• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.310-312
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• 2011

In complex noise environment, digital filter is being used to obtain, transport and storage original voice or image signal. Digital filter can be largely separated FIR(Finite duration impulse response) filter and IIR(Infinite duration impulse response) filter. Among FIR filter, window function has characteristic of linear phase and as can be easily set pass-band frequency, cutoff frequency and so on. In this paper, We compared with established method using transient characteristic and peak side-lobe in order to check filter characteristics after we designed the existing variants of the window function.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1998.11a
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• pp.424-427
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• 1998

입체음향 시스템의 방향감 제어에서 필수적인 머리전달함수(Head-Related Transfer Function)는 일반적으로 FIR 또는 IIR 필터로 구현되며, IIR 필터의 경우 FIR 보다 비교적 저차 모델링이 가능한 장점을 갖는다. 본 논문에서는 Balanced Model Truncation(BMT)를 이용하여 비교적 높은 차수를 가지는 FIR 필터를 IIR 필터로 설계하여 입체음향 시스템의 실시간 구현시 필수적인 계산의 효율을 높이는 방법을 제시하고, 기존의 방법과의 성능의 비교 평가를 하고자 한다.

• 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.

• 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 Korean Institute of Communication Sciences Conference
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• 1991.10a
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• pp.96-99
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• 1991

FIR형은 하드웨어 규모가 커져서 구성이 어렵다는 단점을 가지고 있으나, IIR형은 적은 차수로써 큰 차수의 FIR필터를 대신할 수 있다. IIR형은 적은 계산량과 간단한 하드웨어 구성이라는 장점을 갖고 있으나 수렴성에 있어서는 어려움이 있다. 본 연구에서는 적은 차수로 양호한 수렴특성을 갖는 IIR 필터를 구성하기 위하여 연속시간영역의 직교필터를 이산시간영역으로 변환하고 직교성을 유지하도록 정규화한 정규직교함수계에 의해서 설정된 전달함수를 적응 디지틀필터에 적용하였다. 이 방법을 확인하기 위하여 FIR형에 비교해서 적은 차수로 컴퓨터 시뮬레이션을 수행한 결과 양호한 수렴특성을 확인하였다.

• Journal of Power Electronics
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• v.9 no.3
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• pp.430-437
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• 2009

This paper presents the dynamic characteristics of buck and buck-boost dc-dc converters with digital filters. At first, the PID, the minimum phase FIR filter and the IIR filter controls are discussed in the buck dc-dc converter. Comparisons of the dynamic characteristics between the buck and buck-boost converters are then discussed. As a result, it is clarified that the superior dynamic characteristics are realized in the IIR filter method. In the buck converter, the undershoot is less than 2% and the transient time is less than 0.4ms. On the other hand, in the buck-boost converter, the undershoot is about 3%. However, the transient time is approximately over 4ms because the output capacitance is too large to suppress the output voltage ripple in this type of converter.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.34 no.10
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• pp.48-55
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• 2006

A multi-filter fusion technique is proposed and this technique is applied to the INS/GPS integrated system. IIR-type EKF and FIR-type RHKF filter are fused to provide the advantages of these filters based on the adaptive mixing probability calculated by the residuals and the residual covariance matrices of the filters. In the INS/GPS, this fusion filter can provide more robust navigation information than the conventional stand-alone filter.

• Proceedings of the Korea Contents Association Conference
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• 2005.05a
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• pp.341-345
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• 2005

In this paper, we propose an algorithm for the approximation of FIR filters by IIR filters. The algorithm is based on a concept of the balanced model reduction. Head-related transfer functions(HRTFs) of dummy-head are approximated by 32-order IIR filters. The binaural sounds using the approximated HRTFs are reproduced by headphone, and serves as a cue of sound image localization. Experiment of sound image are carried out for 10 participants with computer simulation and DSP board respectively. The results of the experiments show that the localization using the approximated HRTFs by IIR filters is the same accuracy as the case of FIR filters that simulate the HRTFs.

• 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.