• 한국통신학회논문지
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• 제22권10호
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• pp.2339-2347
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• 1997

The performance of the adaptive noise canceller using LMS algorithm is degraded by the gradient noise due to target speech signals. An adaptive noise canceller with speech detector was proposed to reduce this performande degradation. The speech detector utilized the adaptive prediction-error filter adapted by the NLMS algorithm. This paper discusses to enhance the performance of the adaptive noise canceller forthecorlored noise. The affine projection algorithm, which is known as faster than NLMS algorithm for correlated signals, is used to adapt the adaptive filter and the adaptive prediction error filter. When the voice signals are detected by the speech detector, coefficients of adaptive filter are adapted by the sign-error afine projection algorithm which is modified to reduce the miaslignment of adaptive filter coefficients. Otherwirse, they are adapted by affine projection algorithm. To obtain better performance, the proper step size of sign-error affine projection algorithm is discussed. As resutls of computer simulation, it is shown that the performance of the proposed ANC is better than that of conventional one.

• 한국통신학회논문지
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• 제21권5호
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

• 대한전자공학회논문지
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• 제22권2호
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• pp.18-24
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• 1985

주파수 영역-적응 필터는 시간 영역-원응 필터에 비하여 필터 차수가 일정치 이상으로 높아졌을 경우에는 계산량에 있어서 유리하게 된다. 본 논문에서는 각파수 영역-적응 필터 알고리즘에 관하여 μ-FLMS 알고리즘이라는 하나의 새로운 형태를 재시하였으며, 이 알고리즘과 시간 영역-적응 필터 μ-FLMS 알고리즘의 특성을 비교하였다. 두 필터의 특성을 비교한 결과, 동일한 수검속도하에서 주파수 영역-적응 필터가 계산량에서 유리하였다.

• 한국통신학회논문지
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• 제10권6호
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• pp.308-316
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• 1985

주파수 영역에서 블록 적응 필터(Block Adaptive Filter),의 새로운 구현 방법을 제시하였다. 블록 적응 필터링은 입력값을 Block 단위로 하였을 때 출력에서도 블록 단위 또는 유한한 값을 갖도록 하는 것이다. 본 논문에서는 Gordard 이론을 이용하여 Block Adaptive Filter의 고속 수렴 알고리즘을 전개하고 Saito의 알고리즘과 BLMS알고리즘의 수렴 결과와 비교하였다. 그 결과 본 논문에서 제시한 주파수 영역블럭 적응 필터의 알고리즘(FBAF)이 BLMS알고리즘의 수렴상태보다 빠르게 수렴하며 Satio의 알고리즘의 수렴 오차보다는 줄어들었다.

• 한국음향학회지
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• 제12권1호
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• pp.37-46
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• 1993

The adaptive algorithm for the Volterra filter is considered. Owing to its simplicity, the LMS algorithm for adaptive Volterra filter(AVF) is widely used as in linear adaptive filters. However, the convergence speed is unsatisfactory. For improving the convergence speed, the frequency domain LMS second order adaptive Volterra filter(FLMS-AVF) is proposed and analyzed. We show that the time and frequency domain LMS AVF's have the same steady state performance under approprate conditons. Moreover, it can be shown that this algorithm can improve the convergence speed significantly by applying self-orthogonalizing method.

• 전기학회논문지
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• 제60권1호
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• pp.150-155
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• 2011

In an active noise control systems, an IIR filter may cause a problem in stability beacause of its poles. For IIR filter, its poles goes sometimes out of a unit circle in a z-plane in the transition state, where the adaptive algorithm converges to the optimum value, which causes the system to diverge. Fuzzy LMS algorithm has a better convergence property than conventional LMS algorithms, but is not applicable to IIR filter because of the reasons. Stabilized adaptive algorithm could be improves stability by moving the pole of IIR filer toward the origin forcibly in the transient state, and by introducing forgetting factor to maintain the optimum convergence when it reaches to the steady state. In this paper, We proposed stabilized adaptive fuzzy LMS algorithms with IIR filter structures, for single channel active noise control with ill conditioned signal case. Computer simulations were performed to show the effectiveness of a proposed algorithm.

• Journal of information and communication convergence engineering
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• 제1권4호
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• pp.189-193
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• 2003

A new algorithm is presented for adaptive notch filtering of narrow band or sine signals for embedded among broad band noise. The notch filter is implemented as a constrained infinite impulse response filter with a minimal number of parameters, Based on the recursive prediction error (RPE) method, the algorithm has the advantages of the fast convergence, accurate results and initial estimate of filter coefficient and its covariance is revealed. A convergence criterion is also developed. By using the information of the noise-to-signal power, the algorithm can self-adjust its initial filter coefficient estimate and its covariance to ensure convergence.

• 대한전자공학회논문지
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• 제27권3호
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• pp.145-155
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• 1990

適應 循環 필터에서 시스템 傳達 函數의 極點이 單位圓 밖으로 나감으로써 시스템이 불안정해지는 것을 방지하기 위해 適應 補償器를 채용한 適應 循環 필터 알고리듬을 제안하였고 제안한 알고리듬이 超安定性을 만족한다는 것을 證明하였다. 제안한 알고리듬 適應 雜音 除去器에 응용하여 LS방법의 適應 循環 필터를 이용한 適應 雜音 除去器, 適應 非循環 필터를 이용한 適應 雜音 除去器와의 성능을 比較하여 제안한 알고리듬의 妥當性을 立證하였다.

• 한국정보기술응용학회:학술대회논문집
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• 한국정보기술응용학회 2005년도 6th 2005 International Conference on Computers, Communications and System
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• pp.45-48
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, We know the proposed algorithm is prefer than the existent algorithm.

• 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.