• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.11
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• pp.37-43
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• 1998

This paper proposes a new echo canceller that can be used in a full-duplex digital subscriber loop modem. The proposed echo canceller uses a NLMS-based parallel adaptation NLMS (PA-NLMS) algorithm. The PA-NLMS algorithm that an estimate of the nonstationarity to additive noise ratio (NNR) gained from two distinct NLMS processes is used to select the value of NLMS convergence-controlling parameter has been developed. Numerical results based on computer simulation show that the proposed algorithm has a convergence rate approaching that of the fastest possible NLMS process while improving on its MAC performance considerably.

• Journal of Communications and Networks
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• v.9 no.1
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• pp.18-27
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• 2007

The conventional normalized least mean square (NLMS) algorithm is the most widely used for adaptive identification within a non-stationary input context. The convergence of the NLMS algorithm is independent of environmental changes. However, its steady state performance is impaired during input sequences with low dynamics. In this paper, we propose a new NLMS algorithm which is, in the steady state, insensitive to the time variations of the input dynamics. The square soot (SR)-NLMS algorithm is based on a normalization of the LMS adaptive filter input by the Euclidean norm of the tap-input. The tap-input power of the SR-NLMS adaptive filter is then equal to one even during sequences with low dynamics. Therefore, the amplification of the observation noise power by the tap-input power is cancelled in the misadjustment time evolution. The harmful effect of the low dynamics input sequences, on the steady state performance of the LMS adaptive filter are then reduced. In addition, the square root normalized input is more stationary than the base input. Therefore, the robustness of LMS adaptive filter with respect to the input non stationarity is enhanced. A performance analysis of the first- and the second-order statistic behavior of the proposed SR-NLMS adaptive filter is carried out. In particular, an analytical expression of the step size ensuring stability and mean convergence is derived. In addition, the results of an experimental study demonstrating the good performance of the SR-NLMS algorithm are given. A comparison of these results with those obtained from a standard NLMS algorithm, is performed. It is shown that, within a non-stationary input context, the SR-NLMS algorithm exhibits better performance than the NLMS algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4C
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• pp.343-349
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• 2010

In this paper, we study real time AEC (acoustic echo canceller) based on NLMS adaptive filter. Proposed method improves conversation quality by enhancing the performance of AEC during double talk section and reduces the power consumption by controling the adaption operation of NLMS adaptive filter. Proposed method examines the convergence of the NLMS adaptive filter, stores the estimated echo path and chooses operation of NLMS adaptive filter. Furthermore if double talk is detected, the proposed AEC utilizes the stored echo path optionally considering missed double talk time. When the proposed AEC is used, the performance of the AEC is enhanced although the simple double talk detector is used and the power consumption of the AEC is reduced.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.93-98
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• 2009

In this paper, we propose a new VSS-NLMS algorithm through a simple modification of the conventional NLMS algorithm, which leads to a low complexity algorithm with enhanced performance. The step size of the proposed algorithm becomes smaller as the error signal is getting orthogonal to the input vector. We also show that the proposed algorithm is an approximated normalized version of the KZ-algorithm and requires less computation than the KZ-algorithm. We carried out a performance comparison of the proposed algorithm with the conventional NLMS and other VSS algorithms using an adaptive channel equalization model. It is shown that the proposed algorithm presents good convergence characteristics under both stationary and non-stationary environments despites its low complexity.

• The Journal of the Acoustical Society of Korea
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• v.12 no.5
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• pp.47-53
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• 1993

능동소음제어를 위하여 수렴속도가 빠르면서 정상상태오차를 최소화하는 오차보상 NLMS 알고리듬인 CNLMS 알고리듬을 제안하였다. 이 CNLMS 알고리듬에서는 수렴속도를 빠르게 하기 위하여 입력소음과 오차소음의 전력에 따라 수렴인자를 가변시킨 NLMS 알고리듬을 사용하였다. 도한 정상상태오차를 최소화하기 위하여 소음발생기를 사용하지 않고 오차경로에서 발생하는 소음오차를 더욱 작아지도록 보상하는 보조시스템을 사용하여 안정된 능동소음제어가 되도록 하였다. 이와 같은 시스템의 성능을 기존의 적응디지틀필터인 LMS 및 NLMS 알고리듬을 이용한 필터의 그 성능과 비교하였다. 각 알고리듬에 대한 시뮬레이션을 행한 결과, 제안한 CNLMS 알고리듬의 소음레벨은 LMS 및 NLMS 알고리듬의 것보다 각각 평균 14dB 및 6dB 정도 더 큰 감쇄효과가 있음을 확인하였다.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.4 s.95
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• pp.389-396
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• 2005

In this paper, we propose a pre-distortion method using the NLMS(Normalized Least Mean Square) algorithm to cope with hish PAPR(Peak to Average Power Ratio) problem in OFDM communication system. This proposed scheme estimates the distortion characteristics of HPA, and changes the characteristic against the distortion. Therefore, it can be shown that the adaptive characteristic of the NLMS pre-distorter is good to track the various nonlinear characteristic of HPA, even though HPA characteristic is changed by temperature variation or aging. From the performance analysis, SNR efficiency of NLMS pre-distorter is about $0.5\;\cal{dB}$ less than that of common numerical non-adaptive pre-distorter, when IBO(Input Back Off) is $0\;\cal{dB}$. However, the NLMS pre-distorter is better than the common numerical pre-distorter, because these two pre-distorters have similar performance in higher than $3\;\cal{dB}$ IBO, and the NLMS pre-distorter maintains the constant performance even though characteristic of HPA is changed.

• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.2995-2997
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• 2005

통신망에서 최적 적응 반향제거기(Echo Canceller; EC)는 반향성분이 길게 존재하는 환경에서도 실시간으로 동작할 수 있도록 알고리즘이 간결하여야 하며, 시간에 따라 빠르게 변하는 동특성의 반향경로에서도 동작을 보장할 수 있도록 빠른 수렴특성을 갖아야 한다. 또한, 전화망에서 수십 [ms] 이상의 지연이 발생 할 경우에도 반향제거 성능이 우수해야 한다. 본 논문에서는 이러한 조건을 만족시키기 위해 오차의 크기에 따라 수렴속도를 가변시키는 오차적응 NLMS(Error-Adaptive NLMS) 알고리즘을 제안하였으며, 시뮬레이션을 통해 일반적으로 사용되는 LMS(Least Mean Square) 알고리즘과 이를 개선한 NLMS(Normalized LMS) 알고리즘과 성능을 비교하였다.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.5
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• pp.1001-1004
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• 2010

Acoustic feedback and background noise variation can degrade the performance of an active noise control (ANC) system. In this paper, nonlinear ANC using a non-parametric VSS-NLMS (or NPVSS-NLMS) algorithm and online feedback path modeling is proposed, whereby the conventional linear ANC with online acoustic feedback-path modeling is further extended to nonlinear Volterra ANC with a linear acoustic feedback path. In particular, the step-size of the NPVSS-NLMS algorithm is controlled to reduce the effect of background noise variation in the ANC system. Simulation results demonstrate that the proposed approach yields better nonlinear ANC performance compared with the conventional nonlinear ANC method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6B
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• pp.1203-1210
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• 1999

Coefficients of the adaptive filter are misadjusted by primary signals which are uncorrelated with reference signals of the adaptive filter. In this paper, the normalized sign algorithm is analyzed and compared with the NLMS algorithm by the steady state performance and the transient characteristics when target signals are included in primary signals. The excess mean square error of the NLMS algorithm is proportional to the power of target signals. That of normalized sign algorithm is proportional to the square root of the target signal power. However, the convergence speed of the normalized sign algorithm is slower than that of NLMS algorithm. In this paper, it is shown that theoretical analysis of the steady state performance and the transient characteristics are well consisted with the results of computer simulation.

• Journal of Korea Multimedia Society
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• v.2 no.1
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• pp.99-108
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• 1999

In this paper, a new robust adaptive algorithm is proposed to improve the performance of AEC without computational burden. The proposed adaptive algorithm is based on NLMS algorithm, and its step-size is varied with the reference input signal power and the desired signal power. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. The convergence speed is comparable to NLMS algorithm at AEC application because the echo signals are attenuated about 10∼20 dBSPL. The characteristics of this algorithm is also analyzed and compared with conventional ones in this paper.