• Title/Summary/Keyword: adaptive filter algorithm

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Design of Reliable Adaptive Fitter with Fault Tolerance Using DSP (DSP를 이용한 고장허용을 갖는 신뢰 적응 필터 설계)

  • 유동완;이전우;서보혁
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.50 no.1
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    • pp.8-13
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    • 2001
  • LMS algorithm has been used for plant identifier and noise cancellation. This algorithm has been researched for performance enhancement of filtering. The design and development of a reliable system has been becoming a key issue in industry field because the reliability of a system is considered as an important factor to perform the system's function successfully. And the computing with reliability and fault tolerance is a important factor in the case of aviation, system communication, and nuclear plant. This paper presents design of reliable adaptive filter with fault tolerance. Generally, redundancy is used for reliability. In this case it needs computing or circuit for voting mechanism, or fault detection. Therefore it has simple computing, and practicality for application. And in this paper, reliability of adaptive filter is analyzed. The effectiveness of the proposed adaptive filter is demonstrated to the case studies of plant identifier and noise cancellation by using DSP.

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우리나라 의용생체공학의 현황과 전망

  • 이충웅
    • Journal of Biomedical Engineering Research
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    • v.10 no.2
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    • pp.83-88
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    • 1989
  • This paper is a study on the design of adptive filter for QRS complex detection. We propose a simple adaptive algorithm to increase capability of noise cancelation in QRS complex detection with two stage adaptive filter. At the first stage, background noise is removed and at the next stage, only spectrum of QRS complex components is passed. Two adaptive filters can afford to keep track of the changes of both noise and QRS complex. Each adaptive filter consists of prediction error filter and FIR filter The impulse response of FIR filter uses coefficients of prediction error filter. The detection rates for 105 and 108 of MIT/BIH data base were 99.3% and 97.4% respectively.

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Complexity Reduction of an Adaptive Loop Filter Based on Local Homogeneity

  • Li, Xiang;Ahn, Yongjo;Sim, Donggyu
    • IEIE Transactions on Smart Processing and Computing
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    • v.6 no.2
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    • pp.93-101
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    • 2017
  • This paper proposes an algorithm for adaptive loop filter (ALF) complexity reduction in the decoding process. In the original ALF algorithm, filtering for I frames is performed in the frame unit, and thus, all of the pixels in a frame are filtered if the current frame is an I frame. The proposed algorithm is designed on top of the local gradient calculation. On both the encoder side and the decoder side, homogeneous areas are checked and skipped in the filtering process, and the filter coefficient calculation is only performed in the inhomogeneous areas. The proposed algorithm is implemented in Joint Exploration Model (JEM) version 3.0 future video coding reference software. The proposed algorithm is applied for frame-level filtering and intra configuration. Compared with the JEM 3.0 anchor, the proposed algorithm has 0.31%, 0.76% and 0.73% bit rate loss for luma (Y) and chroma (U and V), respectively, with about an 8% decrease in decoding time.

Research on Noise Reduction Algorithm Based on Combination of LMS Filter and Spectral Subtraction

  • Cao, Danyang;Chen, Zhixin;Gao, Xue
    • Journal of Information Processing Systems
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    • v.15 no.4
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    • pp.748-764
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    • 2019
  • In order to deal with the filtering delay problem of least mean square adaptive filter noise reduction algorithm and music noise problem of spectral subtraction algorithm during the speech signal processing, we combine these two algorithms and propose one novel noise reduction method, showing a strong performance on par or even better than state of the art methods. We first use the least mean square algorithm to reduce the average intensity of noise, and then add spectral subtraction algorithm to reduce remaining noise again. Experiments prove that using the spectral subtraction again after the least mean square adaptive filter algorithm overcomes shortcomings which come from the former two algorithms. Also the novel method increases the signal-to-noise ratio of original speech data and improves the final noise reduction performance.

A Study on Eliminating the Error-Smoothing Filter from HARF Algorithm (HARF 알고리즘에서의 오차 완화 필터 제법에 관한 연구)

  • 신윤기;이종각
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.20 no.4
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    • pp.1-9
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    • 1983
  • In designing the adaptive recursive filter using the MRAS hyperstable output-error identifier, the most crucial point is the design of the error-smoothing filter In this paper, it is shown that by modifyins the HARF algorithm properly, we can obtain an algorithm which requires no error-smoothing filter and has relatively high convergence rate.

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Adaptive Bilinear Lattice Filter(II)-Least Squares Lattice Algorithm (적응 쌍선형 격자필터 (II) - 최소자승 격자 알고리즘)

  • Heung Ki Baik
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.29B no.1
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    • pp.34-42
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    • 1992
  • This paper presents two fast least-squares lattice algorithms for adaptive nonlinear filters equipped with bilinear system models. The lattice filters perform a Gram-Schmidt orthogonalization of the input data and have very good numerical properties. Furthermore, the computational complexity of the algorithms is an order of magnitude snaller than previously algorithm is an order of magnitude smaller than previously available methods. The first of the two approaches is an equation error algorithm that uses the measured desired response signal directly to comprte the adaptive filter outputs. This method is conceptually very simple`however, it will result in biased system models in the presence of measurement noise. The second approach is an approximate least-squares output error solution. In this case, the past samples of the output of the adaptive system itself are used to produce the filter output at the current time. Results of several experiments that demonstrate and compare the properties of the adaptive bilinear filters are also presented in this paper. These results indicate that the output error algorithm is less sensitive to output measurement noise than the squation error method.

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Study on Satellite Vibration Control Using Adaptive Algorithm

  • Oh, Choong-Seok;Oh, Se-Boung;Bang, Hyo-Choong
    • 제어로봇시스템학회:학술대회논문집
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    • 2005.06a
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    • pp.2120-2125
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    • 2005
  • The principal idea of vibration isolation is to filter out the response of the system over the corner frequency. The isolation objectives are to transmit the attitude control torque within the bandwidth of the attitude control system and to filter all the high frequency components coming from vibration equipment above the bandwidth. However, when a reaction wheels or control momentum gyros control spacecraft attitude, vibration inevitably occurs and degrades the performance of sensitive devices. Therefore, vibration should be controlled or isolated for missions such as Earth observing, broadcasting and telecommunication between antenna and ground stations. For space applications, technicians designing controller have to consider a periodic vibration and disturbance to ensure system performance and robustness completing various missions. In general, past research isolating vibration commonly used 6 degree order freedom isolators such as Stewart and Mallock platforms. In this study, the vibration isolation device has 3 degree order freedom, one translational and two rotational motions. The origin of the coordinate is located at the center-of-gravity of the upper plane. In this paper, adaptive notch filter finds the disturbance frequency and the reference signal in filtered-x least mean square is generated by the notch frequency. The design parameters of the notch filter are updated continuously using recursive least square algorithm. Therefore, the adaptive filtered-x least mean square algorithm is applied to the vibration suppressing experiment without reference sensor. This paper shows the experimental results of an active vibration control using an adaptive filtered-x least mean squares algorithm.

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A Study on The Modified Adaptive Median Filter Algorithm Using Odd/Even Multi-Shell (홀/짝 다중 셀을 이용한 수정된 가변 미디언 필터 알고리즘에 관한 연구)

  • Lee, Il-Gwon;Jo, Sang-Bok;Gong, Hyeong-Yun
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.38 no.4
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    • pp.401-410
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    • 2001
  • An adaptive Odd/Even Multi-shell Median Filter(Adaptive O/E MMF) is proposed by using a adaptive threshold strategy and odd/even multishells. This algorithm was verified in MATLAB and implemented on FPGA. The performance of the algorithm is evaluated by adding impulse noise and line missing into well-known images. A proposed adaptive threshold strategy may reduce the computation time and redundant arrangement. odd and even multishells can recover line missing of vertical and diagonal directions respectively Therefore, the algorithm proposed in this work can be effectively used in real time image processing applications.

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A Study on the Performance of a Radar Clutter Suppression Algorithm Based on the Adaptive Clutter Prewhitening Filter and Droppler Filter Bank (Adaptive Clutter Prewhitening Filter와 Doppler Filter Bank를 이용한 레이다 Clutter 제거 알고리듬의 성능에 관한 연구)

  • Kim, Yong-Ho;Lee, Hwang-Soo;Un, Chong-Kwan;Lee, Won-Kil
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.26 no.6
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    • pp.140-146
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    • 1989
  • In many situations, radar targets are embedded in a clutter environment and clutter rejection is required. The clutter is unwanted radar echoes and may arise owing to reflections from ground and weather disturbances and statistical properties of the clutter vary with range and azimuth as well as time. That is, adaptive signal processing is required. In this paper, a clutter suppression algorithm based on the clutter whitening filter (WF) and doppler filter bank(DFB) is described which provides improved performance compared with conventional nonadaptive clutter suppression algorithm that is the cascaded moving target indicator (MTI) and (DFB). The clutter whitening filter algorithm is based on the Burg's maximum entropy method.

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Stimulus Artifact Suppression Using the Stimulation Synchronous Adaptive Impulse Correlated Filter for Surface EMG Application

  • Yeom, Ho-Jun;Park, Ho-Dong;Chang, Young-Hui;Park, Young-Chol;Lee, Kyoung-Joung
    • Journal of Electrical Engineering and Technology
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    • v.7 no.3
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    • pp.451-458
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    • 2012
  • The voluntary EMG (vEMG) signal from electrically stimulated muscle is very useful for feedback control in functional electrical stimulation. However, the recorded EMG signal from surface electrodes has unwanted stimulation artifact and M-wave as well as vEMG. Here, we propose an event-synchronous adaptive digital filter for the suppression of stimulation artifact and M-wave in this application. The proposed method requires a simple experimental setup that does not require extra hardware connections to obtain the reference signals of adaptive digital filter. For evaluating the efficiency of this proposed method, the filter was tested and compared with a least square (LS) algorithm using previously measured data. We conclude that the cancellation of both primary and residual stimulation artifacts is enhanced with an event-synchronous adaptive digital filter and shows promise for clinical application to rehabilitate paretic limbs. Moreover because this algorithm is far simpler than the LS algorithm, it is portable and ready for real-time application.