• 전기학회논문지
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• 제59권3호
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• pp.669-673
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• 2010

This paper proposes a new active noise control (ANC) system for canceling broadband noise. The proposed ANC system is designed based on the multiband-structured delayless subband adaptive filter (MDSAF), which has advantages of fast-convergence speed and higher noise reduction performance by eliminating the aliasing and band-edge effects caused by band-partitioning. The simulation results show that the proposed ANC system has faster convergence speed as compared to the conventional ANC systems and effectively reduces the wideband noise.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2003년도 ICCAS
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• pp.57-60
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• 2003

We have proposed the adaptive noise reduction algorithm using the MDFT. The algorithm proposed use the linear prediction coefficients of the AR method based on Sign algorithm that is the modified LMS instead of the least mean square(LMS). The signals with a random noise tracking performance are examined through computer simulations and confirmed that the high speed adaptive noise reduction processing system is realized with rapid convergence.

• 조명전기설비학회논문지
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• 제18권1호
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• pp.154-160
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• 2004

본 논문에서는 공간 외부의 원거리에 위치하는 소음원에 의하여, 창문 등을 통해 실내 공간에 소음이 전파될 때, 능동소음제어 기법을 이용하여 이를 제어하는 능동 방음막 (active noise curtain)을 구현하였다. 적응필터 알고리즘으로 다중채널 LMS 알고리즘을 사용하는 기법과 소음원이 원거리에 있을 때는 소음의 특성이 평면파에 가깝다는 가정 하에 제어용 스피커 수만큼의 단일채널 LMS 알고리즘을 사용하는 기법을 제시하였으며 적응필터의 안정도를 향상시키기 위하여 IIR 필터의 안정도를 향상시키는 안정화 기법을 제안하였다. TMS320VC33 DSP보드를 이용하여 제안된 기법을 정현파 소음 및 도로 소음에 대해 실험을 수행한 결과 MCLMS를 사용한 경우와 유사하거나 다소 나은 결과를 보였으며 제안한 기법이 실제 적용 시에는 제어용 스피커와 마이크로폰을 일체형으로 제작이 가능하여 하드웨어 실현에 유리하고 설치비도 절약되리라 예상된다.

• The Journal of the Acoustical Society of Korea
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• 제22권1E호
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• pp.33-42
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• 2003

Kuo et al proposed an on-line method for an adaptive prediction error filter for improving secondary path modeling performance in the modeling method of the secondary path. This method have some disadvantages, namely having to use additive noise with the result that noise control performance is not good since it is focused on the estimated performance of the secondary path. In this paper, we proposes a modified Kuo model using gain control parameter and delay. It uses a reference signal for additive noise to improve the problems in the existing Kuo model.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2010년도 춘계학술대회 논문집
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• pp.740-741
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• 2010

• 대한전기학회논문지:시스템및제어부문D
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• 제55권8호
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• pp.375-377
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2006년도 심포지엄 논문집 정보 및 제어부문
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• pp.30-32
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be Improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• 전기학회논문지
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• 제58권11호
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• pp.2299-2301
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• 2009

The adaptive IIR filters for active noise control systems are more effective when acoustic feedback exists, but the adaptive IIR filters could be unstable when the filter algorithm is not yet converged. In this paper, auto-secondary path estimation techniques and a stabilizing process for adaptive Multi-Channel Recursive LMS (MCRLMS) filters are developed to improve the stability of multi-channel active noise control systems. Experiments using a TMS320VC33 digital signal processor in a three dimensional enclosure have performed to show the effectiveness of the proposed algorithm.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 하계학술대회 논문집 D
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• pp.2766-2768
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• 2003

In this paper, positions of microphones and speakers in adaptive noise control systems are selected using optimization techniques. To get the optimal control characteristics of adaptive noise control system, It is necessary to optimize positions of microphones and speakers. Assume that control speakers are placed in a experimental enclosure several place indoors. Acoustic transfer functions from control speaker positions to microphone positions are measured for simulation program by experiments, and simulated annealing methods are used to select optimal positions of speakers and microphones.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2000년도 추계학술대회논문집
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• pp.443-448
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• 2000

In this paper, we propose a novel adaptive feedforward controller for periodic disturbance and noise cancellation, with a frequency tracking capability. It can be added to an existing feedback control system without altering the original closed-loop characteristics, which is based on adaptive algorithm. We introduce novel algorithm "Constrained AFC(adaptive feedforward controller) algorithm" that increase the convergence region regardless of the delay in the closed loop system. In the algorithms, coefficients of the controller are adapted using the residuals of constrained structure which are defined in such a way that the coefficients become time invariant. The proposed algorithm not only estimate the magnitude and phase of the tonal disturbance and noise but also track the frequency of the tone, which changes in quasi-static manner. The frequency tracking algorithm uses the instantaneous frequency approach based on Hilbert transform. A number of computer simulations have been carried out in order to demonstrate the effectiveness of proposed method under various conditions.