• Journal of the Korean Institute of Telematics and Electronics
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• v.24 no.2
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• pp.196-200
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• 1987

The following methods are proposed for implementing digital dual tone multi-frequency (DTMF) receiver: using infinite impulse response(IIR) digital filters, period-counting algorithm, discrete Fourier transform(DFT), and fast Fourier transform(FFT)[2]. The PARCOR(Partical Correlation) analysis method which has been widly used in the speech signal processing area is applied to the dual tone multi-frequency(DTMF) signal detection. This method is easy to implement digitally and stronger to digit simulation of speech than any other methods proposed up to date. Since sampling rate of 4KHz is used in the DTMF receiver for the detection of input DTMF signal originally sampled at 8KHz, it effects two times higher multiplexing efficiency.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.10a
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• pp.231-239
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• 2009

The filtered-x LMS (FX-LMS) algorithm has been applied to the active noise control (ANC) system in an acoustic duct. This algorithm is designed based on the FIR (finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filteredu LMS algorithm (FU-LMS) based on infinite impulse response (IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• Journal of Institute of Control, Robotics and Systems
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• v.19 no.5
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• pp.481-487
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• 2013

This paper presents an FIR (Finite Impulse Response) fixed-interval smoothing filter for fast and exact estimating state variables of a discrete nonlinear system with modeling uncertainty. Conventional IIR (Infinite Impulse Response) filter and smoothing filter can estimate state variables of a system with an exact model when the system is observable. When there is an uncertainty in the system model, however, conventional IIR filter and smoothing filter may cause large errors because the filters cannot estimate the state variables corresponding to the uncertain model exactly. To solve this problem, FIR filters that have fast estimation properties and have robustness to the modeling uncertainty have been developed. However, there is time-delay estimation phenomenon in the FIR filter. The FIR smoothing filter proposed in this paper makes up for the drawbacks of the IIR filter, IIR smoothing filter, and FIR filter. Therefore, the FIR smoothing filter has good estimation performance irrespective of modeling uncertainty. The proposed FIR smoothing filter is applied to the integrated navigation system composed of a magnetic compass based DR (Dead Reckoning) and a GPS (Global Positioning System) receiver. Even when the magnetic compass error that changes largely as the surrounding magnetic field is modeled as a random constant, it is shown that the FIR smoothing filter can estimate the varying magnetic compass error fast and exactly with simulation results.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.22 no.2
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• pp.146-155
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• 2012

The filtered-x LMS(FX-LMS) algorithm has been applied to the active noise control(ANC) system in an acoustic duct. This algorithm is designed based on the FIR(finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filtered-u LMS algorithm(FU-LMS) based on infinite impulse response(IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.5
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• pp.556-563
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• 2002

In this paper, a novel method for designing an M-channel, causal, stable IIR cosine-modulated filter bank (CMFB) with near PR property is proposed. The IIR prototype filter is designed with a simple constraint using lattice stucture with 1st order allpass filter components. The IIR prototype filter which is designed by the proposed method has higher stopband attenuation and sharper roll-off characteristic than the one which is designed by the previously proposed method with similar complexity. The proposed M-channel IIR CMFB which is designed from this IIR prototype filter is applied to subband acoustic echo canceller (AEC). We obtained about 15dB higher ERLE using this subband AEC than when M-channel FIR subband AEC with similar complexity.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.58-65
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• 2001

Recently, a 5-degrees-of-freedom (DOF) reverberation model was proposed as a method of representing subjective perception of reverberation as objective measures[1]. This model approximates sound energy decay curve by five objective measures, widely used in which have been concert hall acoustics. However, it is note worthy that there can be infinite number of impulse responses which correspond to a selected 5-DOF reverberation model. There may exist some filters making very unnatural and unrealistic sound. In this paper, the limitation of the 5-DOF reverberation model when it is used as a filter design criteria is investigated. When a 5-DOF reverberation model is given, additional constraints to get natural reverberation are suggested. This is based on the listening tests for several quite different source sounds.

• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.1794-1795
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• 2011

본 연구에서는 가속도 센서의 값을 디지털 신호 처리 과정을 통하여 저역통과 필터(low pass filter), 벡터의 크기(vector magnitude), 롤(roll) 그리고 피치(pitch)를 계산하는 알고리즘을 적용하였다. 필터의 경우 IIR(Infinite Impulse Response)을 이용하였으며 차수는 9차로 하였다. 피험자의 연령은 $25{\pm}5$세의 10명을 기준으로 실험하였으며 앞, 뒤, 좌, 우 방향으로 직각 낙하하도록 하였고 센서 모듈은 오른쪽 허리의 정중앙에 착용하도록 하여 피험자간의 오차가 발생하지 않도록 하였다. 환자의 낙상을 검출하기 위해서 벡터의 크기를 사용하였고 롤과 피치를 이용하여 환자의 낙상 방향을 검출하였다. 결과적으로 피험자 10명의 경우 낙상의 검출률은 100% 였으며 낙상 방향에 따른 앞, 뒤, 좌, 우 판별 정확도는 95% 정도이다. 낙상 방향의 판별은 사고 후 환자를 다룰 때의 주의할 신체부위를 참고하며 재활 운동 시 하체의 어느 쪽이 낙상의 주요인인지 분석하는 보조 자료가 될 수 있다.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1132-1136
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• 2005

In this paper, adaptive equalizer receiver for a turbo code direct sequence code division multiple access (DSCDMA) by using least mean square (LMS) adaptive algorithm is presented. The proposed adaptive equalizer is using soft output of decision feedback adaptive equalizer (DFE) to examines the output of the equalizer and the Log- maximum a posteriori (Log-MAP) algorithm for the turbo decoding process of the system. The objective of the proposed equalizer is to minimize the bit error rate (BER) of the data due to the disturbances of noise and intersymbol interference (ISI)phenomenon on the channel of the DS-CDMA digital communication system. The computer program simulation results shown that the proposed soft output decision feedback adaptive equalizer provides a good BER than the others one such as conventional adaptive equalizer, infinite impulse response adaptive equalizer.

• International Journal of Aeronautical and Space Sciences
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• v.6 no.2
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• pp.1-16
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• 2005

Adaptive control methods are studied for the Satellite to isolate vibration in spite of the nonlinear system dynamics and parameter uncertainties of disturbance. First, a centralized control scheme is developed based on the particle swarm optimization(PSO) algorithm and feedback theory to automatically tune controller gains. A simulation study of a 3 degree-of-freedom device was conducted to evaluate the performance of the proposed control scheme. Next, since a centralized control scheme is hard to construct model dynamics and not goad at performance when controller and systems environment are easily changed, a decentralized control scheme is presented to avoid these defects of the centralized control scheme from the point of view of production and maintenance. It is based on the adaptive control methodologies to find PID controller parameters. Experiment studies were conducted to apply the adaptive control scheme and evaluate the performance of the proposed control scheme with those of the conventional control schemes.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.115-117
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• 2003

ea­RED(Efficient Adaptive RED)[1][2] 라우터 버퍼 관리 알고리즘 성능 향상을 위해서 ea­RED 라우터 버퍼 사이즈 변화를 예측할 수 있는 예측 알고리즘 모듈의 추가 필요성을 느낀다. 그래서 본 논문에서는 ea­RED 라우터 버퍼 관리 알고리즘의 원형인 RED 라우터 버퍼 관리 알고리즘에 AR(AutoRegression Analysis), IIR(Infinite Impulse Response) MACD(Moving Average Convergence & Divergence), LR_Lines(Linear Regression Lines)등의 예측 알고리즘 모듈을 적용하여 변화를 살펴보고. 결과를 비교. 분석하여 ea­RED 라우터 버퍼 관리 알고리즘 성능 향상에 가장 적합한 예측 알고리즘으로 LR_Lines를 선정했다. ea­RED 라우터 버퍼 관리 알고리즘에 적합한 예측 알고리즘 선정을 위해서 RED 라우터 버퍼 관리 알고리즘을 대신 이용한 이유는 ea­RED 라우터 버퍼 관리 알고리즘의 경우 네트워크 상황에 따라, 버퍼 관련 파라미터 값을 수시로 바꾸기 때문에 예측 알고리즘의 정확성을 판단하는데 어려움이 있지만, RED 라우터의 경우는 버퍼 관련 파라미터 값을 변화시키지 않기 때문에, 좀 더 일관성 있고 정확한 분석을 수행할 수 있기 때문이다.