• Proceedings of the Korean Nuclear Society Conference
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• 1998.05a
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• pp.303-308
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• 1998

SMART 계측제어계통 측정신호의 신뢰성을 높이기 위한 실시간 신호검증알고리듬을 개발하였다. 개발된 알고리듬은 선행고장검출행렬, 아날로그 신호용 다중성 기법, 접촉신호용 논리표 알고리듬, 주파수 신호용 다중성 기법 그리고 아날로그 센서 경증을 위한 통계적 모듈의 5개 모듈로 구성되어 있다. 선행고장검출행렬은 측정 신호 중에서 고장의 가능성이 있는 신호를 추출하여 선정된 신호만을 적절한 알고리듬으로 검증하도록 함으로써 전체적인 수행시간을 감소시킨다. 아날로그 신호검증 모듈은 아날로그 측정신호에 대한 물리적/해석적 다중성에 입각하여 고장신호의 크기, 위치를 검출하며, 접촉신호 검증 모듈은 접촉신호들간의 논리값을 비교하여 발생 불가능한 논리값을 가지는 신호를 고장신호로 검출한다. 주파수신호는 아날로그 신호와 유사한 기법을 구현하였으며, 통계적 모듈은 아날로그 센서 자체의 물리적 건전성을 검사하는 모듈이다. 현재 SMART의 설계가 확정되어 있지 않으므로 개발된 신호검증알고리듬을 시험하기 위해서 여러 주요 공정변수가 표현되는 상용 원자로의 냉각재계통을 대상으로 검증 알고리듬을 구현하였으며, 운전모사기로 모사된 신호를 이용하여 개발된 신호검증알고리듬을 시험하였다. 시험결과 각 모듈별로 적절히 고장을 검출함을 보였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.159-162
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• 2001

음장의 공간 음향적인 특성에 영향을 받은 음성신호를 원래 신호로 복원하기 위해서 본 논문에서는 MPSV (Minimum Phase Space Volume) 방법 을 도입한다 MPSV 방법 은 신호를 복원하기 위해 원래 신호의 어떠한 사전 정보나 가정을 필요로 하지 않고 그 신호의 비선형적인 동적 특이성만을 이용하는 블라인드 디콘볼루션 (Blind deconvolution) 방법이다. 또한, 이 방법을 이용하여 원래 신호를 복원하는 동시에 음장의 충격응답과 같은 시스템 특성까지도 유추가 가능하다.

• Proceedings of the KIEE Conference
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• 2006.07c
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• pp.1333-1334
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• 2006

중, 저압 배전 시스템을 통신망으로 사용하는 전력선 통신(PLC, Power Line Communication)에서 신호결합은 하드웨어 측면에서 매우 중요하다. 본 논문에서는 2-40MHz의 통신 신호를 수백 A의 전류가 흐르는 배전선으로 보내고 추출하는 유도성 신호결합장치와 통신신호 손실에 대하여 조사하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.335-338
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• 2001

잔향음은 시변동성이 존재하는 유동성 경계면 잔향음(해수면, 체적)과 시변동성이 존재하지 않는 고정 경계면 잔향음(해저면)으로 분류된다. 그러나 고정 경계면 잔향음으로 알려진 해저면 잔향음에서도 단주기적 시변동성이 존재하고 있음이 여러 실측자료에서 관측되고 있다. 본 연구는 시변동성의 원인을 파악하고자 실험실에서 부유성 퇴적물의 농토에 따른 후방산란 신호를 측정하였다. 또한 동해에서 측정된 시간에 따른 잔향음신호(80kHz)와 ADCP(4.2MHz) 자료를 비교하여 천해에서의 체적 산란체의 변동이 잔향음 신호에 영향을 미칠 수 있음을 확인하였다. 아울러 본 논문에서는 잔향음 신호의 단주기적 시변동성에 의한 잡음 성분을 제거하여 표준화된 잔향음 신호를 획득하기 위한 방법으로 Low Rank Approximation(LRA)을 제안하였다. 이 기법은 특이해 분해(Singular Value Decomposition, SVD)를 수행하여 실측 자료 행렬로부터 고유치(Eigenvalue)과 고유벡터(Eigenvector)를 추출한 후, 추출된 고유치를 제한적으로 사용하여 근사화 하는 기법으로 시변동성 신호를 제거하는데 효율적인 방법이다.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.35T no.3
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• pp.37-45
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• 1998

In this paper, the different frequency component and time informations from an EMI signal are extracted simultaneously using a wavelet transform and the results of transform in the time and frequency domain are analyzed. Frequencies are extracted from the EMI signal by performing the multiresolution analysis using the Daubechies-4 filter coefficients and the time information through the results of wavelet transform. We have tried the correlation analysis to evaluate the results of wavelet transform. We have chosen the optimal wavelet function for an object signal by comparing the transformed results of various wavelet functions and verified the simulation examples of waveform and harmonic analysis using a wavelet transform. We have proved the denoising effect to the EMI signal using the soft thresholding technique.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.20 no.6
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• pp.81-86
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• 2006

Routing signal lines in PCB, line shapes would be straight or bent. time-domain and frequency-domain evaluation of the signal property and interference are archived by precise Modeling of differential signal line. Some of CAD tools can extract equivalent circuit model parameters. but it takes a long time and heavy loads. This paper introduces a basic RLC equivalent circuit model parameter extraction technique for bent differential structures using genetic algorithm by this technique, we can model equivalent circuit of bent differential structures more faster.

• Journal of the Korea Convergence Society
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• v.8 no.5
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• pp.13-17
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• 2017

Existing Speech feature extracting method in speech Signal, there are incorrect recognition rates due to incorrect speech which is not clear threshold value. In this article, the modeling method for improving speech recognition performance that combines the feature extraction for speech and silence characteristics normalized to the non-speech. The proposed method is minimized the noise affect, and speech recognition model are convergence of speech signal feature extraction to each speech frame and the silence feature normalization. Also, this method create the original speech signal with energy spectrum similar to entropy, therefore speech noise effects are to receive less of the noise. the performance values are improved in signal to noise ration by the silence feature normalization. We fixed speech and non speech classification standard value in cepstrum For th Performance analysis of the method presented in this paper is showed by comparing the results with CHMM HMM, the recognition rate was improved 2.7%p in the speech dependent and advanced 0.7%p in the speech independent.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.87-92
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• 1996

Transient signals in underwater show several characterisrics, that is, short duration, strong nonstationarity, various types of transient sources, which make it difficult to analyze and classify transient signals. In this paper, the feature vector extraction method for transient SOMAR signals is discussed by applying digital signal processing methods to the analysis of transient signals. A feature vector extraction methods using wavelet transform, which enable us to obtain better recognition rate than automatic classification using the classical method, are proposed. It is confirmed by simulation that the proposed method using wavelet transform performs better than the classical method even with smaller number of feature vectors. Especially, the feature vector extraction method using PR-QMF wavelet transform with the Daubechies coefficients is shown to perform well in noisy environment with easy implementation.

• Journal of Digital Convergence
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• v.11 no.7
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• pp.209-214
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• 2013

Improve the recognition performance of speech recognition systems as a method for recognizing human listening skills were incorporated into the system. In noisy environments by separating the speech signal and noise, select the desired speech signal. but In terms of practical performance of speech recognition systems are factors. According to recognized environmental changes due to noise speech detection is not accurate and learning model does not match. In this paper, to improve the speech recognition feature extraction using gamma tone and learning model using acoustic model was proposed. The proposed method the feature extraction using auditory scene analysis for human auditory perception was reflected In the process of learning models for recognition. For performance evaluation in noisy environments, -10dB, -5dB noise in the signal was performed to remove 3.12dB, 2.04dB SNR improvement in performance was confirmed.

• Journal of the Institute of Convergence Signal Processing
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• v.22 no.2
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• pp.79-84
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• 2021

The most important device of the AGV is the guide sensor, and the typical function of this sensor is high accuracy and extraction of the road. If the accuracy of the guide sensor is low or the sensor device is extracted the wrong track, this causes the problems such as the AGV collision, track-out, the load falling due to AGV swing. In order to improve these problems, this study is proposed a signal processing method of the guide sensor based on multi-maskings and the center of gravity method, and evaluated its performance. As a result, the proposed method showed that the mean error of absolute value is 2.32[mm] and it showed performance improvement of 27[%] than the center of gravity method of existence. Therefore, when the proposed signal processing method is applied, It is thought that the posture control and driving stability of the AGV will be improved.