• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2010년도 추계학술대회 논문집
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• pp.543-543
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• 2010

The objective of this research is estimating the location of interested sound source by using the similarity between a beamformer output in time domain and the candidate signal. The waveform of beamformer output at the location of sound source is similar with the waveform emitted by that source. To estimate the location of sound source by using this feature, we define quantified similarity between candidate signal and beamformer output. The candidate signal describes the signal which is generated by interested source. In this paper, similarity is defined by four methods. The two methods use time vector comparison, and the other two methods use time-frequency map or linear prediction coefficients. To figure out the results and performance of localization by using similarities, we demonstrate two conditions. The one is when two pure tone sources exist and the other condition is when several bird sounds exist. As a consequence, inner product with two time-vectors and structural similarity with spectrograms can estimate the locations of interest sound source.

• 전자공학회논문지B
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• 제30B권12호
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yields more accurate detection of the start point of the NET signal.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1992년도 학술논문발표회 논문집 제11권 1호
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1992년도 한국자동제어학술회의논문집(국제학술편); KOEX, Seoul; 19-21 Oct. 1992
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• pp.255-260
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• 1992

It is reported on the methodology of signal detection in noise which is based on a comparison of statistical parameters of observation sample from region of frequency-time noise space where a signal may be present and observation sample from region of this noise space and it is known a priori about the latter that the signal is absent in this region.

• 대한전기학회논문지:시스템및제어부문D
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• 제53권12호
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• pp.820-826
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• 2004

In this paper, studied error of various signal interpolation methods in vibration signal analysis with digital order tracking. Because, interpolation errors are related with sampling rate and amount of calculation. Appled Signal interpolation methods are Lagrange, Newton and Cubic-spline. This paper proposed more proper interpolation method. Also, we suggest guideline for adaptive application of signal interpolation methods with Calculated results.

• 전기학회논문지
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• 제65권9호
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• pp.1583-1589
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• 2016

This paper proposes an improved spread spectrum time domain reflectometry (ISSTDR) using time-frequency correlation and reference signal elimination method in order to have more accurate fault determination and location detection than conventional (SSTDR) despite increased signal attenuation due to the long distance to cable fault location. The proposed method has a two-step process: the first step is to detect a peak location of the reference signal using time-frequency correlation analysis, and the second step is to detect a peak location of the correlation coefficient of the reflected signal by removing the reference signal. The proposed method was validated through comparison with existing SSTDR methods in open-and short-circuit fault detection experiments of low voltage power cables. The experimental results showed that the proposed method can detect correlation coefficients at fault locations accurately despite reflected signal attenuation so that cable faults can be detected more accurately and clearly in comparison to existing methods.

• 한국위성정보통신학회논문지
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• 제10권4호
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• pp.136-140
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• 2015

본 논문에서는 적응형 배열 안테나에 적용되는 대표적 항재밍 기술로 알려진 STAP 및 SFAP 신호처리 방식의 시뮬레이션 성능을 비교 분석하였다. 시뮬레이션을 위해 두 방식의 가중 벡터(weighting vector)를 추정하는 방법으로 공통적으로 MMSE(Minimum Mean Square Error) 알고리즘을 사용하여 다양한 재밍환경에서 시뮬레이션을 통한 성능을 제시하였다. 특히, 재밍 전력 J/S(Jamming to Signal Power Ratio)에 따른 STAP 및 SFAP 성능 비교 분석, 신호대역에 대한 재밍 대역의 비율에 따른 성능 비교 분석 및 두 방식간 BER 성능을 비교하여 재밍 환경에 따른 항재밍 성능을 분석하였다.

• 한국통신학회논문지
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• 제38A권11호
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• pp.973-975
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• 2013

본 논문에서는 미약신호 수신환경, 고감도신호 수신환경, 스펙트럼 일치형 전파방해신호 수신환경 하에서 상용 GPS 수신기의 반송파 신호 전력 대 잡음 전력비 추정 성능을 모의실험을 통해 보이고, 이를 분석한다.

• 한국통신학회논문지
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• 제10권2호
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• pp.82-87
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• 1985

一定한 펄스反復周波數로 動作하는 MTI信號處理器에 대하여 MTI필터나 積分器의 傳達函數가 주어졌을 때 地上클러터에 대한 性能 比較 方法을 考察하였다. MTI信號處理器는 橫斷필터(transversal filter)로, 클러터의 電力密度스펙트럼은 Gaussian形態로 모델화하였고, MTI信號處理器에 대한 MTI改善因子(Improvement Factor)를 算出하여 比較하였다. MTI필터 및 Hanning加重函數를 使用한 積分器와 이들의 從屬接續인 경우에 대하여 正規化된 스펙트럼 너비에 대한 MTI改善因子의 變化를 例示하였다.

• 한국초전도학회:학술대회논문집
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• 한국초전도학회 2008년도 High Temperature Superconductivity Vol.XVIII
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• pp.54-54
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• 2008