• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3C
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• pp.279-285
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• 2006

In this study, a new signal model suitable for UWB indoor environments with random angle spread is proposed to estimate the angle-of-arrivals(AOAs) of UWB cluster signals in an UWB wireless communication. A subspace based estimation technique adopted for this model is investigated and the estimates of the AOA and distribution parameter on the received UWB cluster signals are obtained. The proposed model and estimation technique are verified using computer simulations, and the performance of the estimation error is analyzed.

• Proceedings of the KIEE Conference
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• 1992.07a
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• pp.453-456
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• 1992

A Data-based Covariance Matrix(DCM) is introduced in the Eigenvector(EV) method, among subspace methods of estimating multiple sinusoidal frequencies from finite white noisy measurements. It is shown that the EV with the DCM can obtain the true. frequencies from finite noiseless data Some asymptotic results and further improvement on the DCM are also presented mathematically. Monte-carlo simulations are statistically conducted from the view-points of means and standard deviations in the EV's of DCM and Conventional Covariance Matrix(CCM). Simulations show a great promise for using the DCM, particularly for the cases of short data records, closely spaced frequencies and high signal-to-noise ratios.

• Proceedings of the IEEK Conference
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• 2001.06a
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• pp.69-72
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• 2001

Blind equalization of transmission channel is important in communication areas and signal processing applications because it does not need training sequence, nor does it require a priori channel information. Recently, Tong et al. proposed solutions for this problem exploit the diversity induced by antenna array or time oversampling, leading to the second order statistics techniques, fur example, subspace method, prediction error method, and so on. The linear prediction error method is perhaps the most attractive in practice due to the insensitive to blind equalizer length mismatch as well as for its simple adaptive filter implementation. Unfortunately, the previous one-step prediction error method is known to be limited in arbitrary delay. In this paper, we induce the optimal delay, and propose the adaptive blind equalizer with multi-step linear prediction using RLS-type algorithm. Simulation results are presented to demonstrate the proposed algorithm and to compare it with existing algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7A
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• pp.939-946
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• 1999

A new blind multi-user channel estimation algorithm for the mobile communication systems is proposed. The proposed algorithm exploits the second-order statistics of a received signal and the subspace concept, and requires much less computational complexity than the existing algorithms. The algorithm can reduce the comptational load by estimating the physical channels excluding the spreading codes. We formulate the algorithm using the multi-channel model for asynchronous DS-CDMA systems and perform numerical experiments to evaluate the performance of the proposed algorithm.

• Proceedings of the KIPE Conference
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• 2005.07a
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• pp.192-194
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• 2005

Development of nonlinear loads and compensation instruments make PQ(Power Quality) problem into important issue. Few studies by signal processing and pattern classification as NN(Neural Network), Wavelet Transform, and Fuzzy present feature extraction. A lot of Input features make not always good result and they are difficult to make realtime system. Thus, The dimentionality reduction is indispensable process. PCA(Principal Component Analysis) reduces high-dimensional input features onto a lower-dimensional subspace effectively. It will be useful to apply to realtime system and NN.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.4C
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• pp.336-341
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• 2008

In this paper, beamforming algorithm is proposed which can obtain diversity gain in beamforming system that deploy antenna elements with half-wavelength. The proposed algorithm provides beam-pattern using eigen-vectors that span received signal subspace. The criterion to decide optimal rank of eigen-space used for beamforming is also proposed. A beamforming system applied the proposed algorithm shows better performance with diversity gain as getting larger angle spread. This paper provides a description of proposed algorithm with analysis of the performance using various computer simulations.

• Proceedings of the Korean Society for Emotion and Sensibility Conference
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• 1999.11a
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• pp.388-395
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• 1999

본 논문의 목적은 퍼지 엔트로피를 이용하여 비선형신호를 예측하는 것이다. 이 방법은 분할된 여러 부 공간(subspace)에 대해 입력 데이터로부터 퍼지 엔트로피를 이용하여 각각의 규칙에 등급을 정하여 불필요한 제어규칙을 제거하여 바람직한 규칙베이스를 구성하도록 한 것이다. 적용되는 퍼지 신경망의 기본적인 구조는 퍼지 제어기의 규칙베이스와 추론의 과정을 신경회로망을 이용하여 구현하며 퍼지 제어규칙의 매개변수들은 역전파 알고리즘에 의해 적응되어진다. 또한 매개변수의 수를 줄이기 위하여 제어규칙의 결론부의 출력값은 신경망의 가중치로 구성하였다. 결국 퍼지 신경망의 복잡도를 줄일 수 있다. Mackey-Glass 시계열의 예측에 대한 컴퓨터 시뮬레이션을 통하여 본 논문에서 제안한 방법의 효율성을 입증하고, 제안된 방법을 EEG 생리신호 분석에 이용될 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8A
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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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.1
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• pp.15-23
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• 2003

This paper proposes a fast direction of arrival (DOA) estimation algorithm that can rapidly estimate incidence angles of incoming signals using a pseudo covariance matrix. The conventional subspace DOA estimation methods such as MUSIC (multiple signal classification) algorithms need many sample signals to acquire covariance matrix of input signals. Thus, it is difficult to estimate the DOAs of signals because they cannot perform DOA estimation during receiving sample signals. Also if the D0As of signals are changing rapidly, conventional algorithms cannot estimate incidence angles of signals exactly. The proposed algorithm obtains bearing response and directional spectrum after acquiring pseudo covariance matrix of each snapshot. The incidence angles can be exactly estimated by using the bearing response and directional spectrum. The proposed DOA estimation algorithm uses only concurrent snapshot so as to obtain covariance matrix. Compared to conventional DOA estimation methods. The proposed algorithm has an advantage that can estimate DOA of signal rapidly.

• Journal of Korea Multimedia Society
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• v.19 no.7
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• pp.1127-1136
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• 2016

Speech enhancement algorithm plays an important role in numerous speech signal processing applications. Over the last few decades, many algorithms have been studied for speech enhancement. The algorithms are based on spectral subtraction, Wiener filter, and subspace method etc. They have good performance of speech enhancement, but the performance can be deteriorated in specific noises or low SNR environment. In this paper, a new speech enhancement algorithms are proposed based on adaptive noise canceller. And the proposed algorithm improved performance of adaptive noise cancelling using 2-D binary mask. From objective experimental index, it is confirmed that the proposed algorithm is useful and has better performance than recently proposed speech enhancement algorithms.