• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.7
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• pp.1407-1413
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• 1994

To obtain accurate target information in a radar system, clutter or interference signals must first be effectively removed for target detection. In this paper, the signal is projected onto a constrained orthogonal subspace, so that a minimum variance optimal detector is transformed into an unconstrained detector. The proposed algorithm is equivalent to the conventional optimal detector algorithm, and th algorithm structure shows that the Gram-Schmidt orthogonalization can be achieved to obtain the fast convergence. The performance of the proposed method was observed by simulation experiments.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.10a
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• pp.157-160
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• 2000

다중 워터마킹은 하나의 컨텐츠에 2개 이상의 다른 워터마크를 삽입하는 것으로 각 워터마크는 유일한 키로 정확하게 추출할 수 있어야 한다. 대역확산법(spread spectrum)을 이용한 다중 워터마킹은 삽입되는 워터마크간의 직교성(orthogonality)이 제공되어야 삽입된 워터마크들의 정확한 추출이 가능하다. 랜덤계열과 Hadamard 계열을 이용한 기존의 방법은 직교성이 보장되지 않기 때문에 워터마크가 정확히 추출되지 않은 문제가 발생한다. 따라서, 본 논문에서는 랜덤계열들을 Gram-schmidt 직교화 과정을 이용하여 직교화시킨 후, 직교화 된 랜덤계열로 워터마크를 삽입하여 정확한 추출이 가능한 방법을 제안하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.9C
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• pp.743-748
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• 2008

This paper makes a research on the equivalent Wiener-Hopf equation which can obtain the coefficient of TDL filter on orthogonal input signal in terms of mean square error. Using this result, we can present the coefficient and error of TDL filter directly without inverse orthogonalization process on orthogonal input signal. We make a theoretical analysis on MMSE and show an Wiener-Hopf solution and the proposed equivalent one in mathematical example simultaneously.

• Proceedings of the KOSOMBE Conference
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• v.1997 no.11
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• pp.91-94
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• 1997

In this paper, pseudo-random ternary sequence (PRTS) was used to investigate the cardiovascular and respiratory responses to hypoxia and hypercapnia. The actual input or this study was the changes in inhaled oxygen and carbon dioxide concentrations. It is hard to randomly change the concentration within a given breath. Since PRTS has almost the same statistical properties as Gaussian white noise, plus it is physically realizable, PRTS is used for this study. Using PRTS and Volterra kernels by Gram-Schmidt orthogonalization procedure, the cardiovascular and respiratory responses to hypoxia and hypercapnia were analyzed.

• ETRI Journal
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• v.31 no.2
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• pp.193-200
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• 2009

In this paper, we study the performance of subspace-based multiuser detection techniques for multicarrier code-division multiple access (MC-CDMA) systems. We propose an improvement in the PASTd algorithm by cascading it with the classical Gram-Schmidt procedure to orthonormalize the eigenvectors after their sequential extraction. The tracking of signal subspace using this algorithm, which we call OPASTd, has a faster convergence as the eigenvectors are orthonormalized at each discrete time sample. This improved PASTd algorithm is then used to implement the subspace blind adaptive multiuser detection for MC-CDMA. We also show that, for multiuser detection, the complexity of the proposed scheme is lower than that of many other orthogonalization schemes found in the literature. Extensive simulation results are presented and discussed to demonstrate the performance of the proposed scheme.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.29B no.1
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• pp.34-42
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• 1992

This paper presents two fast least-squares lattice algorithms for adaptive nonlinear filters equipped with bilinear system models. The lattice filters perform a Gram-Schmidt orthogonalization of the input data and have very good numerical properties. Furthermore, the computational complexity of the algorithms is an order of magnitude snaller than previously algorithm is an order of magnitude smaller than previously available methods. The first of the two approaches is an equation error algorithm that uses the measured desired response signal directly to comprte the adaptive filter outputs. This method is conceptually very simple`however, it will result in biased system models in the presence of measurement noise. The second approach is an approximate least-squares output error solution. In this case, the past samples of the output of the adaptive system itself are used to produce the filter output at the current time. Results of several experiments that demonstrate and compare the properties of the adaptive bilinear filters are also presented in this paper. These results indicate that the output error algorithm is less sensitive to output measurement noise than the squation error method.

• The Journal of the Acoustical Society of Korea
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• v.7 no.4
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• pp.31-53
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• 1988

In this work we study extensively the structures and performance characteristics of the block least mean-square (BLMS) adaptive digital filters (ADF's) that can be realized efficiently using the fast Fourier transform (FFT). The weights of a BLMS ADF realized using the FFT can be adjusted either in the time domain or in the frequency domain, leading to the time-domain BLMS(TBLMS) algorithm or the frequency-domain BLMS (FBLMS) algorithm, respectively. In Part Ⅰof the paper, we first present new results on the overlap-add realization and the number-theoretic transform realization of the FBLMS ADF's. Then, we study how we can incorporate the concept of different frequency-weighting on the error signals and the self-orthogonalization of weight adjustment in the FBLMS ADF's , and also in the TBLMS ADF's. As a result, we show that the TBLMS ADF can also be made to have the same fast convergence speed as that of the self-orthogonalizing FBLMS ADF. Next, based on the properties of the sectioning operations in weight adjustment, we discuss unconstrained FBLMS algorithms that can reduce two FFT operations both for the overlap-save and overlap-add realizations. Finally, we investigate by computer simulation the effects of different parameter values and different algorithms on the convergence behaviors of the FBLMS and TBLMS ADF's. In Part Ⅱ of the paper, we will analyze the convergence characteristics of the TBLMS and FBLMS ADF's.

• Journal of applied mathematics & informatics
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• v.26 no.1_2
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• pp.213-222
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• 2008

Iterative methods are often suitable for solving least squares problems min$||Ax-b||_2$, where A $\epsilon\;\mathbb{R}^{m{\times}n}$ is large and sparse. The well known LSQR algorithm is among the iterative methods for solving these problems. A good preconditioner is often needed to speedup the LSQR convergence. In this paper we present the numerical experiments of applying a well known preconditioner for the LSQR algorithm. The preconditioner is based on the $A^T$ A-orthogonalization process which furnishes an incomplete upper-lower factorization of the inverse of the normal matrix $A^T$ A. The main advantage of this preconditioner is that we apply only one of the factors as a right preconditioner for the LSQR algorithm applied to the least squares problem min$||Ax-b||_2$. The preconditioner needs only the sparse matrix-vector product operations and significantly reduces the solution time compared to the unpreconditioned iteration. Finally, some numerical experiments on test matrices from Harwell-Boeing collection are presented to show the robustness and efficiency of this preconditioner.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.3-11
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• 1999

The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5A
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• pp.457-463
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• 2011

This paper proposes theorthogonal reference vectors selection method of the subspace interference alignment. The proposed method selects multiple orthogonal reference vectors instead of using one reference vector for all users at the same time. The proposed scheme selects a reference vector which maximizes a sum-rate for a certain cell, generates orthogonal vectors to the previous selected vector and selects the one of orthogonal vectors whose sum rate is maximized for each cell. Larger channel gain and sum-rate than the previous method can be obtained by selection degree of freedom. The computer simulation demonstrates the proposed method gives higher sum-rate compared with that of the previous reference vector selection method.