• 제목/요약/키워드: Filter convergence

검색결과 898건 처리시간 0.033초

Modified Unscented Kalman Filter for a Multirate INS/GPS Integrated Navigation System

  • Enkhtur, Munkhzul;Cho, Seong Yun;Kim, Kyong-Ho
    • ETRI Journal
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    • 제35권5호
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    • pp.943-946
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    • 2013
  • Instead of the extended Kalman filter, the unscented Kalman filter (UKF) has been used in nonlinear systems without initial accurate state estimates over the last decade because the UKF is robust against large initial estimation errors. However, in a multirate integrated system, such as an inertial navigation system (INS)/Global Positioning System (GPS) integrated navigation system, it is difficult to implement a UKF-based navigation algorithm in a low-grade or mid-grade microcontroller, owing to a large computational burden. To overcome this problem, this letter proposes a modified UKF that has a reduced computational burden based on the basic idea that the change of probability distribution for the state variables between measurement updates is small in a multirate INS/GPS integrated navigation filter. The performance of the modified UKF is verified through numerical simulations.

Equalization of Time-Varying Channels using a Recurrent Neural Network Trained with Kalman Filters (칼만필터로 훈련되는 순환신경망을 이용한 시변채널 등화)

  • 최종수;권오신
    • Journal of Institute of Control, Robotics and Systems
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    • 제9권11호
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    • pp.917-924
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    • 2003
  • Recurrent neural networks have been successfully applied to communications channel equalization. Major disadvantages of gradient-based learning algorithms commonly employed to train recurrent neural networks are slow convergence rates and long training sequences required for satisfactory performance. In a high-speed communications system, fast convergence speed and short training symbols are essential. We propose decision feedback equalizers using a recurrent neural network trained with Kalman filtering algorithms. The main features of the proposed recurrent neural equalizers, utilizing extended Kalman filter (EKF) and unscented Kalman filter (UKF), are fast convergence rates and good performance using relatively short training symbols. Experimental results for two time-varying channels are presented to evaluate the performance of the proposed approaches over a conventional recurrent neural equalizer.

Channel Enlargement of PON System Using Nonreciprocal Multiplexing Filter Based on CWDM

  • Kim, Bong-Kyu;Yoon, Bin-Young;Kwon, Yool
    • ETRI Journal
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    • 제31권2호
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    • pp.231-233
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    • 2009
  • We propose a nonreciprocal filter based on coarse wavelength division multiplexing (CWDM) that reduces the upstream channel insertion loss in a passive optical network (PON). We also propose a method to increase the number of channels/optical network units (ONUs) in PON systems using the proposed filter to reduce the service cost per subscriber. Experimental results show that the PON system with the proposed 4-channel filter can reduce the power budget of the upstream and increase the number of ONUs by 3 to 4 times that of a conventional time-division multiplexing PON.

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A New Approach of State Estimation based on Particle Filter (파티클 필터에 기반한 새로운 상태 예측 방법)

  • Park Seong-Keun;Ruy Kyung-Jin;Hwang Jae-Phil;Kim Eun-Tai
    • Proceedings of the Korean Institute of Intelligent Systems Conference
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    • 한국퍼지및지능시스템학회 2006년도 춘계학술대회 학술발표 논문집 제16권 제1호
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    • pp.245-248
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    • 2006
  • A particle filter is one of the most famous filters. The reason why the particle filter is widely used is that particle deals with the state estimation problem for not only linear models with Gaussian noise but also the non-linear models with non-Gaussian noise and it receives great attention from many engineering fields. In the point of view state estimator, particle filter is feedforward observer. According to the characteristic of dynamic system, the feedforward observer can estimate real state. However, the speed of convergence of feedforward observer between the actual state and the estimated state cannot be satisfied. Since the particle filter is a sort of feedforward observer, the convergence speed of particle filter is slow, and the particle filter cannot estimate actual state like particle collapse problem. In order to overcome the limitation of particle filter as a kind of feedfoward estimator, we propose a new particle filter which has feedback term, called particle filter with feedback. Our proposed method is analyzed theoretically and studied by computer simulation. Comparisons are made with other filtering mehod.

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Compact Spatial Triple-Band-Stop Filter for Cellular/PCS/IMT-2000 Systems

  • Kim, Dong-Ho;Yeo, Jun-Ho;Choi, Jae-Ick
    • ETRI Journal
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    • 제30권5호
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    • pp.735-737
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    • 2008
  • We propose a novel spatial multi-band-stop filter using modified multiple loop array elements to block electromagnetic waves or signals of mobile phones in public facilities. It operates at the following frequency bands: Korean cellular (824 MHz to 894 MHz), Personal Communication Service (PCS) (1.75 GHz to 1.87 GHz), and IMT-2000 (1.92 GHz to 2.17 GHz). Two frequency selective surfaces with modified multiple-loop elements are printed on the top and bottom of a pair-glass pane, which is a pair of glass panes with an air gap between them. A modified multiple-loop element with a meander line is used to make the size of the filter compact. The simulated and measured results show good agreement, which confirms the usefulness of the proposed tri-band spatial filter.

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An Acoustic Echo Canceller By Using the Reduced Lattice Filter Structure (축소격자필터 구조를 사용한 음향반향제거기)

  • 유재하;조성호;윤대희;차일환
    • Journal of the Korean Institute of Telematics and Electronics B
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    • 제32B권11호
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    • pp.1473-1480
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    • 1995
  • When the LMS algorithm is employed in the transversal filter structure, the computational complexity can be kept reasonably low. However, if the impulse response to be estimated is very long or signals involved are highly correlated like a speech the convergence speed becomes slow. The lattice filter is an excellent alternative to improve convergence speed since the lattice structure inherently has the orthogonal property among the backward prediction errors, but at the expense of the excessive computational load. If the input signal to be used can be sufficiently well modeled as a .RHO.-th order autoregressive(AR) process, the reflection coefficients after the .RHO.- th stage will be close to zero. Then, instead of employing the full lattice structure, the joint lattice filter structure can be implemented in conjunction with the transversal filter structure after the .RHO.-th stage. We propose, in this paper, this new lattice/transversal joint structure, and we will call it the reduced lattice filter. Using the reduced lattice filter, we are now able to achieve the performance as good as that of the lattice filter, while maintaining the complexity as low as that of the transversal filter. The proposed filter is particularly useful for an acoustic echo canceller due to the highly correlatedness nature of speeches and the long and frequently changing echo paths.

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The Improvement of High Convergence Speed using LMS Algorithm of Data-Recycling Adaptive Transversal Filter in Direct Sequence Spread Spectrum (직접순차 확산 스펙트럼 시스템에서 데이터 재순환 적응 횡단선 필터의 LMS 알고리즘을 이용한 고속 수렴 속도 개선)

  • Kim, Gwang-Jun;Yoon, Chan-Ho;Kim, Chun-Suk
    • Journal of the Korea Institute of Information and Communication Engineering
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    • 제9권1호
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    • pp.22-33
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    • 2005
  • In this paper, an efficient signal interference control technique to improve the high convergence speed of LMS algorithms is introduced in the adaptive transversal filter of DS/SS. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, is analyzed to prove theoretically the improvement of high convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Also, an increase in the stop-size parameter ${\mu}$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling which is downed the rate of convergence of the adaptive equalizer. Increasing the steady-state value of the average squared error, proposed algorithm also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

A study on Variable Step Size algorithms for Convergence Speed Improvement of Frequency-Domain Adaptive Filter (주파수영역 적응필터의 수렴속도 향상을 위한 가변스텝사이즈 알고리즘에 관한 연구)

  • 정희준;오신범;이채욱
    • Proceedings of the IEEK Conference
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    • 대한전자공학회 2000년도 추계종합학술대회 논문집(4)
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    • pp.191-194
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    • 2000
  • Frequency domain adaptive filter is effective to communication fields of many computational requirements. In this paper we propose a new variable step size algorithms which improves the convergence speed and reduces computational complexity for frequency domain adaptive filter. we compared MSE of the proposed algorithms with one of normalized FLMS using computer simulation of adaptive noise canceler based on synthesis speech.

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CONDITIONS IMPLYING CONTINUITY OF MAPS

  • Baran, Mehmet;Kula, Muammer;Erciyes, Ayhan
    • Journal of the Korean Mathematical Society
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    • 제46권4호
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    • pp.813-826
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    • 2009
  • In this paper, we generalize the notions of preserving and strongly preserving maps to arbitrary set based topological categories. Further, we obtain characterizations of each of these concepts as well as interprete analogues and generalizations of theorems of Gerlits at al [20] in the categories of filter and local filter convergence spaces.

Inverse Filtering for a Modelling Channel Filter (모델화 채널필터에 대한 인버스필터링)

  • 김성호;주창복
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 한국신호처리시스템학회 2000년도 하계종합학술대회논문집
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    • pp.17-20
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    • 2000
  • In a digital communication system, the transmission channel may introduce error into the digital signal being transmitted. It would be useful if a process could be devised so that the error could be removed in order to recover the transmitted digital signal. We design a corrective filter that is inverse filter, which will generate an output signal identical to the input signal. in order for two systems connected in cascade to produce an output which is identical to the input signal, the over-all unit sample response of the cascade connection must be a unit sample function.

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