• 한국음향학회지
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• 제34권2호
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• pp.130-137
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• 2015

본 논문에서는 충격성 소음의 능동 제어를 위해 제안된 Modified Filtered-x Affine Projection Sign Algorithm(MFxAPSA)의 수렴 속도를 향상시키기 위한 새로운 MFxAPSA를 제안하였다. 능동 소음 제어에서 소음원이 충격성 잡음을 포함하는 경우, 무한한 크기의 분산을 갖으려는 성질 때문에 Filtered-x affine Projection Sign Algorithm(FxLMS)와 같이 2차 모멘트를 기반으로 유도된 적응 알고리즘들은 수렴 속도가 매우 느리거나 발산하는 경향이 있다. MFxAPSA는 기존에 제안된 Affine Projection Sign Algorithm(APSA)을 능동 충격성 소음 제어에 적용한 알고리즘이다. APSA은 역행렬 연산을 요구하지 않는다는 장점으로 인해 낮은 연산량을 요구하는 능동 소음 제어에 적합하다. 본 논문에서는 기존의 MFxAPSA와 같이 역행렬 연산을 요구하지 않으면서 더 좋은 수렴 특성을 가지는 새로운 MFxAPSA를 제안하였다. 두 알고리즘의 성능을 비교하는 컴퓨터 모의 실험을 수행하여 제안된 알고리즘의 수렴 특성이 더 좋음을 보였다.

• 전기학회논문지P
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• 제54권3호
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• pp.119-128
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• 2005

In this paper, an robust adaptive backstepping controller is proposed for the chaos system with disturbances. This controller will be applicable to the chaos system of strict-feedback form and utilize the saturation function for decreasing the effect of disturbances derived from unmodelled dynamics and external noise. It shows that backstepping algorithm can be used to solve the problems of nonlinear system very well and robust controller can be designed without the variation of adaptive law. Simulation results are provided to demonstrate the effectiveness of the proposed controller.

• 한국통신학회논문지
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• 제25권6B호
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• pp.1092-1099
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• 2000

An adaptive array system can effectively remove all received interferences by using adaptive algorithms even though the received signal condition is not known. The conventional adaptive array systems, however, cannot remove all interferences adaptively and converge very slowly when the eigenvalue spread of the input covariance matrix is large. In the paper, a new adaptive array system called an automatic gain controller (AGC) Applebaum array and its control algorithm are proposed to overcome the performance degradation of conventional Applebaum array in multiple interference conditions. The performance analysis of the proposed AGC Applebaum array is described under the condition of multiple narrowband interferences. Simulation results show the array output signal-to-noise ratio (SNR) of the AGC Applebaum array increases by 30dB compared to that of the conventional Applebaum array in the simulation condition. The gain of the AGC Applebaum array in the incident direction of a weaker interference is also shown to be lower than that of the conventional Applebaum array.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1992년도 한국자동제어학술회의논문집(국내학술편); KOEX, Seoul; 19-21 Oct. 1992
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• pp.182-187
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• 1992

This paper proposes a method by which the estimated values of the unknown parameters of system are converged to the true values in finite time using adaptive deadbeat controller. After those are converged to the true values, the deviation from these values do not virtually exist or, if any, extremely small. Also we apply this technique of deadbeat convergence to a system contaminated with white noise or colored noise. It is shown that the estimated parameters of those systems approach the true values in finite time even though the performance do not match perfectly with the system without noises.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -1
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• pp.377-380
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• 2002

When it calculates the optimum price of filter coefficient, the many operation quantity is necessary. Is like that the real-time control is difficult and the hardware embodiment expense is big. The case which does not know advance information of input signal or the case where the statistical nature changes with change of surroundings environment is necessary the adaptive filter. Every hour to change a coefficient automatically and system in order to reach to the condition of optimum oneself, the fact that is the adaptive filter. When it does not the quality of input signal or it does not know the environment of surroundings every hour changing, it does not emit not to be, in order to collect, the fact that is the adaptive filter. The case of the Acoustic Echo Canceler does thousands filter coefficients in necessity. It reduces a many calculation quantity to respect, it uses the IIR filter from hour territory. Also it uses the block adaptive filter which has a block input signal and a block output signal. The former there is a weak point where the stability discrimination is always demanded. Consequently, The block adaptive filter is researched plentifully. This dissertation planned the block MADF adaptive filter used to MADf algorithm.

• Journal of Communications and Networks
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• 제7권3호
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2002년도 ICCAS
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• pp.54.4-54
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• 2002

1.Introduction 2. Channel Equalizer Using Adaptive Neural Network 3. Data Modulation and Demodulation Using Haar Transform 4. Binary Signal Transmission Using Haar Transform 5. Conclusion

• 한국정밀공학회지
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• 제14권3호
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• pp.107-113
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• 1997

This paper presents the active control of a flexible cantilever beam vibration. The cantilever beam was excitied by a steady-state harmonic and white noise point force and the control was performed by one piezo ceramic actuator bonded to the surface of the beam. An adaptive controller based on filtered-x LMS algorithm was used and the controller was defined by minimizing the square of the response of error sensor. In the experiment, gap sensor was used as an error sensor while the sinusoidal or white noise was applied as a disturbance. In the case of sinusoidal input, more than 20 dB of vibration reduction was achieved over all range of the natural frequencies and it takes 5 seconds to control the vibration at first natural frequency and 1 second at other natural frequencies. In the case of white noise input, 7 dB of vibration reduction was achieved at the first natural frequency and good control performance was achieved in the considered whole frequency range. Results indicate that the vibration of a flexible cantilever beam could be controlled effectively when the piezo ceramic actuator was used with filtered-x LMS algorithm.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1996년도 하계학술대회 논문집 B
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• pp.1327-1329
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• 1996

This paper presents an active noise control (ANC) algorithm using orthogonalization property of Walsh transform. Conventional ANC algorithm known as filtered-x LMS(FXL) algorithm has a problem of decreasing convergence speed in FIR adaptive filters operating in colored noise environments. Walsh transform decompose an input signal into a set of N uncorrelated components and reduce eigenvalue spread of autocorrelation matrix of input sequences. Computer simulations show that proposed (FXW) algorithm is superier to FXL in convergence speed.

• 대한전기학회논문지
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• 제35권11호
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• pp.503-510
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• 1986

A new algorithm based on the concept of prediction error minimization is suggested to estimate the time varying delay in discrete processes. In spite of the existence of the stochastic noise, this algorithm can estimate time varying delay accurately. Computation time of this algorithm is far less than that of the previous extended parameter methods. With the use of this algorithm, generalized minimum variance control shows good control behavior in simulations.