• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.130-135
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• 2006

There have been experiments dealing with the possibility of the actualization of the ANC system by means of operating the DSP adaptation filter. This filter is composed of various filters(including X-LMS algorithm, Filter-U algorithm, and Full-Feedback-Filter-U algorithm) that use ventilation fans and loudspeakers as a primary source in a circular duct as an experimental device. In this operation, the ANC system using the X - LMS algorithm was found to be more effective in reducing noise than without such system. When applying the input signal of the DSP board Full Feedback-Filtered-U algorithm system while having in mind that the additionally installed second control signal was gone through feedback and mixed into the detection microphone installed near the ventilation fan when using the first ventilation fan, the system was not emitted, but maintained stable during the operation of the control filter. At this point, noise tended to decrease at a maximum of l0dB compared to other algorithms at the frequency band of 170-250Hz.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.7
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• pp.1575-1581
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• 2003

As a society has progressed rapidly toward a highly advanced digital information age, a multimedia communication service for acquisition, transmission and storage of image data as well as voice has being commercialized externally and internally. However, in the process of digitalization or transmission of data, noise is generated by several causes, and researches for eliminating those noises have been continued until now. There were the existing FFT(fast fourier transform) and STFT(short time fourier transform) for removing noise but it's impossible to know information about time and time-frequency localization capabilities has conflictive relationship. Therefore, for overcoming these limits, wavelet transform which is presented as a new technique of signal processing field is being applied in many fields recently. Because it has time-frequency localization capabilities it's Possible for multiresolution analysis as well as easy to analyze various signal. And when two wavelet base were designed to form Hilbert transform pair, wavelet pair provide superior performance than the existing DWT(discrete wavelet transform) in data characteristic detection. Therefore in this parer, we removed impulse noise by using adaptive-length median filter and two dyadic wavelet base which is designed by truncated coefficient vector.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.7
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• pp.1424-1428
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• 2004

In this paper, CMOS analog filters for use in the Analog front End of digital subscriber loop(DSL) chip set are proposed. Designed filters contain receiver continuous-time filters which are composed of lowpass and highpass functions. And their cutoff frequency are 138H1z and 1.1MHz respectively. A low-voltage gm-c integrator is improved and used to design filters. Desisned filters are verified by HSPICE simulation with the 0.25${\mu}m$ CMOS n-well parameter.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.70-76
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• 1996

In the case of an image corrupted with mixed noise, conventional MMSE filter can not remove such a mixed noise properly, because the impulse moise cause a certain bias of the minimum mean-square error estimate at regions close to outliers. In this paper, we proposed the new method or removal of mixed noise by combining MMSE filtering structure with local multi-windowing method according to directions and with ranked-order method. As a result, the improvement of the image quality with the proposed was obtained between about 9.7 and 35.2 times in the sense of NMSE(normalized mean square errors) evaluation than that of MMSE filter. Also, we could obtain the enhanced image in the mixed noisy image from visual and quantitative aspect.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.5 no.2
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• pp.68-73
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• 2012

This paper is an analysis the performance of CDMA system using array antenna beamforming technique in wireless channels. Adaptive array beamforming antenna technique combine receive signal amplitude with phase in array antenna element, and can be incremental spatial filter function a direction of arrival signal using weight value. Through simulation, in this paper, we were an analysis to compare bit error rate of forward and backward channels using array antenna beamforming technique in order to interference signal decrease of CDMA fading enviroment. The result simulation, we get spatial diversity effect by using array antenna system, and improved the performance to MAI interference decrease.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1329-1338
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• 1999

In this paper, an efficient signal interference control technique to improve the convergence speed of LMS algorithm is introduced. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. According as the step-size parameter $\mu$ is increased, the rate of convergence of the algorithm is controlled. Also, a increase in the step-size parameter $\mu$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the mean squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• Proceedings of the KIEE Conference
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• 1999.11c
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• pp.760-762
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• 1999

본 논문에서는 기존의 적응필터인 LMS(Least Mean Square)와 RLS(Recursive Least Square)의 수렴속도의 향상과 안정성을 개선하기 위한 방안을 제안하였다. 제안된 알고리즘은 기존의 시간영역 LMS 알고리즘보다 상당히 빠른 수렴속도를 보일 수 있도록 설계하였다. RLS 알고리즘는 역행렬연산으로 인한 연산량이 많고 자기상관행렬이 positive definite 특성을 잃어버릴 경우 시스템이 수치적으로 불안정하게 되어 발산하는 단점이 있다. 이런한 단점을 보완하기 위해 제안된 알고리즘을 사용하였다. 기존의 알고리즘은 전력 정규화 과정에서 입력신호의 변환이 백색화가 완전히 이루어지지 않게 되어 자기상관행렬이 순수한 대각행렬이 되지 않는 단점을 지니고 있으나, 본 연구에서는 이러한 대각화 과정에서 좀더 많은 정보를 포함하도록 설계하였다. 아울러 제안된 알고리즘을 적응 등화기에 적용하여 수렴속도가 개선됨을 검증하였다.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.20-28
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• 1995

This paper studied adaptive blind equalizer which belong to Bussgang type. It is well known that blind equalizer performs equalization without using a training sequence. Especially, this paper concentrated on real time processing of them. The channel characteristic was obtained from measurements taken in a real urban multipath environment. A T/2 fractionally-spaced equalizer was used at the receiving end. Our computer simulations demonstrated that Stop and Go, Benveniste-Goursat, and optimal Bussgang algorithms have relatively low MSE property. CMA shows faster convergence property than any other of Bussgang type algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.1
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• pp.11-17
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• 2009

This paper deals with the estimation performance DoA (Direction of Arrival) using the 3D adaptive array antenna in W-CDMA signal. For this, we proposed the 5 types of 3D array antenna, and appied for the MUSIC in order to the estimation of DoA algorithm for that antenna, commonly, then the DoA estimation error was used for the estimation performance. In the mobile communication and radio positioning service, performing the spatial filtering after the DoA estimation in array antenna, the quality of receiving signal can improve by the nulling or minimization of interfering signal which is from the undesired direction and the forming of beam which is from the desired direction. The result of DoA estimation and the DoA estimation error by varying the signal to noise ration and the number of interfering signal and power of each type antenna was calculated by computer simulation. As a result of simulation, the other propose antenna has good performance excluding the stack 3D antenna in DoA estimation and the Curved-B type antenna has more superior performance when increasing the number of interfering signal and power in the DoA estimation error.

• Journal of Advanced Navigation Technology
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• v.6 no.2
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• pp.138-150
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• 2002

This paper addresses the problem of combining adaptive polarization processing and space-time processing for further performance improvement of radar target detection in clutter and Jammer environments. Since the most straightforward cascade combinations have quite limited performance improvement potentials, we focus on the development of adaptive processing in the joint polarization-space-time domain. Unlike a direct extension of some existing space-time processing algorithms to the joint domain, the processing algorithm developed in this paper does not need a potentially costly polarization filter bank to cover the unknown target polarization parameter. The performance of the new algorithm is derived and evaluated in terms of the probability of detection and the probability of false alarm, and it is compared with other algorithms that do not utilize the polarization information or assume that the target polarization is known.