• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.20 no.10
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• pp.35-40
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• 2006

In this paper, a current-mode analog variable finite-impulse-response (FIR) filter with variable tap coefficient circuits is proposed for frequency selective wireless personal area network(WPAN) system or terminals. From the circuit simulation the operation of the 7-tap FIR filter is confirmed. The 0.0625-step tap coefficient circuit is designed and fabricated with $0.8[{\mu}m]$ CMOS technology. The proposed FIR filter has a variable length of taps and variable coefficients, so it has a potential for being used to frequency selective WPAN system or frequency selective wireless communication terminals.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.35-44
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• 2004

We Propose the methods to design the finite impulse response (FIR) and the infinite impulse response (IIR) lattice filters using Schur algorithm through the spectral factorization of the covariance matrix by circulant matrix factorization (CMF). Circulant matrix factorization is also very powerful tool used for spectral factorization of the covariance polynomial in matrix domain to obtain the minimum phase polynomial without the polynomial root finding problem. Schur algorithm is the method for a fast Cholesky factorization of Toeplitz matrix, which easily determines the lattice filter parameters. Examples for the case of the FIR filter and for the case of the In filter are included, and performance of our method check by comparing of our method and another methods (polynomial root finding and cepstral deconvolution).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.12
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• pp.2386-2396
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• 1994

In this paper, an optimal algorithm for the design of 1-D FIR filters with finite precision coefficients is proposed. The algorithm is based on the observation that the frequency constraints of a sub-problems(SP) in the branch and bound algorithm, which repeatedly use LP(linear programming), are closely related with those of neighboring SPs. By using the relationship between the SPs, the proposed algorithm reduces the number of constraints required for solving each SP with Lp, whereas the conventional algorithm employs all the constraints, which are required for solving the initial problem. Thus, the overall computational load for the design of FIR filters with finite precision coefficients is significantly alleviated, compared to the conventional branch and bound algorithm. Also, a new branching scheme for the design of FIR filters with sum-of-power-of-two(SOPOT) coefficients is proposed. It is shown that the computational load for the design fo SOPT coefficient filters can be further reduced with the new branching scheme.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.10
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• pp.188-198
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• 2013

Maximally flat (MAXFLAT) half-band filters usually have wider transition band than other filters. This is due to the fact that the maximum possible number of zeros at $z={\pm}1$ is imposed, which leaves no degree of freedom, and thus no independent parameters for direct control of the frequency response. This paper describes a novel method for the design of FIR halfband filters with an explicit control of the transition-band width. The proposed method is based on a generalized Lagrange halfband polynomial (g-LHBP) with coefficients parametizing a 0-th coefficient $h_0$, and allows the frequency response of this filter type to be controllable by adjusting $h_0$. Then, $h_0$ is modeled as a steepness parameter of the transition band and this is accomplished through theoretically analyzing a polynomial recurrence relation of the g-LHBP. This method also provides explicit formulas for direct computation of design parameters related to choosing a desired filter characteristic (by trade-off between the transition-band sharpness and passband & stopband flatness). The examples are shown to provide a complete and accurate solution for the design of such filters with relatively sharper transition-band steepness than MAXFLAT half-band filters.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.1
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• pp.87-96
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• 2002

This paper proposes an efficient decimation filter structure in programmable down converter for software radio. The decimation filter consists of the cascaded integrator-comb(CIC) filter, a compensation filter for CIC, cascaded comb and modified halfband filters, and programmable FIR filter. Since the compensation filter is used in CIC, the passband drooping is compensated and stopband attenuation is improved. Therefor the more decimation can be implemented in CIC filter. The compensation filter in CIC reduced the computational complexity of other decimation filters and the coefficients of PFIR, thereby achieving a significant hardware reduction over existing approaches. We can reduce the multiply operator by 20% in hardware and operation by 50% as compared with PDC of Harris.

• The Journal of the Acoustical Society of Korea
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• v.12 no.4
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• pp.32-38
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• 1993

2-D FIR 필터를 시스토릭 어레이 구조로 실현하는 방법을 제시하였다. 시스토릭 어레이는 1-D FIR 필터로 부분 실현한 후 병렬연겨랗여 구현하였다. 부분 실현한 시스토릭 어레이의 마지막 입력신호를 다음 단의 입력에 직접연결시킴으로써 입력 지연에 사용되는저장요소를 절약 시킨다. 1-D 시스ㅏ토릭 어레이는 지역통신 접근에 의해 DG를 설계한 후 SFG로으ㅟ 사상을 통해 유도하였다. 유도된 SFG는 DG의 노드가 보다 적은수의 PE에 사상됨으로써 PE의 이용률을 개선할 수 잇다. 유도된 구조는 매우 간단하며, 입력 샘플이 공급되어지면 매 샘플링 기간마다 새로운 출력을 얻는 매우 SHB은 데이터 비율(data rate)을 갖는다. 시스토릭 어레이는 규칙적이고, 모듈성이며, local interconnection, highly synchronized multiprocessing 의 특징을 갖기 때문에 VLSI 실현에 매우 적합하다. PE 셀 구조는 높은 처리율, 최소 계산시간과 최소 파이프라인 주기를 갖도록 설계하였다.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.567-570
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• 1988

This paper describes the design of FIR filtering DSP-chip that can be operated without programming. The proposed DSP-chip has not only the improvement of execution time but also selectivity of filter length from N=1 to N=128. Hilbert Transformer can be designed from this chip. FIR filter system is composed of Data memory/Control Unit, external memory and multiplier-accumulator. Data memory/Control Unit is laid out in this paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.9
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• pp.775-784
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• 1990

In this paper, a new window model for designing FIR digital filter is suggested. For the suggested model, two parameters are established and their effects to the characteristics of filter are analyzed. As a result, one of these parameters improves the characteristics of the transient region but degenerates those of peak sidelobe a little bit, and the other contrasts the characteristics with the former. These effects of parameters give selectivity to design window. Therefore the new suggested window model gives flexibility to filter design and its improved characteristics is comfirmed by simulation.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1986.04a
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• pp.85-88
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• 1986

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.295-303
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• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).