• Proceedings of the IEEK Conference
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• summer
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• pp.97-102
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• 2004

The very complicated proplem in spatial processing is effects of phading (Multipath and Delay Spread) and co-channel interference (CCI). The phading is one of principal causes, that form inter-symbol interference (ISI). Spatial-Temproral Adaptive Filter (STAF) has been taken as a solution of this problem, because it can suppress both these types of interference. But the performance of STAF exposes some elemental limitations, in which are the slow convergence of adaptive process and computational complexity. The cause of this is that, STAF must treat a large quantity of information in both space and time. The way that master these limitation is a use of Subband Spatial-Temproral Adaptive Filter (SSTAF). SSTAF reduce computational complexity by pruning off samples of signal and thus it lost some information in time. This draw on attennation of output SINR of SSTAF. The article analyse a optimal solution of this problem by introducing SSTAF with nonuniform weighted distribution.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.41-48
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• 2003

This paper presents an indirect adaptive neuro controller using modified chaotic neural networks(MCNN) for nonlinear dynamic system. A modified chaotic neural networks model is presented for simplifying the traditional chaotic neural networks and enforcing dynamic characteristics. A new Dynamic Backpropagation learning method is also developed. The proposed MCNN paradigm is applied to the system identification of a MIMO system and the indirect adaptive neuro controller. The simulation results show good performances, since the MCNN has robust adaptability to nonlinear dynamic system.

• International Journal of Control, Automation, and Systems
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• v.3 no.1
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• pp.137-142
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• 2005

Biomedical signals such as ECG, EMG, and EEG are extremely important in the diagnosis of patients. It is difficult to filter noise from these signals, and errors resulting from filtering can distort a biomedical signal. Existing systems have shown poor performance when complicated noise appears. Adaptive filtering is selected to contend with these defects. Existing adaptive filters can adjust the filter coefficient with the given filter order, but they can produce an unsuitable order in different environments. In order to solve this problem, an optimal adaptive filter with a dynamic structure was designed. Positive experimental results were obtained.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.687-693
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• 2003

In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.4
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• pp.879-885
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• 2004

This paper describes an adaptive signal processing of digital receiver with digital down convertor(DDC). DDC is composed of numerically controlled oscillator(NCO) and digital low pass filler and the received signal is processed by numerical algorithm. The simulation results of digital receiver using the passband sampling technique are presented and we confirmed that the received low IF signal is converted to zero IF by numerically processed DDC. Direction of arrival(DOA) estimation technique using multiple signal classification(MUSIC) algorithm with high resolution is also discussed. We knew that an accurate resolution of DOA depends on the input sampling numbers and antenna element numbers.

• International journal of advanced smart convergence
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• v.9 no.1
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• pp.177-184
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• 2020

In this paper, we studied to direction of arrival (DoA) estimation to use DoA and optimum weight algorithms in coherent interference channels. The DoA algorithm have been considerable attention in signal processing with coherent signals and a limited number of snapshots in a noise and an interference environment. This paper is a proposed method for the desired signal estimation using MUSIC algorithm and adaptive beamforming to compare classical subspace techniques. Also, the proposed method is combined the updated weight value with LCMV beamforming algorithm in adaptive antenna array system for direction of arrival estimation of desired signal. The proposed algorithm can be used with combination to MUSIC algorithm, linearly constrained minimum variance beamforming (LCMV) and the weight value method to accurately desired signal estimation. Through simulation, we compare the proposed method with classical direction of in order to desired signals estimation. We show that the propose method has achieved good resolution performance better that classical direction arrival estimation algorithm. The simulation results show the effectiveness of the proposed method.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.247-250
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• 2003

디지털 2차 적응 IIR 대역 통과 필터의 진폭 특성을 해석, 적응 계수와 입력 신호의 중심 주파수 그리고 극의 각과의 관계를 해석하고, 적응 계수의 경계를, 기존에 발표된 세 알고리즘을 예로 들어, 평가하였다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.121-124
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• 2000

적응 학습 알고리즘으로 가중치를 변화시키는 단층 신경망의 출력부에 Ramp 임계 함수를 적용하여 입력이 zero-mean Gaussian random vector인 경우 가중치의 stationary point를 구하고, 적응 학습 알고리즘을 유도한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8B
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• pp.1518-1531
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• 1999

The reconstructed images from highly compressed MPEG or H.263 data have noticeable image degradations, such as blocking artifacts near the block boundaries, corner outliers at cross points of blocks, and ringing noise near image edges, because the MPEG or H.263 quantizes the transformed coefficients of 8$\times$8 pixel blocks. A post-processing algorithm has been proposed by authors to reduce quantization effects, such as blocking artifacts, corner outliers, and ringing noise, in MPEG-decompressed images. Our signal-adaptive post-processing algorithm reduces the quantization effects adaptively by using both spatial frequency and temporal information extracted from the compressed data. The blocking artifacts are reduced by one-dimensional (1-D) horizontal and vertical low pass filtering (LPF), and the ringing noise is reduced by two-dimensional (2-D) signal-adaptive filtering (SAF). A comparison study of the subjective quality evaluation using modified single stimulus method (MSSM), the objective quality evaluation (PSNR) and the computation complexity analysis between the signal-adaptive post-processing algorithm and the MPEG-4 VM (Verification Model) post-processing algorithm is performed by computer simulation with several MPEG-4 image sequences. According to the comparison study, the subjective image qualities of both algorithms are similar, whereas the PSNR and the comparison complexity analysis of the signal-adaptive post-processing algorithm shows better performance than the VM post-processing algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.3
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• pp.501-507
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• 1999

At the mobile unit for adaptation the propagation environment, it is necessity to adapt very fast the weight coefficient vector of adaptive array antenna In this paper, for the BPSK and BFSK signals with S/I=2, S/N=10 subband adaptive array signal processing method to the linear array antenna using the LMS & the Kalman filter algorithm is proposed. For the 4 elements equidistance linear array antenna systems LMS and Kalman algorithms with subband adaptive instruction principles using the subband signal processing method are adopted and the computer simulation results to the constant amplitude envelope signals such as BPSK or BFSK can be seen that the convergence characteristics of directional patterns and the signal following characteristics are more fast and stable.