• Title/Summary/Keyword: Adaptive noise estimation

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Two-Step Suboptimal Filters for Linear Dynamic Systems

  • Ahn, Jun-Il;Minhas, Rashid;Shin, Vladimir
    • 제어로봇시스템학회:학술대회논문집
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    • 2005.06a
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    • pp.16-21
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    • 2005
  • This paper considers the problem of state estimation in linear continuous-time systems with multi-sensor environment and observation uncertainties. We propose two suboptimal filtering algorithms for these types of systems. The filtering algorithms consist of two steps: The local optimal Kalman estimates are computed at the first step. And, these local estimates are lineally fused at the second step. The implementation of the two-step filtering algorithms needs a lower memory demand than the optimal Kalman and adaptive Lainiotis-Kalman filters. In consequence of parallel structure of the proposed filters, the parallel computers can be used for their design. The examples exhibit the effect of common noise on the performance of fusion of the local Kalman estimates based on observations from different sensors and in the presence of uncertainties.

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Estimation of Cavity Vibration Frequency in Air Tubes Using Adaptive Filter (적응 필터를 이용한 관내의 공동진동주파수 추정)

  • Yang, Dong-Sung;Su, Sung-Dae;Nam, Hyun-Do
    • Proceedings of the KIEE Conference
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    • 2001.07d
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    • pp.2508-2510
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    • 2001
  • Frequency of cavity vibration in air flowing tube is closely related to a velocity of air. In this research, an instrumentation system to estimate frequency of cavity vibration for measurement of the velocity and quantity of a moving fluid is implemented by using DSP TMS320C32. Measurement of the generated sound wave frequency in cavity is difficult because of environmental noise. Adaptive filters are used to eliminate this noise effectively. The estimated velocity and quantity of a moving fluid by proposed system is compared with the results measured by a standard flow meter.

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On-line Fundamental Frequency Tracking Method for Harmonic Signal and Application to ANC (조화신호의 실시간 기본 주파수 추종 방법과 능동소음제어에의 응용)

  • Kim, Sun-Min;Park, Young-Jin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2000.06a
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    • pp.263-268
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    • 2000
  • In this paper, a new indirect feedback active noise control (ANC) scheme based on the fundamental frequency estimation is proposed for systems with a harmonic noise. When reference signals necessary for feedforward ANC configuration is difficult to obtain, the conventional ANC algorithms for multi-tonal noise do not measure the reference signals but generate them with the estimated frequencies. However, the beating phenomena, in which certain frequency components of the noise vanish intermittently, may make the adaptive frequency estimation difficult. The confusion in the estimated frequencies due to the beating phenomena makes the generated reference signals worthless. The proposed algorithm consists of two parts. The first part is a reference generator using the fundamental frequency estimation and the second one is the conventional feedforward control. We propose the fundamental frequency estimation algorithm using decision rules, which is insensitive to the beating phenomena. In addition, the proposed fundamental frequency estimation algorithm has good tracking capability and lower variance of frequency estimation error than that of the conventional cascade ANF method. We are also able to control all interested modes of the noise, even which cannot be estimated by the conventional frequency estimation method because of the poor SIN ratio. We verify the performance of the proposed ANC method through simulations for the measured cabin noise of a passenger ship and the measured time-varying engine booming noise of a passenger vehicle.

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Active Control of Harmonic Signal Based on On-line Fundamental Frequency Tracking Method (실시간 기본주파수 추종방법에 근간한 조화 신호의 능동제어)

  • 김선민;박영진
    • Journal of KSNVE
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    • v.10 no.6
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    • pp.1059-1066
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    • 2000
  • In this paper. a new indirect feedback active noise control (ANC) scheme barred on the fundamental frequency estimation is proposed for systems with a harmonic noise. When reference signals necessary for feedforward ANC configuration are difficult to obtain, the conventional ANC algorithms for multi-tonal noise do not measure the reference signals but generate them with the estimated frequencies.$^{(4)}$ However, the beating phenomena, in which certain frequency components of the noise vanish intermittently, may make the adaptive frequency estimation difficult. The confusion in the estimated frequencies due to the beating phenomena makes the generated reference signals worthless. The proposed algorithm consists of two parts. The first part is a reference generator using the fundamental frequency estimation and the second one is the conventional feedforward control. We propose the fundamental frequency estimation algorithm using decision rules. which is insensitive to the beating phenomena. In addition, the proposed fundamental frequency estimation algorithm has good tracking capability and lower variance of frequency estimation error than that of the conventional cascade ANF method.$^{(4)}$ We are also able to control all interested modes of the noise, even which cannot be estimated by the conventional frequency estimation method because of the poor S/N ratio. We verify the performance of the proposed ANC method through simulations for the measured cabin noise of a passenger ship and the measured time-varying engine booming noise of a passenger vehicle.

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Secondary Path Estimation Algorithm Based on Residual Music Canceller for Noise Cancelling Headphone (노이즈 캔슬링 헤드폰에 적합한 잔여 음악 제거기 기반의 2차 경로 추정 알고리즘)

  • Ji, Youna;Lee, Keunsang;Park, Youngcheol
    • The Journal of the Acoustical Society of Korea
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    • v.34 no.5
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    • pp.377-384
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    • 2015
  • An active noise control (ANC) algorithm for noise canceling headphone is proposed. In this study, the feedback ANC operated with the filtered-x least mean square algorithm (FxLMS) algorithm is used to attenuate the undesired noise. Also an adaptive residual music canceller (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Simulation results show that a high quality of music sound can be consistently achieved in a time-varying secondary path situation.

An Autoregressive Parameter Estimation from Noisy Speech Using the Adaptive Predictor (적응예측기를 이용하여 잡음섞인 음성신호로부터 autoregressive 계수를 추산하는 방법)

  • Koo, Bon-Eung
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.3
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    • pp.90-96
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    • 1995
  • A new method for autoregressive parameter estimation from noisy observation sequence is presented. This method, termed the AP method, is a result of an attempt to make use of the adaptive predictor which is a simple and reliable way of parameter estimation. It is shown theoretically that, for noisy input, the parameter vector computed from the prediction sequence is closer to that of the original sequence than the noisy input sequence is, under the spectral distortion criterion. Simulation results with the Kalman filter as a noise reduction filter and real speech data supported the theory. Roughly speaking, the performance of the parameter set obtained by the AP method is better than noisy one but worse than the EM iteration results. When the simplicity is considered, it could provide a useful alternative to more complicated parameter estimation methods in some applications.

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Adaptive Channel Estimation Algorithm for DVB-T (DVB-시스템을 위한 적응형 채널 추정 알고리즘)

  • Kim, Seung-Hwan;Lee, Jin-Beom;Lee, Jin-Yong;Kim, Young-Lok
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.6A
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    • pp.676-684
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    • 2008
  • In digital video broadcasting-terrestrial (DVB-T), which is the European digital terrestrial television standard, the orthogonal frequency division multiplexing (OFDM) has been adopted for signal transmission. The main reasons using OFDM are to increase the robustness against the frequency selective fading and impulse noise, and to use available bandwidth efficiently. However, channel variation within an OFDM symbol destroys orthogonality between subcarriers, resulting in inter-carrier interference (ICI), which increases an error floor in proportional to maximum Doppler spread. This paper provides an ICI analysis in both time and frequency domains while existing literatures analyze the ICI effects mainly in frequency domain and proposes the algorithms that estimate the channel impulse response and channel variation using least square (LS) algorithm which is the most simple channel estimation technique. And we propose adaptive channel estimation algorithm that estimates the velocity of terminals. The simulation results show that proposed algorithm has similar performance with about 1.5% computational complexity of noise and ICI reduction LS algorithm in low speed environments.

A study on the DoA Estimation Prformance of Interference Signal in W-CDMA using 3D Adaptive Array Antenna (W-CDMA에서 3차원 적응 배열 안테나를 이용한 방해 신호의 도래각 추정 성능에 관한 연구)

  • Lim, Seung-Gag;Kang, Dae-Soo
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.9 no.1
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    • pp.11-17
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    • 2009
  • This paper deals with the estimation performance DoA (Direction of Arrival) using the 3D adaptive array antenna in W-CDMA signal. For this, we proposed the 5 types of 3D array antenna, and appied for the MUSIC in order to the estimation of DoA algorithm for that antenna, commonly, then the DoA estimation error was used for the estimation performance. In the mobile communication and radio positioning service, performing the spatial filtering after the DoA estimation in array antenna, the quality of receiving signal can improve by the nulling or minimization of interfering signal which is from the undesired direction and the forming of beam which is from the desired direction. The result of DoA estimation and the DoA estimation error by varying the signal to noise ration and the number of interfering signal and power of each type antenna was calculated by computer simulation. As a result of simulation, the other propose antenna has good performance excluding the stack 3D antenna in DoA estimation and the Curved-B type antenna has more superior performance when increasing the number of interfering signal and power in the DoA estimation error.

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Active Tonal Noise Control to Reduce the Low Frequency Tonal Sound (저주파 순음소음저감을 위한 능동 순음 소음제어)

  • 나희승;박영진
    • Journal of KSNVE
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    • v.8 no.6
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    • pp.1037-1042
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    • 1998
  • 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. We also Propose a novel control algorithm (AFC/CAFC) for tonal noise cancellation. The proposed algorithm estimates the magnitude and phase of the tonal noise. The algorithm uses the steepest descent method for the phase/magnitude estimation. Performances of tile CAFC algorithm are presented in comparison with those by the AFC algorithm based on computer simulations and experiments.

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A Kalman Filtering Method for Estimation of Parameters of High Frequency Trans (고주파 과도신호의 파라미터 추정을 위한 칼만 필터링 기법에 관한 연구)

  • Lee, Tae-Hoon;Park, Jin-Bae;Yoon, Tae-Sung;Kho, Jae-Won
    • Proceedings of the KIEE Conference
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    • 1998.07b
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    • pp.620-622
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    • 1998
  • This paper presents a method for estimating parameters of high frequency transient signals when noise is added. The parameters to be estimated are the magnitude, frequency, and decay rate of the signals. An approach based on only the extended Kalman filter (EKF) is highly dependent on choosing a correct value of variance of noise. The proposed method adopts an adaptive Kalman filter (AKF). Having very little information of the noise, This method avoids deterioration of the filter performance caused by choosing an inaccurate variance of the noise. The dependence of the EKF method upon the noise variance and the efficiency of the AKF method are shown.

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