• 한국음향학회지
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• 제19권8호
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• pp.26-34
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• 2000

This paper presents a new adaptive impulse noise cancelling technique based on the adaptive nonlinear suppressing function. The proposed "adaptive threshold algorithm (ATA)" is controlled by the normalized power prior input data term, and this adaptive threshold makes the cancelling system highly robust against additive impulse noise. For the performance evaluation, we have tested the proposed algorithm with the observed signals simulated in various impulsive noise environments and real EMG signals. As a result the proposed algorithm shows superior performance of 51.7% to the available techniques in the points of SNR and MSE.

• 한국통신학회논문지
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• 제22권4호
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• pp.696-705
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• 1997

This paper presents an efficient adaptive predistortion technique compensating linear and nonlinear distortions caused by high-power amplifier (HPA) with memory in OFDM systems. The efficient adaptive data predistortion techniques proposed for compensation of HPA with memory in single carrier systems cannot be applied to OFDM systems since the possible input levels for HPA is infinite in OFDM systems. Also, previous adaptive predistortion techniques, based on Volterra series modeling, are not suitable for real-time implementation due to high computational burden and slow convergence rate. In the proposed approach, the memoryless HPA preceded by a linear filter in OFDM systems is modeled by the Wiener system which is then precompensated by the proposed adaptive predistorter with a minimum number of filter taps. An adaptive algorithm for adjusting the proposed adaptive predistorter is derived using the stochastic gradient method. It is demonstrated by computer simulation that the performance of OFDM system suffering from nonlinear distortion can be greatly improved by the proposed efficient adaptive predistorter using a small number of filter taps.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2014년도 하계학술대회
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• pp.179-180
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• 2014

The main goal of this paper is to present an adaptive filter system using NLMS(Normalized Least mean square) adaptive algorithm for noise cancellation. The proposed algorithm has less computational complexity and better convergence property than the former algorithms like spectral subtraction algorithm, etc. We use TIMIT criterion voice and Noisex-92 for the experiment. The experimental result shows the feasibility of our algorithm for filtering noise from voice effectively.

• The Journal of the Acoustical Society of Korea
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• 제21권3E호
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• pp.119-125
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• 2002

In the conventional generalized side-lobe canceller (GSC), the output is an estimated error signal that causes the adaptive filter weights do not converge to the optimal value. This paper presents a new side-lobe canceller with adaptive compensator th it reduces the misadjustment of the adaptive filter coefficients for the structural problem in the GSC. The adaptive compensator separates the output signal from the estimated error. The newly estimated error signal converges to the zero while the output signal tracks the target signal. This paper shows improvement of the performance by comparing the computer simulation of the output signal of GSC with the output signal of the proposed algorithm.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1996년도 추계학술대회 논문집 학회본부
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• pp.55-57
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• 1996

This paper presents an implementation of the adaptive ANC (Active Noise Control) system improving robustness. The system using the proposed algorithm shows a good performance of control, when the adaptive filter well does not work. We construct a real duct system and use DSP chip for experiment. Experimental results of the proposed algorithm prove the system to be superior than the conventional filtered-x LMS algorithm when the adaptive filter is out of order.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2007년도 추계종합학술대회
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• pp.109-112
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• 2007

디지털 카메라의 출현과 Mobile 장비에서의 카메라 적용으로 인하여 디지털화된 이미지의 화질개선이 지속적으로 연구되고 있다. 특히, 센서로부터 입력된 이미지는 영상으로 출려되기 전 ISP(Image Signal Process) 과정을 거치게 되는데, 이 단계에서 이미지는 고주파 성분의 Noise 제거를 위한 LPF(Low Pass Filter)에 의해 고역의 주파수 성분이 상쇄되는 결과를 가진다. 이에 본 논문에서는 LPF(Low Pass Filter)에 의해 고역의 주파수 성분이 상쇄되는 결과를 가진다. 이에 본 논문에서는 LPF에 의해 Blurring된 이미지를 윤곽선 검출 알고리즘을 사용하지 않고, 이미지 윤곽선이 가질 수 있는 다양한 상태를 고려하여 적절한 계수를 가지는 Adaptive-HPF(High Pass Filter)를 사용함으로써 더욱 선명한 영상을 출력하는 알고리즘을 제안한다. 제안된 알고리즘의 하드웨어 구현시 Total Gate Count는 8700여 개로 Mobile 장치에 적용될 수 있다는 것을 검증하였다.

• 한국멀티미디어학회:학술대회논문집
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• 한국멀티미디어학회 2001년도 추계학술발표논문집
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• pp.46-50
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• 2001

Adaptive signal processing is used to acoustic echo canceller. adaptive noise canceller and adaptive algorithm among adaptive algorithms is mainly used because the structure is simple and computa LMS algorithm has trade-off between the converge speed and the steady state error. In this paper, step-size of adaptive algorithm is varied with orthogonality Principles of optimal filter to get fasts though small steady state error. Time varying step-size is determined proportional to the maximum vector of LMS algorithm. As results of simulations, the adaptive algorithm with proposed time-v compared with conventional ones.

• International Journal of Control, Automation, and Systems
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• 제2권3호
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• pp.310-318
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• 2004

In this paper, an unscented Kalman filter (UKF) for curvilinear motions in an interacting multiple model (IMM) algorithm to track a maneuvering vehicle on a road is investigated. Driving patterns of vehicles on a road are modeled as stochastic hybrid systems. In order to track the maneuvering vehicles, two kinematic models are derived: A constant velocity model for linear motions and a constant-speed turn model for curvilinear motions. For the constant-speed turn model, an UKF is used because of the drawbacks of the extended Kalman filter in nonlinear systems. The suggested algorithm reduces the root mean squares error for linear motions and rapidly detects possible turning motions.

• Journal of Power Electronics
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• 제17권1호
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• pp.149-160
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• 2017

To improve the performance of sensorless induction motor (IM) drives, an adaptive speed estimation method based on a strong tracking extended Kalman filter with a least-square algorithm (LS-STEKF) for induction motors is proposed in this paper. With this method, a fading factor is introduced into the covariance matrix of the predicted state, which forces the innovation sequence orthogonal to each other and tunes the gain matrix online. In addition, the estimation error is adjusted adaptively and the mutational state is tracked fast. Simultaneously, the fading factor can be continuously self-tuned with the least-square algorithm according to the innovation sequence. The application of the least-square algorithm guarantees that the information in the innovation sequence is extracted as much as possible and as quickly as possible. Therefore, the proposed method improves the model adaptability in terms of actual systems and environmental variations, and reduces the speed estimation error. The correctness and the effectiveness of the proposed method are verified by experimental results.

• 한국인터넷방송통신학회논문지
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• 제12권1호
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• pp.195-200
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• 2012

본 논문에서는 동시통화에 강인한 특성을 갖는 알고리즘의 추적성능을 개선시킬 수 있는 새로운 방법을 제안하였다. 반향경로의 변화를 검출하는 방법과 적응필터의 계수 적응식을 변경하는 방법을 제안하였다. 오차신호 대 스케일 파라메터의 비는 반향경로가 변화한 경우와 동시통화가 발생한 경우에 지속시간이 다른 특성을 가지며 이를 이용하여 반향경로 변화를 검출한다. 제안한 적응필터계수 적응식은 잘못된 오차신호선택을 방지하도록 하여 추적 성능을 개선시킨다. 실제 음성신호와 ITU-T G.168에서 제공하는 반향경로를 사용한 실험을 통하여 제안한 방법이 기존의 방법에 비해 4dB 이상 성능을 개선시킴을 확인하였다.