• The Journal of the Acoustical Society of Korea
• /
• v.16 no.6
• /
• pp.85-93
• /
• 1997

Fixed systems are limited in their performances to achieve the more complicated and higher level operations. Accordingly adaptive system, which adjusts itself in accorance with the time-varying environments, has been introduced to camouflarge the defficiency of fixed systems in varying environment, and adaptive filter is the outstanding fields in adaptive system. Adaptive recursive filter is far more efficient in that it can perform the signal processing with relatively lower filter order, but there remains severe problem in stability(convergency). On the basis of hyperstability introduced by V.M. Popov, a hyperstable new adaptive recursive filter is introduced which is theoretically guaranteed in stability. In this paper a more stable algorithm for adaptive recursive filter is devised.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.4C
• /
• pp.377-385
• /
• 2010

In this paper, a general and explicit technique is presented for determining the filter coefficients of maximally flat (MAXFLAT) FIR filter with the magnitude response exactly passing through a prescribed cutoff point. This technique is based on a general formula (i.e. impulse response) with an arbitrary cutoff point and permits direct computation of the coefficients of this filter type with a specified cutoff point. The technique provides an explicit method for choosing the order of flatness of the filter with the specified cutoff point. Also, in the paper, it is shown to give a computationally efficient and accurate solution to the design of the filters with the desired cutoff point.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• v.9 no.1
• /
• pp.635-638
• /
• 2005

The curve shape of the time fluctuation extracted from the electromagnetic signals is very complex. Thus it is important to decide exactly the signal property such as the inflection point for the observed signal. Usually filters elaborately designed are used to detect the signal characteristics. When the noise is added to the signal, the inflection point can be detected using the observation filter. In this paper we propose the design method for a natural observation filter with optimal filter order to extract a definite inflection point for the case of signals with the mixed noise.

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

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.17 no.11
• /
• pp.1185-1198
• /
• 1992

In this paper, the differential weighted algorithm proposed in order to improve th noise smoothing characteristics of conventional Median filter and FIR-Median Hybrid filter. Performance of some image restoration filter(median filter, FIR-Median Hybird filter, FIR-Median Hybrid filter to proposed differential weighted algorithm) are compared and evaluated on the noise smoothing characteristics and sharp edge conservation characteristics. Test and Real images used in this paper are Lenna and Urological images corrupted by impulse, gaussian, exponential and laplacian noise. Experimental results show that the FIR-Median Hybrid filter applied to the differential weighted algorithm are comparatively superior to others. But the filter orders have increased, the more time consumed to image processing. Hence if the adequate filtering by the type of image is selected. now after a great support will be take consideration into the various parts of application by computer science and of medical image processing.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.10 no.11
• /
• pp.3087-3093
• /
• 2009

Digital filters are indispensible element in digital signal processing area. The performance of digital filter based on adding and multiplying operation, such as computational speed and power consuming is determined by the orders and coefficients of filter which has on effect area of semiconductor chip when it is implemented by VLSI technology. In this research, in order to performance improvement of digital filter, we proposed the algorithm to speed-up the operation of digital filter associated with the minimum signed digit representation of binary number system and method to simplify the digital filter design associated with common subexpression elimination. The performance of proposed method is evaluated by the computational speed and design-simplicity by experimental implemented digital filter on FPGA.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.24 no.8B
• /
• pp.1586-1592
• /
• 1999

In DWMT, CMFB is employed in the synthesis/analysis part. The CMFB uses filters of greater length than the DFT, resulting in reduced interference between the carriers. In addition, the CMFB system is computationally efficient and fast algorithms are available for their implementation. Traditional designs for the prototype filters of CMFB usually involve nonlinear optimizations. Thus the required design time is considerably large even for small filter orders. In this paper, a prototype filter design method for CMFB is presented using optimal window method. The design process is reduced to the optimization of a single parameter and consequently the required design time is much less than those of the existing methods. It is shown that the stopband performance of the proposed method is better than that of the Kaiser window method.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.537-540
• /
• 2000

일반적으로 음향 반향 제거에서 반향의 임펄스 응답이 큰 경우 주파수 영역의 알고리듬은 시간 영역 알고리듬에 비해 긴 임펄스 응답에 따른 많은 계산량과 입력신호의 통계적 특성에 의한 영향을 줄일 수 있다. 그러나 주파수 영역 알고리듬에서는 시간 영역의 신호를 주파수 영역으로 변환시킬 때 필터 차수의 2배의 FFT 연산이 필요하게 되어, 긴 차수로 인한 실행 시간 지연이 발생하고 많은 메모리가 필요하다. 이러한 문제점을 감소시키고 수렴성능을 향상시키기 위한 MDF 알고리듬이 제안되었으나 계산량이 많은 단점이 있고, UMDF와 AUMDF 알고리듬은 계산량은 감소되나 수렴성능이 저하되는 문제점이 있다. 본 논문에서는 기존의 MDF 알고리듬과 거의 동일한 수렴성능을 유지하면서 연산량과 메모리를 줄일 수 있는 수정된 AUMDF 알고리듬을 제안하였으며, 모의 실험을 통해 결과를 확인하였다.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.14 no.7
• /
• pp.3425-3430
• /
• 2013

The fast-growing market in wireless communications has led to the development of multi-standard mobile terminals. In this paper, a frequency-tunable active RC bandpass filter for multi-standards wireless communication system is designed using an active inductor. The conventional bandpass filter design methods employ the high order filter or high quality factor Q to improve the stopband attenuation characteristics and frequency selectivity of the passband. The proposed bandpass filter based on the high Q active inductor has an improved frequency characteristics. The center frequency and gain of the designed bandpass filter is tuned by employing the tuning circuit. We have performed the simulation using TSMC $0.18{\mu}m$ process parameter to analyze the characteristics of the designed active RC bandpass filter. The bandpass filter with Q=20.5 has 90MHz half power bandwidth at the center frequency of 1.86GHz. Moreover, the center frequency of the proposed bandpass filter can be tuned between 1.86~2.38GHz for the multi-standards wireless communication system using the capacitor of the tuning circuit.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
• /
• v.20 no.5
• /
• pp.99-106
• /
• 2006

In this paper, implementation of active mufflers using multiple channel adaptive IIR filter is presented. Usually, recursive LMS(RLMS) algorithms for adaptive IIR filters are highly efficient than filtered-X LMS(FXLMS) algorithms, when the order of both algorithms are the same. However, RLMS algorithms usually diverge before the algorithms arenot yet converged. So, the prefilters are presented to improve the stability by pulling the poles of feedback control transfer function in the beginning of active noise control and returning the original poles after the filters converge. The engine noises of diesel engine automobiles and gasoline engine automobiles are analyzed and the mathematical model of an active muffler is derived. Computer simulations and experiments are performed to show the effectiveness of the proposed systems.