• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2009년도 추계학술대회 논문집
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• pp.231-239
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• 2009

The filtered-x LMS (FX-LMS) algorithm has been applied to the active noise control (ANC) system in an acoustic duct. This algorithm is designed based on the FIR (finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filteredu LMS algorithm (FU-LMS) based on infinite impulse response (IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• 한국소음진동공학회논문집
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• 제22권2호
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• pp.146-155
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• 2012

The filtered-x LMS(FX-LMS) algorithm has been applied to the active noise control(ANC) system in an acoustic duct. This algorithm is designed based on the FIR(finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filtered-u LMS algorithm(FU-LMS) based on infinite impulse response(IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 1998년도 추계종합학술대회
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• pp.574-577
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• 1998

In this paper, a new simple and efficient technique to improve the convergence speed of LMS algorithm is proposed in an interference-limited multi-path fading environment as encountered in indoor wireless communications. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. The theoretical analysis shows that the data-recycling in technique can increase convergence speed by (B+1) times without increasing the computational complexity substantially where B is the number of recycled data. The results of the computer simulation demonstrate that the simulation results are in accordance with the theoretical analysis and the superiority of the filter algorithm.

• 스마트미디어저널
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• 제9권3호
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• pp.59-70
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• 2020

This paper presents a fully automatic tool to recognize the liver region from CT images based on a deep learning model, namely Multiple Filter U-net, MFUnet. The advantages of both U-net and Multiple Filters were utilized to construct an autoencoder model, called MFUnet for segmenting the liver region from computed tomograph. The MFUnet architecture includes the autoencoding model which is used for regenerating the liver region, the backbone model for extracting features which is trained on ImageNet, and the predicting model used for liver segmentation. The LiTS dataset and Chaos dataset were used for the evaluation of our research. This result shows that the integration of Multiple Filter to U-net improves the performance of liver segmentation and it opens up many research directions in medical imaging processing field.

• 한국통신학회논문지
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• 제36권6C호
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• pp.343-348
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• 2011

이 논문은 NFxLMS (normalized filtered-x least mean square) 적응 필터기반의 실시간 ANC (active noise control) 무선 헤드셋에 관해서 연구한다. RMS (root mean square) 지연 분포를 측정한 후 채널을 보정해서 학습시간을 줄이고, 학습 시간동안 NFxLMS 필터 계수를 갱신해서 잡음 제거 필터의 수렴속도를 개선하는 방법을 제안한다. 제안한 방법을 실제 잡음 환경에서 이용할 경우에 짧은 학습 시간과 빠른 수렴속도를 가지면서 기존 잡음 제거기와 비슷한 성능을 가지는 잡음 제거기를 구성할 수 있다.

• 한국전기전자재료학회논문지
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• 제25권12호
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• pp.979-983
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• 2012

We proposed the ZnO thin film for a SAW filter by PLD and RF sputtering method. ZnO thin films was pre-deposited on a sapphire substrate as a seed layer by PLD method and then deposited on seed layer by RF sputtering. The surface characteristics of ZnO thin film were investigated by XRD, SEM and AFM. The minimum surface roughness was 1.92 nm and FWHM of rocking curve was $0.92^{\circ}$. We demonstrated the SAW filter with bandwidth of approximately 0.97 MHz and the center frequency of 18.72 MHz using the proposed ZnO thin film.

• 전기학회논문지
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• 제59권3호
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• pp.669-673
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• 2010

This paper proposes a new active noise control (ANC) system for canceling broadband noise. The proposed ANC system is designed based on the multiband-structured delayless subband adaptive filter (MDSAF), which has advantages of fast-convergence speed and higher noise reduction performance by eliminating the aliasing and band-edge effects caused by band-partitioning. The simulation results show that the proposed ANC system has faster convergence speed as compared to the conventional ANC systems and effectively reduces the wideband noise.

• Journal of the Korean Society for Industrial and Applied Mathematics
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• 제22권2호
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• pp.115-123
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• 2018

This paper discusses extended Kalman filter method for solving learning problems of multilayered neural networks. A lot of learning algorithms for deep layered network are sincerely suffered from complex computation and slow convergence because of a very large number of free parameters. We consider an efficient learning algorithm for deep neural network. Extended Kalman filter method is applied to parameter estimation of neural network to improve convergence and computation complexity. We discuss how an efficient algorithm should be developed for neural network learning by using Extended Kalman filter.

• Journal of applied mathematics & informatics
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• 제29권3_4호
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• pp.573-585
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• 2011

SQP method is one of the most important methods for solving nonlinear programming. But it may fail if the quadratic subproblem is inconsistent. In this paper, we propose a modified nonmonotone filter trust region method in which the QP subproblem is consistent. By means of nonmonotone filter, this method has no demand on the penalty parameter which is difficult to obtain. Moreover, the restoration phase is not needed any more. Under reasonable conditions, we obtain the global convergence of the algorithm. Some numerical results are presented.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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• pp.240-240
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• 2000

This paper presents two algorithms based on the concept of the frequency domain adaptive filter(FDAF). First the frequency domain recursive least squares(FRLS) algorithm with the overlap-save filtering technique is introduced. This minimizes the sum of exponentially weighted square errors in the frequency domain. To eliminate discrepancies between the linear convolution and the circular convolution, the overlap-save method is utilized. Second, the sliding method of data blocks is studied Co overcome processing delays and complexity roads of the FRLS algorithm. The size of the extended data block is twice as long as the filter tap length. It is possible to slide the data block variously by the adjustable hopping index. By selecting the hopping index appropriately, we can take a trade-off between the convergence rate and the computational complexity. When the input signal is highly correlated and the length of the target FIR filter is huge, the FRLS algorithm based on the block processing technique has good performances in the convergence rate and the computational complexity.