• Title/Summary/Keyword: VSSLMS

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An Implementation of Adaptive Noise Canceller using Instantaneous Signal to Noise Ratio with DSP Processor (순시신호 대 잡음비 알고리즘을 이용한 적응 잡음 제거기의 DSP 구현)

  • Lee, Jae-Kyun;Ryu, Boo-Shik;Kim, Chun-Sik;Lee, Chae-Wook
    • Journal of the Institute of Convergence Signal Processing
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    • v.10 no.3
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    • pp.158-163
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    • 2009
  • LMS(Least Mean Square) algorithm requires simple equation and is used widely because of the low complexity. If the convergence speed increase, LMS algorithm has a divergence in case of sharp environment changes. And if a stability increase, the convergence speed becomes slow. This algorithm based on a trade off between fast convergence and system stability. To improve this problem, VSSLMS (Variable Step Size LMS) algorithm was developed. The VSSLMS algorithm improved the convergence speed and performance as adjusting step size using error signal. In this paper, I-VSSLMS algorithm is proposed tor improve the performance of adaptive noise canceller in real-time environments. The proposed algorithm is applied to adaptive noise canceller using TMS320C6713 DSP board and we did simulation by real time. Then we compared performance of each algorithm and demonstrated that proposed algorithm has superior performance.

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Research about Adjusted Step Size NLMS Algorithm Using SNR (신호 대 잡음비를 이용한 Adjusted Step Size NLMS알고리즘에 관한 연구)

  • Lee, Jae-Kyun;Park, Jae-Hoon;Lee, Chae-Wook
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.4C
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    • pp.305-311
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    • 2008
  • In this paper, we proposed an algorithm for adaptive noise cancellation (ANC) using the variable step size normalized least mean square (VSSNLMS) in real-time automobile environment. As a basic algorithm for ANC, the LMS algorithm has been used for its simplicity. However, the LMS algorithm has problems of both convergence speed and estimation accuracy in real-time environment. In order to solve these problems, the VSSLMS algorithm for ANC is considered in nonstationary environment. By computer simulation using real-time data acquisition system(USB 6009), VSSNLMS algorithm turns out to be more effective than the LMS algorithm in both convergence speed and estimation accuracy.

Categorized VSSLMS Algorithm (Categorized 가변 스텝 사이즈 LMS 알고리즘)

  • Kim, Seon-Ho;Chon, Sang-Bae;Lim, Jun-Seok;Sung, Koeng-Mo
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.8
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    • pp.815-821
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    • 2009
  • Information processing in variable and noisy environments is usually accomplished by means of adaptive filters. Among various adaptive algorithms, Least Mean Square (LMS) has become the most popular for its robustness, good tracking capabilities and simplicity, both in terms of computational load and easiness of implementation. In practical application of the LMS algorithm, the most important key parameter is the Step Size. As is well known, if the Step Size is large, the convergence rate of the algorithm will be rapid, but the steady state mean square error (MSE) will increase. On the other hand, if the Step Size is small, the steady state MSE will be small, but the convergence rate will be slow. Many researches have been proposed to alleviate this drawback by using a variable Step Size. In this paper, a new variable Step Size LMS(VSSLMS) called Categorized VSSLMS (CVSSLMS) is proposed. CVSSLMS updates the Step Size by categorizing the current status of the gradient, hence significantly improves the convergence rate. The performance of the proposed algorithm was verified from the view point of convergence rate, Excessive Mean Square Error(EMSE), and complexity through experiments.

Compensation for Nonlinear RE Power Amplifier using a Variable Step-Size LMS algorithm

  • Kim, Hyoun kuk;Park, Ke young;Lee, Yong min
    • Proceedings of the IEEK Conference
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    • 2002.06a
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    • pp.153-156
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    • 2002
  • An adaptive predistorr is proposed to compensate for the nonlinear distortion of a high power amplifier (HPA) in 16 QAM system. It fumed out that the proposed predistorter using a variable step-size least mean square (VSSLMS) algorithm is stable and can reduce the Total Distortion (TD) to 0. 1dB at the HPA output backoff=0.0 dB.

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Design of a New VSS-Adaptive Filter for a Potential Application of Active Noise Control to Intake System (흡기계 능동소음제어를 위한 적응형 필터 알고리즘의 개발)

  • Kim, Eui-Youl;Kim, Ho-Wuk;Lee, Sang-Kwon
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2009.10a
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    • pp.231-239
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    • 2009
  • The filtered-x LMS (FX-LMS) algorithm has been applied to the active noise control (ANC) system in an acoustic duct. This algorithm is designed based on the FIR (finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filteredu LMS algorithm (FU-LMS) based on infinite impulse response (IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

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Design of a New VSS-Adaptive Filter for a Potential Application of Active Noise Control to Intake System (흡기계 능동소음제어를 위한 적응형 필터 알고리즘의 개발)

  • Kim, Eui-Youl;Kim, Byung-Hyun;Kim, Ho-Wuk;Lee, Sang-Kwon
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.22 no.2
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    • pp.146-155
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    • 2012
  • The filtered-x LMS(FX-LMS) algorithm has been applied to the active noise control(ANC) system in an acoustic duct. This algorithm is designed based on the FIR(finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filtered-u LMS algorithm(FU-LMS) based on infinite impulse response(IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

Implementation of Adaptive Noise Canceller Using Instantaneous Gain Control Algorithm (순시 이득 조절 알고리즘을 이용한 적응 잡음 제거기의 구현)

  • Lee, Jae-Kyun;Kim, Chun-Sik;Lee, Chae-Wook
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.46 no.6
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    • pp.95-101
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    • 2009
  • Among the adaptive noise cancellers (ANC), the least mean square (LMS) algorithm has probably become the most popular algorithm because of its robustness, good tracking properties, and simplicity of implementation. However, it has non-uniform convergence and a trade-off between the rate of convergence and excess mean square error (EMSE). To overcome these shortcomings, a number of variable step size least mean square (VSSLMS) algorithms have been researched for years. These LMS algorithms use a complex variable step method approach for rapid convergence but need high computational complexity. A variable step approach can impair the simplicity and robustness of the LMS algorithm. The proposed instantaneous gain control (IGC) algorithm uses the instantaneous gain value of the original signal and the noise signal. As a result, the IGC algorithm can reduce computational complexity and maintain better performance.