• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.3
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• pp.105-112
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• 2007

In general, the earlier methods for the design of MAXFLAT FIR filters have existent problems due to the approximation algorithms used to approach MAXFLAT(maximally flat) response in the passband and the stopband.. The proposed approach advanced by using of MAXSHCUT(maximally sharp cutoff) condition in this paper clearly overcomes these problems. In this approach, we use a key parameter represented with filter-order and cutoff-frequency parameters for obtaining the lowpass filters with the MAXFLAT and MAXSHCUT characteristics in the frequency domain. Consequently, this design technique leads to new MAXFLAT and MAXSHCUT FIR digital filter, which can achieve sharp-cutoff responses with the stopband attenuation exceeding 100 dB almost everywhere.

• 제어로봇시스템학회:학술대회논문집
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• 1987.10b
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• pp.470-472
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• 1987

This paper deals with the continuous-discrete estimation problem using FIR filters and performs modeling error analyses of the FIR filters, compared to Kalman filter and the limited memory filters, via computer simulations. It is shown that, the less driving noise the system has, the better performance the FIR filter exhibits and that this characteristic appears rare distinctly in nonlinear system than in linear systems.

• Journal of Electrical Engineering and Technology
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• v.4 no.3
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• pp.423-428
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• 2009

In FIR (Finite Impulse Response) filter applications, Nth-band FIR digital filters are known to be important due to their reduced computational requirements. The conventional methods for designing FIR filters use iterative approaches such as the well-known Parks-McClellan algorithm. The Parks-McClellan algorithm is also used to design Nth-band FIR digital filters after Mintzer's research. However, a disadvantage of the Parks-McClellan algorithm is that it needs a large amount of design time. This paper describes a direct design method for Nth-band FIR Filters using Chebyshev polynomials, which provides a reduced design time over indirect methods such as the Parks-McClellan algorithm. The response of the resulting filter is equi-ripple in passband. Our proposed method produces a passband response that is equi-ripple to within a minuscule error, comparable to that of Mintzer's design method which uses the Parks-McClellan algorithm.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.1
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• pp.49-55
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• 2003

The use of digital filters in the signal process field is increasing rapidly with development of the modern industrial society. Especially, detail processors, Y/C separators, ghost removing filters, standard converters (NTSC to PAL or PAL to NTSC) and noise reducers, all of which use digital filters, tend to be used in digital video and audio processing, CATV and various communication fields. Generally, there are two different digital filters, the Rf (infinite impulse response) filter and the FIR (finite impulse response) filter in digital filter. In this paper, we have designed FIR filter which has the phase linearity and the easiness of creation. In the design of the FIR digital filter, the window function is used to alleviate the ripples caused by Gibbs Phenomenon around the cut off frequency of the band pass. But there're some problems to choose proper window function for the design destination due to its fixed values. Therefore, in this paper, we designed a modified Hanning window with new parameter which is adaptively chosen corresponding to design objectives. The digital filter was simulated to prove the validity of the model and it was compared with the Hamming, the Manning, the Blacknan and the Kaiser window function. And we have used peak side-lobe and transient characteristics as standard of judgement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4A
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• pp.324-329
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• 2002

In the high-speed/low-power digital filter applications like wireless communication systems, canonical signed digit(CSD) linear phase finite impulse response(FIR) filter structures are widely investigated. In this paper, we propose a high-speed/low-power CSD linear phase FIR filter structure using vertical common sub-expression. In the conventional linear phase CSD filter, horizontal common sub-expressions are utilized due to the inherent horizontal common sub-expression of symmetrical filter coefficients. We use the fact that their MSBs are also equal since adjacent filter coefficients have similar values in the linear phase filter Through the examples, it is shown that our proposed structure is more efficient in case that precision of implementation is lower, and tap length are longer.

• The Journal of the Acoustical Society of Korea
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• v.4 no.2
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• pp.3.2-3.2
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• 1985

본 논문은 전 전자식 교환기에 사용되는 MFC 수신기를 설계하기 위해서 FIR-Filter를 사용한 논문이다. FIR-filter를 설계함에 있어서, Remetz Algorithm을 사용한 Mellellan의 방식이 적용되었으며 인접주파수를 정확히 구분하기 위하여 85차의 고차필터를 설계하였고, 필터의 출력 데이터 15개의 합으 로써 각 주파수에 대한 판별값으로 정하였다. 인접 주파수를 구분하는 출력 power의 결과는 최근의 FGT-MFC 수신기의 출력 power의 결과와 비교 하였을 때, S/N 비면에서 5dB 이상의 향상을 보였으 며, 133개의 입력데이타를 사용하는 FGT-MFC 수신기보다 4.2ms 빠른 응답특성을 나타냈다.

• 제어로봇시스템학회:학술대회논문집
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• 1988.10a
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• pp.172-177
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• 1988

In this paper, a new fast algorithm of FIR least squares filter with linear phase is presented. The general unknown statistics case is considered, whereby only sample records of the data are available. Taking advantage of the near-to-Toeplitz+Hankel structure of the resulting normal equation, a fast algorithm which gurantees the linear phase constraint, is developed that recursively produces the filter coefficient of linear phase FIR filter for a single block of data.

• The Journal of the Acoustical Society of Korea
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• v.4 no.3
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• pp.3-7
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• 1985

본 논문은 전 전자식 교환기에 사용되는 MFC 수신기를 설계하기 위해서 FIR-Filter를 사용한 논문이다. FIR-filter를 설계함에 있어서, Remetz Algorithm을 사용한 Mellellan의 방식이 적용되었으며 인접주파수를 정확히 구분하기 위하여 85차의 고차필터를 설계하였고, 필터의 출력 데이터 15개의 합으 로써 각 주파수에 대한 판별값으로 정하였다. 인접 주파수를 구분하는 출력 power의 결과는 최근의 FGT-MFC 수신기의 출력 power의 결과와 비교 하였을 때, S/N 비면에서 5dB 이상의 향상을 보였으 며, 133개의 입력데이타를 사용하는 FGT-MFC 수신기보다 4.2ms 빠른 응답특성을 나타냈다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.250-255
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the frequency domain algorithm is prefer than the existent time domain. we analyzed the Wavelet algorithm, short-length fast running FIR algorithm, fast-short-length fast running FIR algorithm and proposed algorithm.

• Journal of Institute of Control, Robotics and Systems
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• v.1 no.1
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• pp.20-24
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• 1995

In this paper we analyze the performance bounds of the optimal FIR filter in continuous time systems with modeling uncertainty. The performance bounds are presented by the estimation error convariance and they are here expressed by the upper bounds of the difference of the estimation error covariance between the real and nominal values in case of the system with model uncertainties whose upper bounds are imperfrctly known a priori. The performance bounds of the optimal FIR filter are compared with those of the Kalman filter via a numerical example applied to the estimation of the motion of an aircraft carrier at sea, which shows the former has better performances than the latter.