• Title/Summary/Keyword: adaptive filter algorithm

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A METHOD TO FORMULATE ALGORITHM OF ADAPTIVE CONTROL OF ACTIVE NOISE CONTROL

  • Takahiko Ono;Ken'iti Kodo
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1994.06a
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    • pp.775-780
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    • 1994
  • A new simple method to formulate the adaptive algorithm to control the coefficients of FIR filter is introduced. The filter is used in the active noise control system. The introduced algorithm includes the LMS algorithm as a special case. The validity of the theoretical result is confirmed by the computer simulation.

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Scalar Adaptive Kalman Filtering for Stellar Inertia! Attitude Determination

  • Jung, Jae-Woo;Cho, Yun-Cheol;Bang, Hyo-Choong;Tahk, Min-Jea
    • International Journal of Aeronautical and Space Sciences
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    • v.3 no.2
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    • pp.88-94
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    • 2002
  • This paper describes attitude determination algorithm for the low earth orbit(LEO) spacecraft using stellar inertial sensors. The cascaded gyro/star tracker extended Kalman filter is constructed to fuse two sensor data. And then the smoothing of the measurement are proposed for an unreasonable jump of star tracker. The smoothing algorithm for the rejection of star tracker error jumps is designed by scalar adaptive filter. The proposed algorithms operate to process the measurement of gyro/star tracker Kalman filter, therefore, it is comparatively simple to apply these methods to other integration systems. Simulations to gyro/star tracker integrated system show that the proposed method is effective.

Audio Enhancement Algorithm Using Adaptive Perceptual Filter (적응 지각 필터를 이용한 오디오 음질 개선 알고리즘)

  • 엄혜영;한헌수;홍민철;차형태
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.8
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    • pp.687-693
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    • 2003
  • In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

Station Based Detection Algorithm using an Adaptive Fading Kalman Filter for Ramp Type GNSS Spoofing (적응 페이딩 칼만 필터를 이용한 기준국 기반의 램프 형태 GNSS 기만신호 검출 알고리즘)

  • Kim, Sun Young;Kang, Chang Ho;Park, Chan Gook
    • Journal of Institute of Control, Robotics and Systems
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    • v.21 no.3
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    • pp.283-289
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    • 2015
  • In this paper, a GNSS interference detection algorithm based on an adaptive fading Kalman filter is proposed to detect a spoofing signal which is one of the threatening GNSS intentional interferences. To detect and mitigate the spoofing signal, the fading factor of the filter is used as a detection parameter. For simulation, the effect of the spoofing signal is modeled by the ramp type bias error of the pseudorange to emulate a smart spoofer and the change of the fading factor value according to ramp type bias error is quantitatively analyzed. In addition, the detection threshold is established to detect the spoofing signal by analyzing the change of the error covariance and the effect of spoofing is mitigated by controlling the Kalman gain of the filter. To verify the performance analysis of the proposed algorithm, various simulations are implemented. Through the results of simulations, we confirmed that the proposed algorithm works well.

A design of adaptive equalizer using the transversal walsh filter and the optimal LMS algorithm (횡단형 월쉬필터와 최적 LMS 기법을 이용한 적응 등화기의 설계)

  • 김종부
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.33B no.3
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    • pp.1-8
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    • 1996
  • This paper proposes a novel transversal filter and an optimal LMS algorithm, and show how these can be realized as an adaptive equalizer. The transversal filter consists of a walsh and block pulse functions. in the LMS algorithm with equalizers, the convergence factor is an improtant design parameter because it governs stability and convergence speed. The conventional adaptation techniques use a fixed time constant convergence factor by the trial and error method. In this paper, an optimal method in the choice of the convergence factor is proposed. The proposed algorithm is obtrained that is tailored for each filter tap and is updated at each iteration. The performance of the proposed algorithm is compared iwth those of the conventional TDL and DFT equalizers by computer simulations.

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Complex Fuzzy Logic Filter and Learning Algorithm

  • Lee, Ki-Yong;Lee, Joo-Hum
    • The Journal of the Acoustical Society of Korea
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    • v.17 no.1E
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    • pp.36-43
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    • 1998
  • A fuzzy logic filter is constructed from a set of fuzzy IF-THEN rules which change adaptively to minimize some criterion function as new information becomes available. This paper generalizes the fuzzy logic filter and it's adaptive filtering algorithm to include complex parameters and complex signals. Using the complex Stone-Weierstrass theorem, we prove that linear combinations of the fuzzy basis functions are capable of uniformly approximating and complex continuous function on a compact set to arbitrary accuracy. Based on the fuzzy basis function representations, a complex orthogonal least-squares (COLS) learning algorithm is developed for designing fuzzy systems based on given input-output pairs. Also, we propose an adaptive algorithm based on LMS which adjust simultaneously filter parameters and the parameter of the membership function which characterize the fuzzy concepts in the IF-THEN rules. The modeling of a nonlinear communications channel based on a complex fuzzy is used to demonstrate the effectiveness of these algorithm.

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High Performance De-interlacing Algorithm Based on Region Adaptive Interpolation Filter

  • Yang, Yang;Chen, Xiangdong;Wang, Jin;Jeong, Jechang
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2011.11a
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    • pp.200-203
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    • 2011
  • In order to convert interlaced video into progressive scanning format, this paper proposed a high performance de-interlacing algorithm based on region adaptive interpolation filter design. Specifically, usage of the 6-tap filter is only for the most complex region, but for the smooth and regular edge region, much more correlated filter such as 2-tap or 4-tap filter should be used instead. According to the experimental results, the proposed algorithm has achieved noticeably good performance.

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A Study on the Stand-Alone GPS Jump Error Smoothing Scheme (Stand-Alone GPS 점프오차 스무딩 기법 연구)

  • Lee, Tae-Gyoo;Kim, Kwangjin;Park, Heung-Won
    • Journal of Institute of Control, Robotics and Systems
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    • v.7 no.12
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    • pp.1015-1023
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    • 2001
  • error behaviour can be considered as a linear combination of low amplitude random noise and abrupt jumps. The reason of jump appearance can be explained by the semi-shading effects(buildings, trees), jamming, high dynamic of vehicle and so on. This study describes the stand-alone GPS error jump smoothing algorithm which is developed based on the scalar adaptive filter. The algorithm consists of the coarse jump smoothing and the fine jump smoothing. On the coarse smoothing step, GPS velocities or position differences are used as the measurement for the scalar adaptive filter. The purpose of adaptive filter is to smooth the jump errors. The coarse positions are detennined by the integration of smoothed velocities. On the fine smoothing step, the differences between GPS positions and the coarse positions are smoothed by another scalar adaptive filter. The reason of fine smoothing is based on the facts that smoothing accuracy depends on the variance ofusefuJ signa\. The coarse smoothing which deal with the difference of positions provides the rough error removing. So the coarse smoothed velocities can have much more low amplitude than the raw ones. The fine smoothing procedure provides high quality of filtering process. Simulation results show the efficiency of proposed scheme.

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Improving Covariance Based Adaptive Estimation for GPS/INS Integration

  • Ding, Weidong;Wang, Jinling;Rizos, Chris
    • Proceedings of the Korean Institute of Navigation and Port Research Conference
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    • v.1
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    • pp.259-264
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    • 2006
  • It is well known that the uncertainty of the covariance parameters of the process noise (Q) and the observation errors (R) has a significant impact on Kalman filtering performance. Q and R influence the weight that the filter applies between the existing process information and the latest measurements. Errors in any of them may result in the filter being suboptimal or even cause it to diverge. The conventional way of determining Q and R requires good a priori knowledge of the process noises and measurement errors, which normally comes from intensive empirical analysis. Many adaptive methods have been developed to overcome the conventional Kalman filter's limitations. Starting from covariance matching principles, an innovative adaptive process noise scaling algorithm has been proposed in this paper. Without artificial or empirical parameters to be set, the proposed adaptive mechanism drives the filter autonomously to the optimal mode. The proposed algorithm has been tested using road test data, showing significant improvements to filtering performance.

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Adaptive Noise Reduction on the Frequency Domain using the Sign Algorithm.

  • Lee, Jae-Kyung;Yoon, Dal-Hwan;Min, Seung-Gi
    • 제어로봇시스템학회:학술대회논문집
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    • 2003.10a
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    • pp.57-60
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    • 2003
  • We have proposed the adaptive noise reduction algorithm using the MDFT. The algorithm proposed use the linear prediction coefficients of the AR method based on Sign algorithm that is the modified LMS instead of the least mean square(LMS). The signals with a random noise tracking performance are examined through computer simulations and confirmed that the high speed adaptive noise reduction processing system is realized with rapid convergence.

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