펄스성 잡음이 부가된 디지털 신호를 복원하기 위하여 적응 LMS알고리즘과 RLS 알고리즘을 사용하여 TDL 등화기, 결정 궤환 등화기, 그리고 격자 등화기의 성능을 평가하고 비교하였다. 오차 성능 분석 결과, 페이딩이 존재하고 임펄스성과 가우스성 잡음이 존재하는 채널에서 10-3BER을 얻기 위해서, 격자 등화기는 LMS이 등화기보다 3.0dB, RLS TDL 등화기보다 3.9dB의 신호대 잡음비(SNR)여유를, 그리고 LMS DFE 등화기보다 0.5dB, RLS DFE등화기보다 -0.5dB의 SNR 여유를 갖음을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.9
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• pp.2456-2469
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• 1996

A digital satellite communication channel has a nonlinearity with memory due to saturation characeristis of the high poer amplifier in the satellite and transmitter/receiver linear filter used in the overall system. In this paper, we propose a complex radial basis function network(CRBFN) based adaptive equalizer for compensation of nonlinearities in digital satellite communication channels. The proposed CRBFN untilizes a complex-valued hybrid learning algorithm of k-means clustering and LMS(least mean sequare) algorithm that is an extension of Moody Darken's algorithm for real-valued data. We evaluate performance of CRBFN in terms of symbol error rates and mean squared errors nder various noise conditions for 4-PSK(phase shift keying) digital modulation schemes and compare with those of comples pth order inverse adaptive Volterra filter. The computer simulation results show that the proposed CRBFN ehibits good equalization, low computational complexity and fast learning capabilities.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.709-717
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• 2002

A new multi constant modulus algorithm (MCMA) for a blind decision feedback equalizer is proposed. In order to avoid the error propagation problem in the conventional DFE structure, Feed-Back Filter coefficients are updated only after Feed-Forward Filter coefficients are sufficiently converged to the steady state. Therefore, it has the problem of slow convergence speed characteristics. To overcome this drawback, the proposed MCMA algorithm uses not only new cost function considering the minimum distance between the received signal and the representative value containing the statistical characteristics of the transmitted signal, but also adaptive step-size according to the equalizer outputs to fast convergence speed of FBF. Simulations were carried out under the certified communication channel environment to evaluate a performance of the proposed equalizer. The simulation results show that the proposed equalizer has an improved convergence and SER performance compared with previous methods. The proposed techniques offer the possibility of practical equalization for cable modem and terrestrial HDTV broadcast (using 8-VSB or 64-QAM) applications.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.9
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• pp.62-70
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• 1995

In some practical applications of the LMS Algorithm the coefficient adaptation can be performed only after some fixed delay. The resulting algorithm is known as the Delayed Least Mean Square (DLMS) algorithm in the literature. There exist analyses for this algorithm, but most of them are based on the unrealistic independence assumption between successive input vectors. Inthis paper we consider the DLMS algorithm with decreasing step size .mu.(n)=n/a, a>0 and prove the almost-sure convergence ofthe weight vector W(n) to the Wiener solution W$_{opt}$ as n .rarw. .inf. under the mixing unput condition and the satisfaction of the law of large numbers. Computer simulations for decision-directed adaptive equalizer with decoding delay are performed to demonstrate the functioning of the proposed algorithm.m.