• 한국통신학회논문지
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• 제27권8A호
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• pp.753-762
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• 2002

Blind equalization of transmission channel is important in communication areas and signal processing applications because it does not need training sequences, nor dose it require a priori channel information. In this paper, an adaptive blind MMSE channel equalization technique based on second-order statistics in investigated. We present an adaptive blind MMSE channel equalization using multichannel linear prediction error method for estimating cross-correlation vector. They can be implemented as RLS or LMS algorithms to recursively update the cross-correlation vector. Once cross-correlation vector is available, it can be used for MMSE channel equalization. Unlike many known subspace methods, our proposed algorithms do not require channel order estimation. Therefore, our algorithms are robust to channel order mismatch. Performance of our algorithms and comparisons with existing algorithms are shown for real measured digital microwave channel.

• 한국컴퓨터정보학회논문지
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• 제24권2호
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• pp.57-66
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• 2019

This paper compared the performance with an overview of channel equalization techniques used in high-speed serial transceivers, including the homogeneous architecture and associated components for the GHz interconnect of backplane and cable channels. It also used the MATLAB tool to present system analysis and simulation results for continuous time equivalent structures. In the case of conventional continuous equalization, high frequency deficits occur due to the use of a comparator that is difficult to implement as well as the low speed limit. In this paper, the channel equalization technique based on the power spectrum analysis of clocks was used to compensate for the frequency loss, and the application of the TX+Channel and TX+Equalizer filters enabled the measurement of attenuation and equivalence without comparators. The application of blender and band-pass filters at high speeds also showed significant effectiveness.

• 한국통신학회논문지
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• 제28권4C호
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• pp.384-391
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• 2003

Blind channel identification and equalization attempt to identify the communication channel and to remove the inter-symbol interference caused by a communication channel without using any known trainning sequences. In this paper, we propose a blind adaptive channel identification and equalization algorithm with phase offset compensation for single-input multiple-output (SIMO) channel. It is based on the one-step forward multichannel linear prediction error method and can be implemented by an RLS algorithm. Phase offset problem, we use a blind adaptive algorithm called the constant modulus derotator (CMD) algorithm based on condtant modulus algorithm (CMA). Moreover, unlike many known subspace (SS) methods or cross relation (CR) methods, our proposed algorithms do not require channel order estimation. Therefore, our algorithms are robust to channel order mismatch.

• The Journal of the Acoustical Society of Korea
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• 제26권2E호
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• pp.63-68
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• 2007

Using the neural network model for oriented principal component analysis (OPCA), we propose a solution to the data least squares (DLS) problem, in which the error is assumed to lie in the data matrix only. In this paper, we applied this neural network model to channel equalization. Simulations show that the neural network based DLS outperforms ordinary least squares in channel equalization problems.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제4권3호
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• pp.259-265
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• 2004

In this study, a hybrid genetic algorithm merged with simulated annealing is presented to solve nonlinear channel blind equalization problems. The equalization of nonlinear channels is more complicated one, but it is of more practical use in real world environments. The proposed hybrid genetic algorithm with simulated annealing is used to estimate the output states of nonlinear channel, based on the Bayesian likelihood fitness function, instead of the channel parameters. By using the desired channel states derived from these estimated output states of the nonlinear channel, the Bayesian equalizer is implemented to reconstruct transmitted symbols. In the simulations, binary signals are generated at random with Gaussian noise. The performance of the proposed method is compared with those of a conventional genetic algorithm(GA) and a simplex GA. In particular, we observe a relatively high accuracy and fast convergence of the method.

• 한국전자파학회논문지
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• 제18권12호
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• pp.1408-1418
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• 2007

본 논문에서는 주파수 선택적 시변 채널 OFDM 시스템에서 파일럿 심볼을 이용하는 채널 예측 및 등화 방식의 성능을 분석하고 문제점 및 개선 방안을 제시한다. 주파수 영역 채널 행렬의 대역폭을 제한하기 위하여 시간 영역 윈도우를 도입하며, 채널 예측에 LS(Least Square) 및 선형 MMSE(Minimum Mean Square Error) 예측 방식을 이용한다. 채널 등화에는 채널 예측에 이용되는 파일럿 심볼의 존재가 고려된 선형 MMSE 및 결정 귀환 등화 방식을 이용한다. 계산량을 감소시키기 위하여 채널 등화 알고리즘에 요구되는 선형 연립 방정식의 풀이에 band LU 행렬 분해 알고리즘을 도입하며, 컴퓨터 시뮬레이션을 통하여 기존 방식과의 성능을 비교한다. 결정 귀환 등화 방식에 시간 영역 윈도우를 적용하는 경우, 결정 귀환 관련 행렬의 대역폭이 제한되지 않아 성능이 저하되는 문제점을 분석하였다.

• Journal of Communications and Networks
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• 제6권1호
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• pp.19-25
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• 2004

In this paper, we present a trellis-based blind channel estimation and equalization technique coupling two kinds of adaptive Viterbi algorithms. First, the initial blind channel estimation is accomplished by incorporating the list parallel Viterbi algorithm with the least mean square (LMS) updating approach. In this operation, multiple trellis mappings are preserved simultaneously and ranked in terms of path metrics. Equivalently, multiple channel estimates are maintained and updated once a single symbol is received. Second, the best channel estimate from the above operation will be adopted to set up the whole trellis. The conventional adaptive Viterbi algorithm is then applied to detect the signal and further update the channel estimate alternately. A small delay is introduced for the symbol detection and the decision feedback to smooth the noise impact. An automatic switch between the above two operations is also proposed by exploiting the evolution of path metrics and the linear constraint inherent in the trellis mapping. Simulation has shown an overall excellent performance of the proposed scheme in terms of mean square error (MSE) for channel estimation, robustness to the initial channel guess, computational complexity, and channel equalization.

• Journal of Communications and Networks
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• 제18권1호
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• pp.8-18
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• 2016

In this paper, we propose a number of blind equalization approaches for time-varying andmulti-path channels. The approaches employ cost reference particle filter (CRPF) as the symbol estimator, and additionally employ either least mean squares algorithm, recursive least squares algorithm, or $H{\infty}$ filter (HF) as a channel estimator such that they are jointly employed for the strategy of "Rao-Blackwellization," or equally called "mixture filtering." The novel feature of the proposed approaches is that the blind equalization is performed based on direct channel estimation with unknown noise statistics of the received signals and channel state system while the channel is not directly estimated in the conventional method, and the noise information if known in similar Kalman mixture filtering approach. Simulation results show that the proposed approaches estimate the transmitted symbols and time-varying channel very effectively, and outperform the previously proposed approach which requires the noise information in its application.

• International Journal of Contents
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• 제2권4호
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• pp.19-24
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• 2006

In this paper, the performance of pre-equalization technique which can be applicable for the B-WLL uplink is evaluated and compared to post-equalization technique under three kinds of rain attenuation channels such as rain, intermittent light rain and thundershower. The BER performance comparisons of two algorithms (LMS and RLS) are investigated in the context of channel models and the length of training sequence. From the simulation results, it is shown that the post-equalization outperforms only at quite good channel conditions such as AWGN, while the pre-equalization can guarantee better BER performance at every channel conditions, especially performance gain increases as the severity of channel increases. It is concluded that the pre-equalizer using LMS algorithm is preferable at delay-tolerant situation where the complexity of algorithm is not a strict factor, while one using RLS is suitable for fast burst transmission with a relatively short training sequence.

• 한국멀티미디어학회논문지
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• 제7권6호
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• pp.781-790
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• 2004

This paper presents a fourth-order cumulants based iterative algorithm for blind channel equalization. It is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum phase characteristic of the channel. In this approach, the transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel outputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple reordering and scaling. Both a closed-form and a stochastic version of the proposed algorithm are tested with three-ray multi-path channels in simulation studies, and their performances are compared with a method based on conventional second-order cumulants. Relatively good results are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.