• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.5 s.335
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• pp.13-22
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• 2005

IEEE 802.11a is a standard for the high-speed wireless local area network (WLAN), supporting from 6 up to 54 Mbps in a 5 GHz band. We propose a channel equalization algerian and a sampling clock recovery algorithm by utilizing the Viterbi decoder output of the IEEE 802.11a WLAN standard. The proposed channel equalizer adaptively compensates channel variations. The proposed system uses re-encoded Viterbi decoder outputs as reference symbols for the adaptation of the channel equalizer. It also extracts sampling phase information with the Viterbi decoder outputs for fine adjustment of the sampling clock. The proposed sampling clock recovery and equalizer are more robust to noise and frequency selective fading environments than conventional systems using only four pilot samples.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3A
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• pp.328-336
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• 2006

A novel pre-equalization method for terrestrial DTV transmitters is presented. A pre-equalizer has been used in transmitters to correct group delay and amplitude distortions caused by a channel filter. In the proposed pre-equalizer, an equation-error adaptive IIR filtering scheme is adopted unlike the conventional pre-equalization using FIR filtering schemes. The pole-zero modelling property of IIR filters improves the signal-to-noise ratio and may deal with diverse linear distortions existing in DTV transmitters as well as the channel filter distortion. Simulation results show that the proposed IIR pre-equalizer performs better than the FIR pre-equalizer in terms of the residual mean-square error.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.5 no.3
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• pp.169-173
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• 2012

This paper investigates the performance of Turbo equalization in wireless multipath channels. Turbo equalization mainly consists of a SISO(soft-in soft-out) equalizer and a SISO decoder. Iterative channel estimators can improve the accuracy of channel estimates by soft information fed back from the SISO decoder. Comparing iterative channel estimators with LMS(least mean square) and RLS(recursive least squares) algorithms, which are the most common algorithms to estimate and track a time-varying channel impulse response, the iterative channel estimator with RLS converges more faster than the one with LMS. However, the difference of BER(bit error rate) performances gradually decreases as the number of iterations for Turbo equalization increases.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.10A
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• pp.1794-1801
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• 2001

다중 반송파 변조의 일종인 OFDM(Orthogonal Frequency Division Multiplexing) 방식은 기존의 주파수 분할 다중화 기법에 각 부 채널들 사이의 직교성까지 보장함으로써 사용 주파수 효율을 극대화시킬 수 있는 방식이다. 그러나 주파수 비 선택적 다중경로 채널에서 높은 데이터 율을 갖는 OFDM 신호의 전송은 주파수 영역에서 인접 심벌간 간섭과 동일한 형태를 갖는 승산성 왜곡에 의해 영향을 받는다. 본 논문에서는 결정 궤환 시퀀스 등화기를 OFDM 시스템에 적용하여 신호를 등화하고, 등화를 통하여 주파수 영역에서의 인접 심벌간 간섭을 보상하는 방법을 제안한다. 결정 궤환 시퀀스 등화를 이용하여 최우 시퀀스 등화가 갖는 복잡도를 크게 줄이면서 주파수 오프셋이 추정될 수 있음을 설명하고 모의 실험을 통해 OFDM 시스템의 성능 개선을 보인다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2004.10a
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• pp.481-484
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• 2004

본 논문에서는 적응형 OFDM 시스템의 채널 추정 알고리즘과 무선 채널환경에서 시스템에 미치는 영향을 분석하고 추정된 결과를 이용하여 적응형 등화기를 적용한 적응형 OFDM 시스템의 성능을 BER과 성좌도를 통하여 분석한다. 분석결과, 기존의 고정된 변조 방식을 사용하는 시스템인 경우 고속의 데이터 전송을 위해 높은 지수의 변조모드(64QAM)를 사용함으로써 채널 환경에 민감하게 심볼오류 발생률이 증가함을 알 수 있었다. 하지만 채널상태에 따라 가변을 취하는 적응형 OFDM 시스템의 경우 64QAM방식에 비해 (BER=10-2)를 기준으로 약 8dB 성능의 이득이 있음을 알 수 있고 적응형 등화기를 적용함에 따라 왜곡을 받은 신호 파형의 보상이 가능함을 알 수 있다. 무수한 장애의 요인을 갖은 무선 채널환경에서 효율적인 데이터 전송을 위해 적응변조시스템이 요구되어짐을 알 수 있었다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.276-280
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• 2004

본 논문에서는 Matlab Tool을 이용하여 적응변조 OFDM 시스템 GUI 시뮬레이터를 설계하였다. 설계된 시뮬레이터를 이용하여 적응변조 기법을 위한 채널 추정 알고리즘과 무선 채널환경에서 시스템에 미치는 영향을 분석하고 추정된 결과를 이용하여 적응형 등화기를 적용한 적응변조 OFDM 시스템의 성능을 BER과 성좌도를 통하여 분석한다. 분석결과, 기존의 고정된 변조 방식을 사용하는 시스템인 경우 고속의 데이터 전송을 위해 높은 지수의 변조모드(64QAM)를 사용함으로써 채널 환경에 민감하게 심볼오류 발생률이 증가함을 알 수 있었다. 하지만 채널상태에 따라 가변을 취하는 적응변조 OFDM 시스템의 경우 64QAM방식에 비해 (BER=10-2)를 기준으로 약 8㏈ 성능의 이득이 있음을 알 수 있고 적응형 등화기를 적용함에 따라 왜곡을 받은 신호 파형의 보상이 가능함을 알 수 있다. 무수한 장애의 요인을 갖은 무선 채널환경에서 효율적인 데이터 전송을 위해 적응변조시스템이 요구되어짐을 알 수 있었다.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.32-38
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• 2001

Multi-carrier data transmission system performance is mostly limited by Inter- symbol-interference that is caused by a dispersive characteristic of the transmission channel. In order to enhance the system performance to meet the service requirements of local access, the channel impulse response shortening method incorporated with a channel frequency response compensation method is proposed. For a fast and efficient implementation of the equalizer proposed, Kalman and LMS algorithms are successively used. To verify the channel equalization performance, a set of computer simulation is performed on a filter bank based multitone system operating in a typical high-speed local area data transmission environment. The results showed us a comparable signal-to-interference improvement over the conventional multitone equalization scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.279-285
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• 2003

In this paper we propose a blind decision feedback equalizer (DFE) that is characterized by the fact that it does not require channel estimation. Because the output of the optimized multistep prediction error filter (PEF) can be represented as a product of the channel partial impulse response and the transmitted sequence, a backward multistep PEF can be used as the blind DFE feedforward filter (FFF). The corresponding feedback filter (FBF) is obtained from the symbol -rate partial channel impulse response. The proposed algorithm has several advantages over existing blind channel estimation techniques, including stable performance without the necessity of exact channel order estimation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.2
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• pp.299-306
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• 2011

In this paper, a blind adaptive equalization algorithm based on PDF-distance minimization and a set of Delta functions is introduced and its superior robustness against impulsive noise and multipath characteristics of underwater communication channels is proved. The conventional CMA based on MSE has shown to be incapable of coping with impulsive noise, and correntropy blind algorithm has also revealed to yield not satisfying performance for the mission. On the other hand, the blind adaptive equalization algorithm based on PDF-distance minimization and a set of Delta functions has been proved to solve effectively the problem of impulsive noise and multipath characteristics of underwater communication channels through theoretical and simulation analysis.

• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.394-402
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• 2015

CMA (Constant Modulus Algorithm) is one of the well-known algorithms in blind acoustic channel equalization. Generally, CMA converges slowly and the speed of convergence is dependent on a step-size in the CMA procedure. Many researches have tried to speed up the convergence speed by applying a variable step-size to CMA, e.g. the orthogonal projection CMA and algebraic optimal step-size CMA. In this paper, we summarize these two algorithms, and we propose a new CMA with improved convergence performance. The improvement comes from the switching between the orthogonal projection CMA and algebraic optimal step-size CMA. In simulation results, we show the performance improvement in the time invariant channels as well as in time varying channel.