• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.689-692
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• 2003

In this paper, we propose an adaptive beamforming algorithm for array antenna. The proposed beamforming algorithm, based on Block LMS (Block - Least Mean Squares) algorithm, has a variable step size from coefficient update. This method shows some advantages that the convergence speed is fast and the calculation time can reduced using a block LMS algorithm from frequency domain. As the adaptive parameter approaches a stationary state, it could reduce the number of filter coefficient update with the help of various step size. In this paper we compared the efficiency of the proposed algorithm with a standard LMS algorithm which is a representative method of adaptive beamforming.

• 대한의용생체공학회:의공학회지
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• 제25권4호
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• pp.315-321
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• 2004

본 연구에서는 다채널 디지털 보청기에서 적용될 수 있는 적응 궤환 제거 알고리즘을 제안하였다. 제안된 적응궤환 제거기는 궤환 검출을 위한 적응 노치 필터와 궤환 제거를 위한 NLMS (normalized least mean square) 적응필터로 구성되어 있다. 제안된 적응 궤환 제거 알고리즘을 다채널 보청 알고리듬과 결합하였다. 다채널 보청 알고리즘은 MDCT (modified discrete cosine transform) 필터뱅크를 이용하여 주파수 대역별 청력 손실을 보상하도록 구성하였다. 제안된 알고리즘을 포함한 완성된 보청 알고리즘의 성능을 컴퓨터 시뮬레이션을 통해 평가하였으며 상용 DSP보드를 이용하여 실시간 구현을 확인하였다.

• 한국음향학회지
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• 제33권5호
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• pp.335-340
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• 2014

본 논문에서는 충격성 잡음(impulsive noise) 환경 하에서 고유치 분포가 큰 입력 신호를 다루기 위해 제안된 부호 부밴드 적응 필터(Sign Subband Adaptive Filter, SSAF)의 성능을 향상시키기 위한 새로운 SSAF를 제안하였다. 기존에 제안된 SSAF는 각각의 부밴드 입력 신호를 모든 부밴드 입력 신호의 $l_2-norm$으로 정규화하기 때문에 밴드의 수를 증가시켜도 수렴속도가 향상되지 않는다. 본 논문에서는 부밴드 입력 신호를 각각의 부밴드 입력 신호의 $l_2-norm$으로 정규화하며 밴드의 수를 증가시킴에 따라 수렴속도가 증가하는 새로운 부호 부밴드 적응필터를 제안하였다. 시스템 식별 환경에서 두 알고리즘의 성능을 비교하는 컴퓨터 모의 실험을 수행하여 제안된 알고리즘의 수렴속도가 더 빠름을 보였다.

• 한국음향학회지
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• 제6권2호
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• pp.15-18
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• 1987

승산을 요하지 않는 적응 디지탈 필터링 알고리즘이 논의되었다. 제안된 알고리즘은 델타 변조 디지탈 필터를 사용하였으며 승산없이 적응 디지탈 필터를 실현하기 위하여 필터계수는 SIGN 알고리즘으로 새로이 재조정된다. 결과적으로, 제안된 알고리즘은 단순히 UP/DOWN 계수동작으로 실현될 수 있음을 보였다. 제안된 적응 디지탈 필터링 알고리즘과 다른 알고리즘을 시스템 Identification문제에 적용하여 수렴특성을 조사하였다.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.1222-1227
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• 2005

In this paper, an efficient scheme of the adaptive notch filter is presented for rejecting the narrow bandwidth interferences(NBI) in GPS receiver. Designed is the lattice IIR double notch filter for more efficient suppression of the NBI with less computational complexity. The algorithm is of recursive prediction error form and uses a special constrained model of IIR with a minimal number of parameters. This paper chooses seven different jamming scenarios including one without jamming for evaluating the proposed filter algorithm. The simulation results to the jamming scenarios show that the proposed algorithm adjusts the double notch filter effectively for the given JSR, and provides better SNR than the conventional algorithms. Finally, it is shown that the advantages of the proposed filter algorithm can range as high as JSR 79dB in time domain processing. Also, the ADNF(adaptive double notch filter) guarantees that more than SNR 10dB of GPS receiver can be always maintained. In conclusion, there is enough evidence to believe that the proposed algorithm will perform quite well for removing interference signals.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2004년도 ICCAS
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• pp.1577-1580
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• 2004

This paper proposes a design and implementation of transversal adaptive digital filter using LMS (Least Mean Squares) adaptive algorithm. The filter structure is based on Distributed Arithmetic (DA) which is able to calculate the inner product by shifting and accumulating of partial products and storing in look-up table, also the desired adaptive digital filter will be multiplierless filter. In addition, the hardware implementation uses VHDL (Very high speed integrated circuit Hardware Description Language) and synthesis using FLEX10K Altera FPGA (Field Programmable Gate Array) as target technology and uses Leonardo Spectrum and MAX+plusII program for overall development. The results of this design are shown that the speed performance and used area of FPGA. The experimental results are presented to demonstrate the feasibility of the desired adaptive digital filter.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.182-185
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• 2000

In this paper, we present a new adaptive filter structure which is based on polyphase decomposition of the filter to be adapted. This structure uses wavelet transform to acquire transform-domain coefficients of the input signal. With this coefficients RLS algorithm is used for adaptation. Particularly, using the polyphase parallel structure, we can trace the system which has very long impulse response with only increasing the subband, and show that computational savings can be achieved. The proposed structure was applied to system identification for performance estimation and compared with fullband adaptive filter.

• 한국통신학회:학술대회논문집
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• 한국통신학회 1991년도 추계종합학술발표회논문집
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• pp.119-122
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• 1991

In this paper, a new adaptive IIR notch filter employing all pass filter is proposed. Proposed all pass filter is composed of all pole and all zero sections, each of which utilizes modulation lattice filter [11]. And, adaption algorithm for proposed notch filter is also derived. In addition, the error surface for proposed IIR adaptive notch filter is analyzed. Computer simulation results reveal that the proposed adaptation algorithm works well for low SNR(signal to noise ratio) single and multiple sinusoids. And it is shown that for estimation time varying frequency, the parameter which is related to notch bandwidth is important than any other parameters.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2001년도 ICCAS
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• pp.98.5-98
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• 2001

An efficient adaptive nonlinear compensator, based on a wavelet transform domain adaptive Volterra filter along with a modified escalator algorithm, is proposed to speed up the convergence rate of an adaptive LMS algorithm. In particular, it is well known that the e.g., slow convergence speed of an adaptive LMS algorithm depends on the statistical characteristics (e.g., large eigenvalue spread) of the corresponding auto-correlation matrix of the input vector. To solve such a convergence problem, the proposed approach utilizes a modified escalator algorithm and a wavelet transform domain adaptive LMS Volterra filtering technique, which leads to diagonalization of the auto-correlation matrix of the ...

• 대한전기학회:학술대회논문집
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• 대한전기학회 1995년도 하계학술대회 논문집 B
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• pp.758-760
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• 1995

The purpose of noise cancellation is to estimating signals corrupted by additive noise or interference. In this paper, an adaptive noise canceller is built from a Walsh filter with a new adaptive algorithm. The Walsh filter consists of a Walsh function. Since the Walsh functions are either even or odd functions, the covariance matrix in the tap gain adjustment algorithm can be reduced to a simple form. In this paper, minimization of the mean squre error is accomplished by a proposed adaptive algorithm. The conventional adaptation techniques use a fixed time constant convergence factor by trial and error methods. In this paper, a convergence factor is obtained that is tailored for each adaptive filter coefficient and is updated at each block iteration.