• Proceedings of the KIEE Conference
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• 2003.11c
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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.

• Journal of Biomedical Engineering Research
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• v.25 no.4
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• pp.315-321
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• 2004

In this study, we proposed an adaptive feedback cancellation algorithm for multi-band digital healing aids. The adaptive feedback canceller (AFC) is composed of an adaptive notch filter (ANF) for feedback detection and an NLMS (normalized least mean square) adaptive filter for feedback cancellation. The proposed feedback cancellation algorithm is combined with a multi-band hearing aid algorithm which employs the MDCT (modified discrete cosine transform) filter bank for the frequency-dependent compensation of hearing losses. The proposed algorithm together with the MDCT-based multi-channel hearing aid algorithm has been evaluated via computer simulations and it has also been implemented on a commercialized DSP board for real-time verifications.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.335-340
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• 2014

In this paper, we propose a new sign subband adaptive filter to improve the convergence rate of the conventional sign subband adaptive filter which has been proposed to deal with colored input signal under the environment with impulsive noise. The existing sign subband adaptive filter does not increase the convergence speed by increasing the number of subband because each subband input signal is normalized by $l_2-norm$ of all of the subband input signals. We devised a new sign subband adaptive filter that normalizes each subband input signal with $l_2-norm$ of each subband input signal and increases the convergence rate by increasing the number of subband. We carried out a performance comparison of the proposed algorithm with the existing sign subband adaptive filter using a system identification model. It is shown that the proposed algorithm has faster convergence rate than the existing sign subband adaptive filter.

• The Journal of the Acoustical Society of Korea
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• v.6 no.2
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• pp.15-18
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• 1987

Multiplication free adaptive digital filtering algorithms are discussed. The proposed. The proposed algorithm uses delta modulation digital filter and the relevant filter weights are updated using the SIGN algorithms to realize an adaptive digital filter without multiplication operations. It is shown that the resulting algorithm can be implemented using simple up/down counting operations. The convergence characteristics of the proposed adaptive digital filtering algorithm and .others are investigated for a system identification problem.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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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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• 2004.08a
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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.

• Proceedings of the IEEK Conference
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• 2000.06d
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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.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1991.10a
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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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• 2001.10a
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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 ...

• Proceedings of the KIEE Conference
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• 1995.07b
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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.