• 한국통신학회논문지
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• 제18권10호
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• pp.1413-1421
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• 1993

제어대상에 제한된 제어 입력 신호에 대하여 좋지 않은 효과를 줄일 수 있고, 이것은 제어 입력 신호의 억제 되어진 부분을 보상하는 보상 루프를 구성함으로서 실행된다. 보상 루프의 보상기 설계와 안정도 조사는 Kalman-Szego-보조정리를 이용하고 그 결과로서 작은 오차신호에 대해서도 제어대상에 대한 제어 입력 신호의 많은 범위를 활용하는 것이 가능하다.

• 전자공학회논문지B
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• 제30B권3호
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• pp.10-21
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• 1993

Bilinear models are attractive for adaptive filtering applications because they can approximate a large class of nonlinear systems adequately, and usually with considerable parsimony in the number of coefficients compared with Volterra models. But bilinear filters have stability problem because they involve nonlinear feedback. Adaptive algorithms for bilinear filters may be diverge and have poor convergence characteristics when input signal is large In this paper, necessary and sufficient condition for mean square stability of bilinear filters for given input signal statistics is briefly described, and the method obtaining the input bound to guarantee the stability of bilinear filters is presented. New RPEM algorithm, which does not diverge and has the superior convergence characteristics compared with the conventional RPEM algorithm when input signal is large, is derived by applying the time-varying Kalman filtering concept to the conventional RPEM algorithm.

• 대한전자공학회논문지SP
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• 제45권2호
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• pp.90-94
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• 2008

적응 필터의 입력 신호의 상관도 (correlation)가 클 경우 LMS 알고리즘의 수렴 속도는 상당히 느려지게 된다. 본 논문에서는 입력 신호의 상관도가 높은 상황에서 수렴 속도를 향상시킬 수 있는 적응 필터링 알고리즘을 제안한다. 입력 신호에 대하여 직교성을 가지도록 변환을 인위적으로 가하여 LMS 알고리즘의 한계를 극복한다. 제안한 알고리즘의 성능 향상은 시스템식별 모델을 통하여 그 수렴 속도의 개선을 확인하며 또한 시변 환경 하에서 적응 필터의 시변 추적 능력을 통해 보여 진다.

• 대한기계학회논문집
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• 제16권9호
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• pp.1722-1727
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• 1992

본 연구에서는 이러한 불확실한 시스템에 대하여 정규화 형태의 추정기를 적 용한 적응 입출력 선형화 제어기의 설계에 대해 연구하였으며, 신호 성장속도(signal growth rates)의 개념을 도입하여 안정성을 해석하였다. 정규화 형태의 추정기를 적 용함으로써 큰 불확실성에 대해 보다 안정한 수렴 특성을 얻을 수 있음을 시뮬레이션 을 통해 보였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -3
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• pp.2004-2007
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• 2002

In the linear combination type stereophonic echo canceller, it is known not to converge the coefficient vector of the adaptive filter to a correct echo path. In this report, we analyze the convergence value of the filter coefficient vector of the stereo echo canceling algorithm using input signals of all channels in relation to this problem. In this analysis, one of the two inputs to the un-known system and adaptive one are assumed to be a delayed and attenuated version of the other signal as a model of the input signal with a strong cross-correlation. As a result, it is shown for the coefficient vectors not to converge to echo paths, and nor to converge to the value which depends on the time delay and the attenuation of the input signal. We show that the computer simulation result are corresponding to our analytical results.

• 한국항해항만학회지
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• 제32권2호
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• pp.133-140
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• 2008

In this paper, a fault detection method for operator failures using the observation technique is proposed. The suggested algorithm is extended using the conventional sensor/actuator fault detection method. First, it is assumed that operator failure affects human work operations, as it is an external input signal. With this assumption, a human work model with operator failure is suggested. Second, an unknown input observer with proportional and integral gains is introduced. The characteristic of this observer of estimating an external signal without an exact input is shown, and the conditions for the detection of an operator failure are proposed. Finally, by simulating the container crane operations, it is verified that the observer can accurately detect an operator failure and estimate its magnitude from the given internal signal.

• 전기학회논문지
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• 제64권1호
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• pp.149-158
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• 2015

This paper proposes a subband filtering technique to the MUSIC and the ESPRIT algorithm for estimating the magnitude and frequency of the harmonics of power signal. In proposed method, the input power signal is decomposed to the odd harmonics and the even harmonics respectively by the filter bank system. The amplitude and the frequency estimation of the decomposed harmonics are carried out using the MUSIC and the ESPRIT method. Subband filtering can reduce the autocorrelation matrix size of input data, and spectrum leakage between adjacent harmonics. Therefore, this subband technique has advantage in computational cost and estimation accuracy compared to fullband MUSIC and ESPRIT. To demonstrate the performance of the method, computer simulations are performed to the synthesized input signal, and experiment results are compared in subband and fullband cases.

• The Journal of the Acoustical Society of Korea
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• 제15권3E호
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• pp.78-85
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• 1996

The set of admissible time-variations in the input signal can be separated into two categories : slow parameter changes and large parameter changes which occur infrequently. A common approach used in the tracking of slowly time-varying parameters is the exponential recursive least-squares(RLS) algorithm. There have been a variety of research works on the error analysis of the exponential RLS algorithm for the slowly time-varying parameters. In this paper, the focus has been given to the error analysis of exponential RLS algorithms for the input data with abrupt property changes. The voiced speech signal is chosen as the principal application. In order to analyze the error performance of the exponential RLS algorithm, deterministic properties of the exponential RLS algorithms is first analyzed for the case of abrupt parameter changes, the impulsive input(or error variance) synchronous to the abrupt change of parameter vectors actually enhances the convergence of the exponential RLS algorithm. The analysis has also been verified through simulations on the synthetic speech signal.

• 한국정밀공학회지
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• 제21권5호
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• pp.91-98
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• 2004

Human uses level difference and time difference to get space information. Therefore this paper shows that method to presume direction of sound source by time difference and to mark presumed position. The position means direction from geometrical center of sensors to the sound source. To get the time difference of microphones input level, we will be explained about arrangement of microphones which used for the sensor to take the sound signal. It is included distance among the 3 microphones and distance between microphones and sound source. Secondly, input signals are transmitted to CPU througth digital process. CPU is used to DSP(Digital Signal Processor) for manage the signal by real time. Finally, the position of sound source is perceived by an explained algorithm in this paper.

• 한국정보통신학회:학술대회논문집
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• 한국정보통신학회 2014년도 춘계학술대회
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• pp.57-58
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• 2014

디지털 필터는 신호처리 시스템에서 매우 중요한 역할을 수행한다. 일반적으로 입력 신호는 디지털 필터의 전달함수에 의해 출력이 결정된다. 그러나 입력신호가 다양한 소리 환경에 노출되어 있어 확인이 어려운 경우가 발생할 수도 있다. 본 연구에서는 노이즈 환경에 노출된 입력신호로부터 원입력신호를 추출하기 위한 역문제를 고려하였다.