• Proceedings of the KIEE Conference
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• 1990.07a
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• pp.106-112
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• 1990

This paper presents the FIR filtering scheme for digital distance protection. In most faults, there exists severe oscillation of apparent impedance according to distortion in transient waveforms. In these cases, this scheme provides more accurate fault detection and fault location, as compared with conventional schemes for digital distance protection or Kalman filtering schemes. The test of these facts were performed on EMTP result data. For the real time implementation of the proposed schemes, the motorola DSP 56001 was used to the main calculation board of the digital protective relaying system.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.25-28
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• 1999

본 연구에서는 finite impulse response (FIR) 필터에 의해 인식기의 입력 특징벡터가 필터링되는 경우에 hidden Markov model (HMM) 파라미터를 적응시키는 새로운 기법을 제안한다. 제안한 적응 기법은 필터링에 의해 변환된 특징벡터에 대해 HMM 파라미터를 다시 학습시킬 필요가 없으며 주어진 FIR필터 계수만을 사용하여 HMM 파라미터를 적응시킬 수 있다. 개발된 FIR필터링에 대한 HMM 파라미터 적응 기법은 연속 숫자음 인식 실험에서 재학습 방법과 비교 실험한 결과 low-pass 필터의 경우에 재학습 방법과 비슷한 인식 성능을 나타내었다.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1E
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• pp.15-22
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• 1992

잡음이 있는 환경하에서는 음성 인식의 성능이 현저하게 떨어지게 된다. 본 논문에서는 이렇나 잡음의 영향에 강한 거리척도를 제안하고자 한다. 우리는 잡음이 더해진 음성신호의 특징벡터를 깨끗한 음성신호의 특징벡터가 FIR 시스템을 거쳐 변형된 것이라고 가정한다. 여기서 FIR 시스템은 잡음의 영 향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템 계수잡음의 영향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템계수들은 RLS 적응 알고리즘을 이용하여 구한다. 제안된 거리척도는 적응 여파 기의 예측 오차에 관한 식으로 표시되어진다. 여러 가지 적응 여파기의 구조중 단일 채널 일차 FIR 구 조가 가장 좋은 음성 인식 성능을 보이며, 이 경우 효과적인 거리척도 알고리즘을 구할 수 있다. 여러 가지 신호대 잡음비에 관하여 화자독립 격리단어 인식 실험을 DTW 알고리즘을 이용하여 수행하여 본 결과 제안된 거리척도가 거의 모든 신호대 잡음비에 대하여 우수한 성능을 보였다.

• Journal of Institute of Control, Robotics and Systems
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• v.21 no.11
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• pp.1057-1063
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• 2015

This paper proposes a new nonlinear filtering algorithm that combines the unscented Kalman filter (UKF) and the finite impulse response (FIR) filter. The proposed filter is called the unscented Kalman/FIR hybrid filter (UKFHF). In the UKFHF algorithm, the UKF is used as the main filter, which produces state estimates under ideal conditions. When failures of the UKF are detected, the FIR filter is operated. Using the output of the FIR filter, the UKF is reset and rebooted. In this way, the UKFHF recovers from failures. The proposed UKFHF is applied to indoor human localization using wireless sensor networks. Through simulations, the performance of the UKFHF is demonstrated in comparison with that of the UKF.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.10
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• pp.2375-2381
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• 2014

In case the coefficient set of an FIR filter is represented in the canonic signed digit (CSD) format with a few nonzero digits, it is possible to implement high data rate digital filters with low hardware cost. Designing an FIR filter with CSD format coefficients, whose number of nonzero signed digits is minimal, is equivalent to finding sparse nonzero signed digits in the coefficient set of the filter which satisfies the target frequency response with minimal maximum error. In this paper, a compressive sensing based CSD coefficient FIR filter design algorithm is proposed for multiplierless and high speed implementation. Design examples show that multiplierless FIR filters can be designed using less than two additions per tap on average with approximate frequency response to the target, which are suitable for high speed filtering applications.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.1
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• pp.150-158
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• 2010

Filtering is necessary for the SRC(sample rate converter), that is used to change the sampling rate of a digital signal. The larger the conversion ratio of the sampling rate becomes, the more signal processing is needed for the filter, which means more complexity on realization. Thus it is important to reduce the amount of signal processing for the case of substantial conversion ratios. In this paper it is presented an efficient design method of a multistage FIR(finite impulse response) filter, with which the rate conversion occurs in stages rather than in one step. In this method, filter searching is performed exhaustively over all possible factorization of the conversion ratio, and also the filter complexity is measured based on direct realization rather than on estimation. It has been shown a designed multistage filter to have a less number of multiplications for filtering operation in comparison with a conventionally designed one. It has also been found that by allowing some variations of the filter architecture such as a halfband filter or a filter with multiple transition bands, the number of multiplications can be reduced further.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.2
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• pp.208-213
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• 2010

Generally, the mal factor for speech recognition is the background noise in speech recognition. The noise is the reason to reduce the speech recognition performance. Owing to the fact, the place to recognize is very important. To improve the recognition performance from the sound having noise, we implemented the noise filtered Wiener filter at the signal process step which adopted the FIR filter. In FIR filter, it deal with the filtered speech signal which is appropriate frequency range of human speech frequency range. Therefore, we make the recognition system distinguish between noise and speech sound from the incoming speech signal.

• Transactions on Control, Automation and Systems Engineering
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• v.2 no.1
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• pp.7-12
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• 2000

A new speech enhancement algorithm for speech corrupted by slowly varying additive colored noise is suggested based on a state-space signal model. Due to the FIR structure and the unimportance of long-term past information, the receding horizon (RH) FIR filter known to be a best linear unbiased estimation (BLUE) filter is utilized in order to obtain noise-suppressed speech signal. As a special case of the colored noise problem, the suggested approach is generalized to perform the single blind signal separation of two speech signals. It is shown that the exact speech signal is obtained when an incoming speech signal is noise-free.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.33.3-33
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• 2001

This paper considers a fault detection and diagnosis using estimation method in uncertain systems. In the state estimation method, we use the robust H$\infty$ FIR filtering algorithm. A novel aspect of the fault detection technique described here is that it explicitly accounts for the effects of simplified models and errors due to the linearization of nonlinear systems at an operating point.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.3
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• pp.2-17
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• 1999

Interpolated second order polynomials(ISOP's) are proposed to design efficient cascaded integrator-comb(CIC)-based decimation filters for a programmable downconverter. It is shown that some simple ISOP's can effectively reduce the passband droop caused by CIC filtering with little degradation in aliasing attenuation. In addition, ISOP's are shown to be useful for simplifying halfband filters that usually follow CIC filtering. As a result, a modified half band filter(MHBF) is introduced which is simpler than conventional halfband filters. The proposed decimation filter for a programmable downconverter is a cascade of a CIC filter, an ISOP, MHBF's and a programmable finite impulse response(FIR) filter. A procedure for designing the decimation filter is developed. In particular, an optimization technique that simultaneously designs the decimation filter is developed. In particular, an optimization technique that simultaneously designs the ISOP and programmable FIR filters is presented. Design examples demonstrate that the proposed method leads to more efficient programmable downconverters than existing ones.