• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1996.06a
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• pp.3-7
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• 1996

In this paper, the performances of OFDM and MC-CDMA systems in a slowly-varying multipath fading environment is investigated. Time variation of the multipath channel leads to both a change of an optimal coefficient in one-tap equalizer and a loss of subchannel orthogornality, resulting in significant performance loss. A new simple one-tap equalizer which can reduce the effect of slowly time-varying multipath channel is proposed by taking into account time-variation of multipath profile and modifying the previous equalization techniques. It is demonstrated by computer simulation that the performances of OFDM and MC-CDMA systems can be improved by using the proposed one-tap equalizers when the multipath channel is slowly varying.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.3
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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.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.66 no.4
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• pp.247-251
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• 2017

This paper proposes a low power transmission technique for single-carrier modulation with frequency domain equalization. As time domain signals and frequency domain signals have unique corresponding functions, inserting zeros after each symbol causes a repetition in other domain, so maximal ratio combining technique using repetitive transmission can be applied in the frequency domain. In this paper, we configure transmit signals to insert zeros after each symbols for single-carrier modulation with frequency domain equalization and maximal ratio receive combining block in the receiver structures, propose a structure for transmitter and receiver and show that its performance is better than the traditional algorithm by simulations.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.8-10
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• 2011

컬러는 영상처리 분야에서 중요한 단서로 사용될 수 있는 정보이다. 하지만 실제로 촬영한 영상의 경우에는 빛과 카메라 특성 등 다양한 요소들의 영향으로 인해 이미지 간 컬러 정보의 불일치가 빈번히 일어난다. 따라서 컬러가 다른 여러 장의 영상을 입력 영상으로 사용하는 경우, 입력 영상간 컬러를 동일하게 맞춰 주어야 한다. 이를 수행함에 있어서, 어떠한 이미지를 레퍼런스 이미지로 결정할 것인가는 매우 중요한 문제이다. 이에 본 논문에서는, 히스토그램 등화(histogram equalization) 기법을 이용하여 입력 이미지들의 비용을 결정해줌으로써, 레퍼런스 이미지를 결정하는 방법을 제시한다. 스테레오 매칭을 통해 다양한 밝기의 입력 영상에서 가장 좋은 결과를 얻을 수 있는 레퍼런스 이미지를 결정할 수 있음을 보였다.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.7
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• pp.65-75
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• 1996

The probability distributio function (pdf) of transmitted symbols should be asymmectric for recovering the received data in the 3rd order blind equalizer system. In this paper, we hav edesigned the blind equalizer using symmetric-asymmetric (SA) and asymmetric-symmetric (AS) transforms for less computational complexities and robustness in the noisy environments. The method of SA and AS transform was peformed using natural logarithmic operation. This paper proves that the proposed in this paper can be performed as real time operation. This paper proves that the compression factor k has no effet on transmitted symbols. Also 3rd order equalization method proposed in this paper can be performed as real time operation. As a result of computer simulatio, the computational complexity of proposed algorithm is reduced to be an half of 4th order method and MSE is enhanced as 10dB at the case of 4-PAM and 15dB at the case of 8-PAM respectivley. Conclusively, we have found that 3rd order blind equalizer can be implemented when the pdf of transmitter is asymmetric.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.445-447
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• 2005

MCMA(modified constant modulus algorithm) accomplishes blind equalization and carrier phase recovery simultaneously. But, the error level of MCMA is not zero when the equalizer converges completely. Because the MCMA uses a special signal point instead of a original signal point. MCMA-DO(decision-directed) improves the steady-state performance but the performance of equalizer is decided by switching time between the MCMA and the DD. In this paper, according to the residual ISI(intersymbol interference) of the equalizer output, the most suitable switching time is decided automatically.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.279-280
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• 2008

In order to improve the performance and simplify the structure of the conventional detectors in high density magnetic channels, a new equalizer based on bilinear recursive polynomial (BRP) models, which uses the previously estimated sequence, is proposed. The performance is compared with the conventional equalizers and the maximum likelihood sequence detection (MLSD) bound.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.1C
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• pp.37-46
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• 2003

As Information Security Technology has become rather transparent, wide, and integrated than in part, exclusive, and separated, A necessity of the study about integrating the separated distributed security systems into one module, has grown However, there is no integrated framework which can manage all separate security systems as one integrated one yet. Accordingly, we propose a new policy based network admirustrative model in this paper which can integrate individual security systems and distributed control way into one effectively.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2004.11a
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• pp.123-127
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• 2004

지상파 디지털 TV 방식 중의 하나인 ATSC 8-VSB방식은 전송된 데이터 중 유효 데이터의 비율이 높고 채널이 좋은 경우 동기 및 등화 과정이 간단한 장점이 있지만, 실내수신이나 이동수신 환경에서는 동기 및 등차 과정이 어려운 단점이 있다. 현재 주파수 동기에는 FPLL 알고리즘을 이용하고 있는데, 파일롯의 위치에서의 좌우 스펙트럼의 비대칭 및 채널 환경에 따른 파일롯 신호의 감쇄가 발생하여 양의 주파수 옵셋이 발생했을 때에 비해 음의 주파수 옵셋이 발생했을 때의 동기획득이 어려운 단점이 있다. 본 논문은 파일롯 추출필터와 인접된 필터를 통과한 신호의 파워비율을 이용하여 얻은 제어신호를 이용하여, 기존 FPLL의 동작 범위를 넓힘과 동시에 수렴속도를 빠르게 할 수 있는 방법을 제안하였다.

• Phonetics and Speech Sciences
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• v.4 no.2
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• pp.79-86
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• 2012

In this paper, we propose a novel approach to improve histogram equalization for speaker identification. Our method collects all speech features of UBM training data to make a reference distribution. The ranks of the feature vectors are calculated in the sorted list of the collection of the UBM training data and the test data. We use the ranks to perform order-based histogram equalization. The proposed method improves the accuracy of the speaker recognition system with short utterances. We use four kinds of speech databases to evaluate the proposed speaker recognition system and compare the system with cepstral mean normalization (CMN), mean and variance normalization (MVN), and histogram equalization (HEQ). Our system reduced the relative error rate by 33.3% from the baseline system.