• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.521-521
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• 2000

This paper investigates the problem of robust H$_{\infty}$ filter with FIR(Finite Impulse Response) structure for linear continuous time-varying systems with sampled-data measurements. It is assumed that the system is subject to real time-varying uncertainty which is represented by the state-space model having parameter uncertainty. The robust H$_{\infty}$ FIR filter is proposed for the continuous-time linear parameter uncertain systems. It is also derived from the equivalence relationship between the robust linear H$_{\infty}$ FIR filter and the robust linear H$_{\infty}$ filter with sampled-data measurements.

• Proceedings of the IEEK Conference
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• 2003.07b
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• pp.827-830
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• 2003

This paper proposes an area-efficient FIR filter architecture for sampling rate conversion of hi-fi audio data. Sampling rate conversion(SRC) block converts audio data sampled at 96KHz down to 48KHz sampled data and vice versa. 63-tap FIR filter coefficients have been synthesized that gives 100dB stop band attenuation and 5.2KHz transition bandwidth. Time-shared filter architecture requires only one multiplier and accumulator for 63-tap filter operation. This results in huge hardware saving of up to 10~19 times smaller compared with traditional FIR structure.

• Journal of KIEE
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• v.11 no.1
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• pp.21-26
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• 2001

This paper considers the problem of robust H$\infty$ filter is derived by using the equivalence relationship between the FIR filter and the recursive filter, that would be guarantee a prescribed H$\infty$ performance in the continuous-time context, irrespective of the parameter uncertainty and unknown initial states.

• Journal of the Korean Institute of Intelligent Systems
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• v.25 no.4
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• pp.349-354
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• 2015

This paper presents the stability conditions of the Takagi-Sugeno (T-S) fuzzy sampled-data filter. The error system between the T-S fuzzy system and fuzzy filter is presented. In the sense of the Lyapunov stability analysis, the stability conditions are given, which can be represented in terms of linear matrix inequalities (LMIs). The proposed stability conditions utilize the different approach from the conventional methods, and have better performance than that of the conventional ones. The simulation example is given to show the effectiveness of the proposed method.

• Transactions on Control, Automation and Systems Engineering
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• v.3 no.2
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• pp.83-88
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• 2001

The robust linear H(sub)$\infty$ FIR filter, which guarantees a prescribed H(sub)$\infty$ performance, is designed for continuous time-varying systems with unknown cone-bounded nonlinearity. The infinite horizon filtering for time-varying systems is systems is investigated in therms of two Riccati equations by the finite moving horizon.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2427-2430
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• 2003

A uniformly sampled digital pulse-width modulation adopting a pre-compensation filter scheme for applications in high-resolution digital-to-analog data conversion is described. It is shown that linearization of the intrinsic distortion resulting in uniformly sampled pulse-width modulation can be achieved by using a non-integer delay digital filter embedded within a noise shaping re-quantizer.

• Transactions on Control, Automation and Systems Engineering
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• v.2 no.4
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• pp.255-261
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• 2000

This paper presents the results of the robust H(sub)$\infty$ FIR filtering for a class of nonlinear continuous time-varying systems subject to real norm-bounded parameter uncertainty and know Lipschitz nonlinearity under sampled measurements. We address the problem of designing filters, using sampled measurements, which guarantee a prescribed H(sub)$\infty$ performance in continuous time-varying context, irrespective of the parameter uncertainty and unknown initial states. The infinite horizon causal H(sub)$\infty$FIR filter are investigated using the finite moving horizon in terms of two Riccati equations with finite discrete jumps.

• Smart Structures and Systems
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• v.21 no.2
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• pp.187-194
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• 2018

This paper focuses on the $H_{\infty}$ filter design problem for offshore steel jacket platforms. Its objective is to design a full-order state observer for offshore platforms in presence of unknown disturbances. To make the method more practical, it is assumed that the measured variables are available at discrete-time instants with time-varying sampling time intervals. By modelling the sampling intervals as a bounded time-varying delay, the estimation error system is expressed as a time-delay system. As a result, the addressed problem can be transformed to the problem of stability of dynamic error between the system and the state estimator. Then, based on the Lyapunov-Krasovskii Functional (LKF), a stability criterion is obtained in the form of Linear Matrix Inequalities (LMIs). According to the stability criterion, a sufficient condition on designing the state estimator gain is obtained. In the end, the proposed method is applied to an offshore platform to show its effectiveness.

• Journal of KSNVE
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• v.2 no.2
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• pp.117-123
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• 1992

In the study, the design method of reconstruction filter is studied for synchronized sampling which is necessary for order tracking in rotating machinery. The original data sampled at constant intervals, using fixed anti- aliasing filters, is reconstructed by digital reconstruction filter and is resampled at new sampling times calculated by a suitable shaft angle encoder pulse arrival times in order to synchronize with shaft velocity. In addition to eliminating the tracking synthesizer and filters, this new method has no phase noise due to phase-locked loops.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.16 no.4
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• pp.276-280
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• 2016

A fixed rate speech coder based on the filter bank and the non-uniform sampling technique is proposed. The non-uniform sampling is achieved by the detection of inflection points (IPs). A speech block is band passed by the filter bank, and the subband signals are processed by the IP detector, and the detected IP patterns are compared with entries of the IP database. For each subband signal, the address of the closest member of the database and the energy of the IP pattern are transmitted through channel. In the receiver, the decoder recovers the subband signals using the received addresses and the energy information, and reconstructs the speech via the filter bank summation. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is confirmed. The signal-to-noise ratio (SNR) performance of the proposed method is comparable to that of the uniform sampled pulse code modulation (PCM) below 20 kbps data rate.