• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 하계종합학술대회 논문집
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• pp.468-473
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• 1999

Neural networks for adaptive channel equalizers have been resorted to recently in digital communication systems. In this paper, the characteristics and the application areas and etc. for neural networks as adaptive channel equalizers are examined through simple examples.

• 한국통신학회논문지
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• 제19권5호
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• pp.939-951
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• 1994

최소 평균 제곱(least mean squares) 알고리듬에 기반을 둔 적응 동화기의 수렴속도는 입력신호의 공분산 행렬에 의해 결정된다. 공분산 행렬의 고유값 spread가 1에 가까울때 수렴속도는 매우 빠르다. 본 논문에서는 LMS 기반 적응 동화기의 빠른 수렴을 위해 통계적으로 주어진 채널에 적합한 변형된 기준신호를 제안하였다. 이론적인 분석과 모의실험을 통해 LMS 기반 적응 동화기의 빠른 수렴을 얻는데 이 변형 방법이 효과적임을 알 수 있었다.

• Journal of the Optical Society of Korea
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• 제15권3호
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• pp.232-236
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• 2011

We analyze the linear equalizers used in optically amplified on-off-keyed (OOK) systems to combat chromatic dispersion (CD) and polarization mode dispersion (PMD), and we derive the mathematical minimum mean squared error (MMSE) performance of these equalizers. Currently, the MMSE linear equalizer for optical OOK systems is obtained by simulations using adaptive approaches such as least mean squared (LMS) or constant modulus algorithm (CMA), but no theoretical studies on the optimal solutions for these equalizers have been performed. We model the optical OOK systems as square-law nonlinear channels and compute the MMSE equalizer coefficients directly from the estimated optical channel, signal power, and optical noise variance. The accuracy of the calculated MMSE equalizer coefficients and MMSE performance has been verified by simulations using adaptive algorithms.

• ETRI Journal
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• 제31권2호
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• pp.162-172
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• 2009

A linear-prediction-based blind equalization algorithm for single-input single-output (SISO) finite impulse response/infinite impulse response (FIR/IIR) channels is proposed. The new algorithm is based on second-order statistics, and it does not require channel order estimation. By oversampling the channel output, the SISO channel model is converted to a special single-input multiple-output (SIMO) model. Two forward linear predictors with consecutive prediction delays are applied to the subchannel outputs of the SIMO model. It is demonstrated that the partial parameters of the SIMO model can be estimated from the difference between the prediction errors when the length of the predictors is sufficiently large. The sufficient filter length for achieving the optimal prediction is also derived. Based on the estimated parameters, both batch and adaptive minimum-mean-square-error equalizers are developed. The performance of the proposed equalizers is evaluated by computer simulations and compared with existing algorithms.

• 한국정보통신학회논문지
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• 제22권2호
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• pp.253-260
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• 2018

천해 환경에서의 수중 음향 통신 채널은 전형적으로 시변 다중 경로 페이딩 채널 특성을 나타낸다. 이러한 채널 전송을 통해 수신된 신호는 시간 지연 및 진폭의 중첩에 의해 심볼 간 간섭을 유발한다. 이를 보완하기 위해 여러 기술이 사용되었으며, 그 중 하나가 음향 등화기이다. 본 연구에서는 심볼 간 간섭을 보상하기 위해 feed-forward equalizer (FFE), decision direct equalizer (DDE), decision feedback equalizer (DFE) 및 DFE와 결합된 DDE의 4 종류의 등화기와 등화기의 계수를 조정하기 위해 normalized least mean square (NLMS) 알고리즘과 recursive least square (RLS) 알고리즘의 2 종류의 알고리즘을 적용하였다. 그 결과 비선형 등화기에서는 신호 대 잡음비 6 dB 이상에서 상당한 성능 향상을 발견할 수 있었으며, DFE와 DDE의 조합은 어떤 경우에도 최고의 성능을 발휘하였다.

• 전자공학회논문지B
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• 제33B권6호
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• pp.164-175
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• 1996

In this thesis, to cope with severe intersymbol interference and nonlinear distotions of the digital magnetic recording channel a neural decision feedback equalizer (NDFE) and an adaptive neural equalizer (NE) are applied. The digital magnetic recording channels with various recording densities and different types of the nonlinear distortions are considered. The computer simulation shows that as the nonlinear distortion is increased, the neural equalizers (NdFE, NE) have advantages of approximately 2-4 dB in signal to noise ratio (SNR) over the onventional eualizers to reach sme bit error rate and, a sthe recording density is increased, 1~5 dB of SNR improvement are also gained. Especially the NdFE gives a superior performance over the other equalizers when there is a severe nonlinear distortion in the digital magnetic channel.

• 한국통신학회논문지
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• 제25권8B호
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• pp.1445-1452
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• 2000

일반적으로 통신채널은 심볼간 간섭(ISI)과 가산성 백색잡음(AWGN) 이 존재하는 환경으로 모델링할 수 있다 이러한 채널환경에서 비터비(Viterbi) 알고리즘은 송신측에서 전송한 데이터를 검출하기 위한 최적의 방법이다 하지만 비터비 알고리듬은 채널의 길이에 지수적으로 증가하는 많은 연산량 때문에 적합한 방법이 되지 못한다 이러한 문제점을 해결하기 위한방법들의 일종으로서 적응 등화기를 이용한 채널 등화 기법들이 널리 사용되고 있으나 채널이 혼합 위상 특성을 가지는 경우에는 비효율적인 방법이다 본 논문에서는 혼합 위상 특성을 가지는 채널을 위한 효율적인 채널 등화기를 제안하였으며 기존의 채널 등화 기법들과 비교하여 제안한 방법의 우수성을 보였다.

• 전자공학회논문지A
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• 제32A권8호
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• pp.11-18
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• 1995

This paper proposes a nonlinear adaptive equalizer which is based on fuzzy rules and fuzzy inference of several affine mapping for the received channel data. The proposed nolonlinear adaptive equalizers with the significantly lower computational complexity. Also it can be applied to the on-line adaptation environments owing to its fast convergence characteristics and the lower computational load. When using the decision feedback vectors, this equaalizer can be easily realized in the form of the DFE structure with out the requirement for the perfect channel knowledge as in the case of the fuzzy adaptive filter.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1991년도 학술발표회 논문집
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• pp.38-41
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• 1991

RLS algorithms are required for applications to adaptive line enhancers, adaptive equalizers for voiceband telephone and HF modems, and wide-badn digital spectrum mobile raio in which their convergence time and tracking speed are significant. The fast QR RLS algorithm satisfies above the requirements. Its computational complexity is linearly proportional to the tap number of a filter, N and its performance remains numerically stable. From the result of simumulation, the fast QR RLS algorithm represented Cioffi is better than gradient based algorithm in its initial performance when being applied to an adaptive line enhancer for cancelling noise.

• 전자공학회논문지B
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• 제33B권3호
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• pp.1-8
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• 1996

This paper proposes a novel transversal filter and an optimal LMS algorithm, and show how these can be realized as an adaptive equalizer. The transversal filter consists of a walsh and block pulse functions. in the LMS algorithm with equalizers, the convergence factor is an improtant design parameter because it governs stability and convergence speed. The conventional adaptation techniques use a fixed time constant convergence factor by the trial and error method. In this paper, an optimal method in the choice of the convergence factor is proposed. The proposed algorithm is obtrained that is tailored for each filter tap and is updated at each iteration. The performance of the proposed algorithm is compared iwth those of the conventional TDL and DFT equalizers by computer simulations.