• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.179-182
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• 2002

In this paper, we propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the proposed algorithm is suitable for adaptive signal processing area using speech or acoustic field.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.351-359
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• 2003

In this paper, we proposed a method for acoustic echo cancellation based on independent component analysis. When the large acoustic noise is picked up by the microphone, the performance of echo cancellation decreased. We used two microphones that received echo signal which is linearly mixed with the noise, then separated the echo signals from the received signals with independent component analysis algorithm. The separated echo signal is used for the reference signal of adaptive algorithm which leads to better performance of the echo cancellation. Computer simulation results show the validity of the proposed method.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.144-155
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• 1998

The array weights that will maximize the SNR for any type of noise environment are determined by the function of the antenna design configuration and the directions of receiving target and interference signals. The conventional SLCs(sidelobe cancellers) using the SNR maximization perform worst from the saturation of the receiving system of main channel when the main antenna has pattern with high gain at the arrival angle of strong interference. In this paper, the new SLC is accomplished by using two independent antenna architecture. Main antenna is implemented with adaptive nulling, which is used for rejecting high-power interference primarily. Auxiliary antenna is realized with adaptive array for receiving interference signal to be suppressed completely, which has a characteristics of sufficient gain for every direction. The new SLC is implemented with above both antennas. We show that the new SLC, which consists of the adaptive nulling main antenna and the adaptive array auxiliary antenna, is useful in reducing the effect of strong interference like jammer, because the adaptive nulling at main antenna prevents its receiver and signal processor for saturation by strong interference. The proposed SLC has improved SNR over the conventional SLCs. The improved SNR at sidelobe region is typically more than 7 dB for a given test signal. Moreover, it improves the SNR of about 20 dB under strong interference at mainlobe.

• Proceedings of the Korea Multimedia Society Conference
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• 2004.05a
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• pp.179-182
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• 2004

무선통신분야에서 LMS5(Least Mean Square) 알고리즘은 식이 간단하고 계산량이 비교적 적기 때문에 널리 사용되고 있다. 그러나 시간영역에서 처리할 경우 입력신호의 고유치 변동폭이 넓게 분포되어 수렴속도가 저하하는 문제점이 있다. 이를 해결하기 위하여 신호를 FFT(Fast Fourier Trasnform)나 DCT(Discrete Cosine Transform)로 변환하여 신호간의 상관도를 제거함으로써 시간영역에서 LMS알고리즘을 적용할 때 보다 수렴속도를 크게 향강시킬 수 있다. 본 논문에서는 수렴속도 향상을 위해 시간영역의 적응 알고리즘을 직교변환인 고속웨이브렛(wavelet)변환을 이용하여 변환영역에서 수행하며, 짧은 필터계수를 가지는 DWT(Discrete Wavelet Transform)특성에 맞는 Fast running FIR 알고리즘을 이용하여 WTLMS(Wavelet Transform LMS)적응알고리즘을 통신시스템에 적용한다. 적응 알고리즘의 성능향상을 위하여 시간에 따라 적응상수의 크기를 가변시켜 수렴 초기에는 큰 적응상수로 따른 수렴이 가능하도록 하고 점차 적응상수의 크기를 줄여서 misadjustment도 줄이는 방법의 적응 알고리즘을 제안하였다. 제안한 알고리즘을 실제로 적응잡음제거기(adaptive noise canceler)에 적용하여 컴퓨터 시뮬레이션을 하였으며, 각 알고리즘들의 계산량, 수렴속도를 이용하여 각각 비교, 분서하여 그 성능이 우수함을 입증하였다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.19-22
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• 2000

최근 이동전화의 사용이 급격히 확산됨에 따라 핸즈프리 단말기를 이용한 전화통신의 필요성이 대두되고 있다. 차량내 핸즈프리 통신상황의 경우 근거리에 위치한 스피커와 마이크로폰의 커플링에 의해 발생하는 음향반향과 차량내에 존재하는 배경잡음은 통화 품질을 크게 저하시킨다. 본 논문에서는 이동통신에 적합한 음향반향제거기와 잡음제거기의 결합시스템을 제안하고, 이를 고정 소수점 DSP를 이용하여 실시간 구현하였다. 실시간 구현을 위하여 음향반향제거기에는 NLMS 알고리즘에 의해 구동되는 제한된 차수의 적응반향제거기법을 사용하였다. 잔여반향 및 배경잡음제거를 위해 CDMA방식의 셀룰라 이동통신에 사용되는IS-127 EVRC음성 부호화기의 표준안에 포함된 잡음제거방식을 사용하였다. 제안된 시스템을 16 비트 고정소수점DSP인 OAK DSP Core를 이용하여 약 18.6MIPS의 연산량으로 실시간 구현되었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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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.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.7-10
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• 2000

This paper focuses on the development of speech enhancement techniques for hands-free audio terminals, including two major problems : noise cancellation and acoustic echo cancellation. The objective is to find a joint structure to get a near-end speech signal with minimum distortion and low levels of echo and noise. To solve the two problems, a new promising technique is studied and tested in computer simulation conditions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.756-763
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• 2009

The Least Mean Square (LMS) algorithm is often used to restore signal corrupted by additive noise. A major defect of this algorithm is that the excess Mean Square Error (EMSE) increases linearly according to speech signal power. This result reduces the efficiency of performance significantly due to the large EMSE around the optimum value. Choosing a small step size solves this defect but causes a slow rate of convergence. The step size must be optimized to satisfy a fast rate of convergence and minimize EMSE. In this paper, the Instantaneous Gain Control (IGC) algorithm is proposed to deal with the situation as it exists in speech signals. Simulations were carried out using a real speech signal combined with Gaussian white noise. Results demonstrate the superiority of the proposed IGC algorithm over the LMS algorithm in rate of convergence, noise reduction and EMSE.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.109-112
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• 1999

본 연구에서는 신호원에 대한 사전 정보 없이 혼합된 신호로부터 잡음 신호만을 선택적으로 제거할 수 있는 새로운 형태의 능동 소음 제거기(Active noise canceller)를 제안한다. 음성신호와 같은 독특성을 갖는 신호의 분리에 효과적으로 사용되는 동적 재귀 신경망 (Dynamic recurrent neural network)을 원하는 신호원에 섞인 잡음신호를 분리하여 선택적으로 제거하기 위한 능동소음제거기의 전처리기로 미용한다. 능동 소음 제거기는 분리된 잡음 신호에 대한 역 위상 신호를 적응적으로 발생함으로써 특정 위치에서 원하는 신호만을 선택적으로 남길 수 있도록 한다. 컴퓨터를 이용한 시뮬레이션에서는 제안된 시스템이 선택적인 소음제거에 효과적임을 보인다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.699-702
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• 2009

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows the improved SNR of about more than 20 dB.