• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
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• pp.389-392
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• 2002

This paper propose a new GSC (Generalized Sidelobe Canceller) structure, called HFLMS-GSC. The number of complex multiplication required is reduced to one half compared to FLMS-GSC. The simulation results show that mean square error converging and jamming signal removing characteristics are almost the same compared to FLMS-GSC, although the complexity is reduced significantly. As a result, the proposed structure is good for real time implementation, since it has low complexity compared to previous GSC structures.

• 한국산업정보학회논문지
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• 제12권3호
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• pp.19-30
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• 2007

본 논문은 광대역 무선 액세스를 위하여 다중 수신안테나를 갖는 OFDMA 시스템의 타이밍 딜레이 추정기의 구현 복잡도를 낮추는 방안을 제안한다. 타이밍 딜레이 값을 추정하기 위해 각 수신안테나 별로 푸리에 연산과 역 푸리에 연산을 수행하므로, 다중 수신안테나를 사용하는 경우에는 계산 복잡도가 큰 단점이 있다. 먼저, 각 안테나에 수신된 레인징 심볼의 위상을 회전시키는 구조를 제안함으로써 각 안테나의 역 푸리에 연산을 제거하여 구현 복잡도를 크게 개선할 수 있다. 둘째로, N점/M구간 푸리에 연산을 수행하고 시간 대역 평균 전력 추정기 대신 주파수 대역 평균 전력 추정기를 포함한 구조를 제안함으로써 성능 저하 없이 복잡도를 크게 낮출 수 있다. 기존 방식에 대하여 제안된 두 가지 구조의 복잡도 개선량을 보여주고, 시뮬레이션 결과를 통해 성능 비교를 실시한다.

• 전기학회논문지
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• 제60권3호
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• pp.612-619
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• 2011

Satellite navigation system such as GPS is becoming more important infrastructure for positioning, navigation and timing. But satellite navigation system is vulnerable to interferences because of the low received power, complementary navigation system such as eLoran is needed. In order to develop eLoran/GPS navigation system, integrated eLoran/GPS navigation algorithm is necessary. In this paper, new integrated eLoran/GPS navigation algorithm is proposed. It combines the position domain integration and the range domain integration to get accurate position with less computational burden. Also an eLoran/GPS evaluation platform is designed and performance evaluation of the proposed algorithm using the evaluation platform is given. The proposed algorithm gives an accuracy of the range domain integration with a computational load of the position domain integration.

• 전자공학회논문지B
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• 제33B권11호
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• pp.62-71
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• 1996

In this paper we propose a method that improves PSNR at low bit rate and reduces computational complexity in fractal image coding based on discrete wavelet transform. The proposed method, which uses significant coefficient tree, improves PSNR of the reconstructed image and reduces computational comlexity of mapping domain block onto range block by matching only the significant coefficients of range block to coefficients of domain block. Also, the proposed method reduces error propagation form lower resolution subbands to higher resolution subbands by correcting error of lower resolution subbands. Some experimental results confirm that the proposed method reduces encoding and decoding time significantly and has fine reconstructed images having no blocking effect and clear edges at low bit rate.

• Journal of Communications and Networks
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• 제16권4호
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• pp.447-457
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• 2014

In this paper, we investigate the frequency domain channel estimation for multiple-input multiple-output (MIMO) single-carrier frequency-division multiple-access (SC-FDMA) systems. In MIMO SC-FDMA, code-division multiplexed (CDM) pilots such as cyclic-shifted Zadoff-Chu sequences have been adopted for channel estimation. However, most frequency domain channel estimation schemes were developed based on frequency-division multiplexing of pilots. We first develop a channel estimation error model by using CDM pilots, and then analyze the mean-square error (MSE) of various minimum MSE (MMSE) frequency domain channel estimation techniques. We show that the cascaded one-dimensional robust MMSE (C1D-RMMSE) technique is complexity-efficient, but it suffers from performance degradation due to the channel correlation mismatch when compared to the two-dimensional MMSE (2D-MMSE) technique. To improve the performance of C1D-RMMSE, we design a robust iterative channel estimation (RITCE) with a frequency replacement (FR) algorithm. After deriving the MSE of iterative channel estimation, we optimize the FR algorithm in terms of the MSE. Then, a low-complexity adaptation method is proposed for practical MIMO SC-FDMA systems, wherein FR is performed according to the reliability of the data estimates. Simulation results show that the proposed RITCE technique effectively improves the performance of C1D-RMMSE, thus providing a better performance-complexity tradeoff than 2D-MMSE.

• ETRI Journal
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• 제37권1호
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• pp.32-42
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• 2015

Orthogonal frequency-division multiplexing (OFDM) suffers from spectral nulls of frequency-selective fading channels. Linear precoded (LP-) OFDM is an effective method that guarantees symbol detectability by spreading the frequency-domain symbols over the whole spectrum. This paper proposes a computationally efficient and low-cost implementation for discrete Hartley transform (DHT) precoded OFDM systems. Compared to conventional DHT-OFDM systems, at the transmitter, both the DHT and the inverse discrete Fourier transform are replaced by a one-level butterfly structure that involves only one addition per symbol to generate the time-domain DHT-OFDM signal. At the receiver, only the DHT is required to recover the distorted signal with a single-tap equalizer in contrast to both the DHT and the DFT in the conventional DHT-OFDM. Theoretical analysis of DHT-OFDM with linear equalizers is presented and confirmed by numerical simulation. It is shown that the proposed DHT-OFDM system achieves similar performance when compared to other LP-OFDMs but exhibits a lower implementation complexity and peak-to-average power ratio.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2004년도 춘계종합학술대회
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• pp.136-139
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• 2004

본 논문은 MPEG-2 동영상의 축소전환 시 IDCT기법을 이용한 임베디드 시스템 구현에 대한 연구이다 일반적으로 압축된 MPEG-2 동영상의 축소전환은 입력 비트열을 완전 복호화한 후, 저대역 필터링과 서브 샘플링을 수행하는 것이다. 그러나 이 방법은 큰 메모리와 많은 계산량을 요구하는 단점이 있다. 최근 이러한 문제점을 해결하기 위해 DCT 영역에 축소 전환하는 방법이 제안되었다. 이 방법은 고해상도 프레임 메모리의 1/4만을 요구한다. 이는 완전히 축소된 영상이 프레임 메모리에 저장되기 때문이다 그러나 이로 인한 필도 정보의 손실은 움직임 보상 단계에서 심각한 오차를 일으키며 영상 화질을 저하시킨다. 임베디드 시스템에서는 동영상의 화질을 유지하며 계산량이 적은 축소 변환 기법이 필요하다. 공간적인 영역에서 축소 변환 방법과 주파수 영역에서의 축소 변환 기법을 임베디드 시스템에 적용하였을 때 동영상의 프레임 속도에 대하여 비교하여 보았다. 주파수 영역에서 축소 변환을 수행하였을 때 평균 29 frame/sec로 주파수 영역에서의 변환 기법이 25% 우수하였다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제16권11호
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• pp.3761-3779
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• 2022

The performance of existing recognition algorithms for binary phase shift keying (BPSK) and quadrature phase shift keying (QPSK) signals degrade under conditions of low signal-to-noise ratios (SNR). Hence, a novel recognition algorithm based on features in the graph domain is proposed in this study. First, the power spectrum of the squared candidate signal is truncated by a rectangular window. Thereafter, the graph representation of the truncated spectrum is obtained via normalization, quantization, and edge construction. Based on the analysis of the connectivity difference of the graphs under different hypotheses, the sum of degree (SD) of the graphs is utilized as a discriminate feature to classify BPSK and QPSK signals. Moreover, we prove that the SD is a Schur-concave function with respect to the probability vector of the vertices (PVV). Extensive simulations confirm the effectiveness of the proposed algorithm, and its superiority to the listed model-driven-based (MDB) algorithms in terms of recognition performance under low SNRs and computational complexity. As it is confirmed that the proposed method reduces the computational complexity of existing graph-based algorithms, it can be applied in modulation recognition of radar or communication signals in real-time processing, and does not require any prior knowledge about the training sets, channel coefficients, or noise power.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2009년도 정보 및 제어 심포지움 논문집
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• pp.37-39
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• 2009

In the research of speech recognition, locating the beginning and end of a speech utterance in a background of noise is of great importance. Since the background noise presenting to record will introduce disturbance while we just want to get the stationary parameters to represent the corresponding speech section, in particular, a major source of error in automatic recognition system of isolated words is the inaccurate detection of beginning and ending boundaries of test and reference templates, thus we must find potent method to remove the unnecessary regions of a speech signal. The conventional methods for speech endpoint detection are based on two simple time-domain measurements - short-time energy, and short-time zero-crossing rate, which couldn't guarantee the precise results if in the low signal-to-noise ratio environments. This paper proposes a novel approach that finds the Lyapunov exponent of time-domain waveform. This proposed method has no use for obtaining the frequency-domain parameters for endpoint detection process, e.g. Mel-Scale Features, which have been introduced in other paper. Comparing with the conventional methods based on short-time energy and short-time zero-crossing rate, the novel approach based on time-domain Lyapunov Exponents(LEs) is low complexity and suitable for Digital Isolated Word Recognition System.

• 한국전자파학회논문지
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• 제16권10호
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• pp.965-972
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• 2005

본 연구에서는 DTV, HDTV 서비스를 동시에 제공하는 차세대 위성 방송시스템의 표준안인 DVB-S2에서 채널 부호화 알고리즘으로 채택한 LDPC 부호의 복호 알고리즘에 대해 연구를 하였다. 샤논의 한계에 근접하기 위해서는 큰 블록 사이즈의 LDPC 부호어 길이와 많은 반복 횟수를 요구한다. 이는 많은 계산량을 요구하며, 그리고 이에 따른 전력 소비량(power consumption)을 야기시키므로 본 논문에서는 세 가지 형태의 low complexity LDPC 복호 알고리즘을 제시한다. 첫째로 큰 블록 사이즈와 많은 반복 회수는 많은 계산량과 power 소모량을 요구하므로 성능 손실 없이 반복 횟수를 줄일 수 있는 SUBSET 방법을 이용한 복호 알고리즘, 둘째로 early stop 알고리즘에 대해 연구하였고, 셋째로 비트 노드 계산과 체크 노드 계산시 일정한 신뢰도 값보다 크면 다음 반복시 계산을 하지 않는 early detection 알고리즘에 대해 연구하였다.