• Title/Summary/Keyword: Filtered-x LMS Algorithm

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Modified FxLMS Algorithm for Active Noise Control and Its Real-Time Implementation

  • Mu, Xiangbin;Ko, JinSeok;Rheem, JaeYeol
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.9
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    • pp.172-176
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    • 2013
  • This paper presents a modified filtered-x least mean square (FxLMS) algorithm to improve the stability of active noise control (ANC) system in realistic environment. A real-time ANC system employing modified FxLMS is designed and implemented on digital signal processor (DSP) board. The ANC system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz and the performance is measured in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone with maximum 20 dB SPL attenuation can be generated around the location of error microphone.

Nonlinear Compensation of A Secondary Path in Active Noise Control Using A Modified Filtered-X LMS Algorithm (수정된 FXLMS 알고리듬을 이용한 능동소음제어 시스템 2차 경로 비선형 특성 적응보상 기법)

  • Jeong, I.S.;Ahn, K.Y.;Nam, S.W.
    • Proceedings of the KIEE Conference
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    • 2004.05a
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    • pp.22-25
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    • 2004
  • In active noise control (ANC) system, the convergence behavior of the Filtered- X Least Mean Square (FXLMS) algorithm may be affected by nonlinear distortion in the secondary path as in the power amplifiers (e.g., saturation), loudspeakers and transducers. This distortion may yields degrading the error reduction performance of the ANC systems. In this paper, the authors of this paper propose a more improved and stable FXLMS algorithm to compensate for the undesirable nonlinearity of the secondary-path, whereby the third-order Volterra model was employed for the identification of the nonlinear secondary-path. In particular, the proposed approach was based on the modification of the conventional FXLMS algorithm. Finally, the simulation results showed that the proposed approach yields better convergence property and more stable performance in the ANC systems.

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Transform domain algorithm for Improving Convergence Speed of Broadband Active Noise Control (광대역 능동소음제어의 수렴속도개선을 위한 변환영역 알고리듬)

  • Ahn, Doo-Soo;Kim, Jong-Boo;Lee, Tae-Pyo;Yim, Kook-Hyun
    • Proceedings of the KIEE Conference
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    • 1998.07b
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    • pp.644-646
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    • 1998
  • The main drawback of filtered-X LMS(FXLMS) algorithm for the ANC of broadband noises is its low convergence speed when the filtered reference signals are strongly correlated, producing a large eigenvalue spread in correlation matrix. This correlation can be caused either by autocorrelation of the signals of the reference sensors, or by coupling between the error path which introduces intercorrelation in the filtered reference signals. In this paper, we introduce a transform domain FXLMS(TD-FXLMS) algorithm that has a high convergence speed by orthogonal transform's decorrelation properties.

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Noise Attenuation Effect According to the Direction of Secondary Sound Source in Duct ANC System (Duct ANC System에서 부가음원 방향별 소음감소효과)

  • Lee, Eung-Suk;Lee, Hyung-Seok
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.19 no.3
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    • pp.251-260
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    • 2009
  • In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of canceling speaker in ANC system. Automobile exhaust noise was recorded at 800 rpm, 3500 rpm and 5000 rpm of a diesel engine. Directions of canceling speaker can be set to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for the experimentation. DSP board with TMS320C6416 chip of Texas Instrument Co. used to control the ANC system. The algorithm of this ANC system applied the Filtered-x-LMS algorithm that is modified to compensate for a property of DSP input signal and the secondary-path effect. As an experiment result, the direction of canceling speaker was proved to influence the reduction effect of noise. The $150^{\circ}$ duct in the attenuation effect of noise showed a better result than the $90^{\circ}$ or $30^{\circ}$ duct.

An Adaptive Active Noise Cancelling Model Using Wavelet Transform and M-channel Subband QMF Filter Banks (웨이브릿 변환 및 M-채널 서브밴드 QMF 필터뱅크를 이용한 적응 능동잡음제거 모델)

  • 허영대;권기룡;문광석
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.25 no.1B
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    • pp.89-98
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    • 2000
  • This paper presents an active noise cancelling model using wavelet transform and subband filter banks based on adaptive filter. The analysis filter banks decompose input and error signals into QMF filter banks of lowpass and highpass bands. Each filter bank uses wavelet filter with dyadic tree structure. The decomposed input and error signals are iterated by adaptive filter coefficients of each subband using filtered-X LMS algorithm. The synthesis filter banks make output signal of wideband with perfect reconstruction to prepare adaptive filter output signals of each subband. The analysis and synthesis niter hants use conjugate quadrature filters for Pefect reconstruction. Also, The delayed LMS algorithm model for on-line identification of error path transfer characteristics is used gain and acoustic time delay factors. The proposed adaptive active noise cancelling modelis suggested by system retaining the computational and convergence speed advantage using wavelet subband filter banks.

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Active Noise Control Algorithm having Fast Convergence (빠른 수렴성을 갖는 능동 소음제어 알고리즘에 관한 연구)

  • 나희승;박영진
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 1998.04a
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    • pp.670-677
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    • 1998
  • Many of the adaptive noise control systems utilize a form of the least mean square (LMS) algorithm. In the active control of noise, it is common practice to locate an error microphone far from the control source to avoid the near-field effects by evanescent waves. Such a distance between the control source and the error microphone makes a certain level of time-delay inevitable and, hence, may yield undesirable effects on the convergence properties of control algorithms such as filtered-x LMS. This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. Performances of the new LMS algorithms are presented in comparison with those by the conventional algorithms based on computer stimulations and experiments.

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Performance of CLMS Algorithm for Real-time Application in ANC Systems of Road Noise Input (도로소음 입력의 ANC시스템에서 실시간 적용의 CLMS 알고리즘의 성능)

  • Moon, Hak-Ryong;Shon, Jin-Geun
    • The Transactions of the Korean Institute of Electrical Engineers P
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    • v.63 no.4
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    • pp.260-265
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    • 2014
  • Recently, many active noise control (ANC) systems, which employ the adaptive filter controlling method, have been reported for eliminating unwanted noise. ANC systems based on the filtered-X least mean square (FXLMS) algorithm have a problem with compensating the acoustic feedback of secondary route. It is difficult to apply the real time, because transfer function of secondary route must be measured by off-line method to solve this problem. In this paper, we propose the ANC system that applies a correlation LMS(CLMS) algorithm for improving a problem of transfer function measurement. The proposed algorithm is based on input of road noise. The proposed ANC systems have an advantage of real-time process without degradation of performance, although there are many calculation compared with FXLMS algorithm.

Adaptive Active Noise Control in a Duct Using Improved SLMS Algorithms (개선된 SLMS 알고리즘을 이용한 덕트 내에서의 능동소음제어)

  • Seo, Sung-Dae;Nam, Ju-Hyung;Ahn, Dong-Jun;Nam, Hyun-Do
    • Proceedings of the KIEE Conference
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    • 2007.10a
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    • pp.433-434
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    • 2007
  • In this paper, active control of noise in a HVAC duct is considered. Most adaptive control filters have used FIR structures based on filtered-x LMS algorithms. But, the IIR structures are more desirable for the active control of duct noise in order to remove the poles introduced by the acoustic feedback and presented an algorithm to adjust the coefficients of an IIR filter using the recursive least mean square (RLMS) algorithm. A smoothed LMS algorithm is proposed to improve a convergent speed of filter parameters when the noise is wide band and power of input is time varying. And computer simulations have performed to show the effectiveness of the proposed algorithm.

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Characteristics of Expanded-CLMS Algorithm for Performance Improvement in ANC Systems for Road Noise Calming (도로소음 정온화를 위한 ANC시스템에서 성능개선을 위한 Expanded-CLMS 알고리즘의 특성)

  • Moon, Hak-ryong;Shon, Jin-geun
    • The Transactions of the Korean Institute of Electrical Engineers P
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    • v.64 no.3
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    • pp.169-174
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    • 2015
  • Noise problem that occurs on the road is raising a lot of problems in the economic, social and environmental aspects. The active noise control (ANC) systems based on the filtered-X least mean square(FxLMS) algorithm have a problem with compensating the acoustic feedback of secondary route. However, newly proposed correlation-LMS(CLMS) and expanded CLMS algorithms have slightly much calculation and are minutely behind performance, these have a advantage not in measuring transfer function onerously so that we can easily adapt these in real time. The CLMS and expanded CLMS algorithm was developed to improve the real-time implementation performance under the variable input noise such as road noise environment. In this paper, we compared and analyzed their performance. From the results of the Matlab simulation for an ANC system, it is shown that expanded CLMS algorithms are more convergence speed and keep the desirable performance even in the input of road noise situation.

Adaptive Feedforward Rejection of Microactuator Resonance in Hard Disk Drive Dual-stage Actuator Servo (하드디스크 드라이브 마이크로 구동기의 공진 영향 제거를 위한 적응 피드포워드 제어)

  • Oh, Dong-Ho;Lee, Seung-Hi;Baek, Sang-Eun;Na, Hee-Seung
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2000.06a
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    • pp.1596-1600
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    • 2000
  • We propose a novel adaptive feed forward controller (AFC) design method for rejecting the effect of micro actuator resonance in the design of dual-stage actuator servo systems for disk drives. Microactuator's resonance is one of important issues in dual-stage actuator servo, which varies up to ${\pm}10%$ per product and even during operation. We derive an adaptive algorithm for the proposed AFC design, which turns out to be identical to the delayed-x LMS algorithm which is a special form of the filtered-x LMS algorithm. In the algorithm, coefficients of the AFC are adapted by the residuals of constrained structure defined in such a way that the coefficients become time invariant. Contrary to the conventional AFC, it considers the phase delay of closed-loop transfer function at resonance frequency for system stability. We also apply an adaptive algorithm with frequency tracking capability. The frequency tracking algorithm is induced by the orthogonality of AFC coefficients. Computer simulations are carried out to demonstrate effect of the proposed AFCs.

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