• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.8
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• pp.375-377
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• ETRI Journal
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• v.25 no.5
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• pp.345-355
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• 2003

In this paper, we propose a method for designing a class of M-channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop-band attenuation, which is also taken care of in the proposed algorithm.

• Journal of the Korean Institute of Telematics and Electronics
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• v.24 no.1
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• pp.123-132
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• 1987

In this paper, we studied on the design of elliptic digital filters using the bilinear z transformation method, and proposed a design procedure satisfying prescribed specifications. The magnitude characteristics of digital filters are compared with its of analog filiters by computer simulation. Finaly we considered the quantization effects of digital filters. In cascader realization of fixed-point digital filters under dynamic range constraints, the outpout noise for IIR digital filters depends on the pole-zero pairing and ordering of the second order sections. Therfore an optmization procedure to finding a good ordering and pairing is very desirable. Thus, we proposed a sub-optimization procedure for finding "near optimal" solution.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.2
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• pp.151-158
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• 2004

In this paper, an inverse filtering technique for the digital filter model is proposed. This technique enables us to obtain a stable non-causal m inverse filter by transforming (approximating) it to a causal stable inverse system. In practice, a causal FIR approximation to this inverse filter is proposed. It can be shown that the impulse response of the inverse filter for all-pass systems is simply the mirror image of the impulse response for the system. Specially, due to this symmetric property of the impulse response of all-pass systems, the proposed technique is more useful for all-pass systems than other systems. In order to illustrate the proposed inverse filtering technique, four examples are presented. Two of them are for all-pass filters. The other two examples are for IIR and FIR filters. Also, computer simulations demonstrate that the proposed technique works very well.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.8B
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• pp.1092-1099
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• 2001

본 논문에서는 무선 통신 시스템의 중간주파수 처리 단을 디지털로 신호 처리하는 DDC(Digital Down Converter)의 저전력 아키텍처를 제안한다. FIR 필터의 계산량을 줄이기 위해서 개발된 Interpolated FIR 필터가 DDC의 데시메이션 필터로 널리 사용되고 있다. 본 논문은 이와 같은 Interpolated FIR 필터의 개념이 IIR 필터에도 적용될 수 있음을 보이고, 전력 소모와 구현 면적이 기존의 Interpolated FIR 구조보다 더욱 감소된 Interpolated IIR 필터 구조를 제안하였다. CDMA IS-95 DDC 사양의 데시메이션 필터를 FIR 구조, Interpolated FIR 구조, IIR 구조, Interpolated IIR 구조로 구현하여 이 4가지 구조들의 전력소모와 구현 면적을 비교하였으며 제안된 Interpolated IIR 구조가 기존의 Interpolated FIR 구조에 비하여 15.2%의 소모전력 감소와 35.3%의 구현면적의 감소를 달성할 수 있음을 보인다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.788-795
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• 2000

Scattered look-ahead pipelining method can be efficiently used for high sample rate or low-power applications of digital recursive filters. Although the pipelined filters are guaranteed to be stable by this method, these filters suffer from large round off noise when the poles are crowed within some critical regions. To avoid this problem, a low-noise implementation technique was proposed using constrained Remez exchange algorithm. By the constrained filter design approach, the desired filter spectrum is satisfied while some of the pole angles are constrained to avoid pole crowding within critical regions. In the proposed approach, to obtain improved spectrum characteristics or better round off noise properties, the radius of the angle-constrained pole is optimized depending on the direction of the pole movement.

• The KIPS Transactions:PartA
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• v.12A no.1 s.91
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• pp.7-12
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• 2005

Scattered look-ahead(SLA) pipelining method can be efficiently used for high-speed or low-power applications of digital II R filters. Although the pipelined filters are guaranteed to be stable by this method, these filters suffer from large roundoff noise when the poles are crowded within some critical regions. An angle and radius constrained II R fille. design approach using modified Remez exchange algorithm and least squares algorithm is proposed to avoid tight pole-crowding in pipelined filters, resulting in improved frequency responses and reduced coefficient sensitivities. Experimental results demonstrate that our proposed method leads to chip area reduction by $33{\%}$ and low power by $45{\%}$ against the conventional method.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.30-32
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be Improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• Proceedings of the KIEE Conference
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• 2005.05a
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• pp.15-17
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• 2005

Recently genetic algorithm techniques have widely used in adaptive and control schemes for production systems. However, generally it costs a lot of time for learning in the case applied in control system. Furthermore, the physical meaning of genetic algorithm constructed as a result is not obvious. And this method has been used as a learning algorithm to estimate the parameter of a genetic algorithm used for identification of the process dynamics of nonlinear IIR filter and it was shown that this method offered superior capability over the genetic algorithm. A genetic algorithm is used to solve the parameter identification problem for linear and nonlinear digital filters. This paper goal estimate nonlinear IIR filter parameter using the genetic algorithm.

• Journal of the Semiconductor & Display Technology
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• v.20 no.2
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• pp.77-81
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• 2021

In this paper, a low-csst ECG system using a digital filter was implemented. After amplifying the analog ECG signal, it is converted into a digital signal and filtered. The developed ECG module is miniaturized by removing the analog filter element that occupies the existing volume and replacing it with a digital filter using a 3-stage Butterworth filter which is one of IIR filters. It uses a serial monitoring program with C# to check and save the ECG waveform measured on a computer screen. The ECG system using a developed digital filter in this paper uses a low-cost processor instead of an expensive, high-end processor, and its size and price are reduced by converting the analog filter to a digital filter. In addition, since the waveform of the developed ECG system is similar to the actual ECG waveform of MIT-BIU, it is considered that the existing analog filter can be replaced with the developed digital filter.