• Proceedings of the KIEE Conference
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• 1990.07a
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• pp.51-54
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• 1990

A parametrization for a linear system is presented to design a direct model reference adaptive pole placement controler. This parametrized model is one of the structured nonminimal models. The exponentially weighted least-squres algorithm is employed to estimate the control parameters. The direct adaptive controller has the exponential weighting properties by the proposed method of selecting the characteristic polynomials of the sensitivity function filters in connection with the reference models.

• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.331-334
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• 2003

In this paper, using B-splines as universial approximators, we have obtained a plant parametrization which permits the construction of an adaptive observer. The particular property of this parametrization is that the dynamic order of the filters in this design does not depend on the number of parameters in the plant parametrization. This appears to be a beneficial property especially because the number of such parameters tends to be very high for universial approximator based designs.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.413-416
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• 2002

In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.171-174
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• 2000

This paper proposes a pipelined adaptive filter based on affine projection algorithm with order 2. This filter gives a better convergence performance than that of LMS or NLMS pipeline algorithm and has same latency with the pipeline algorithm based on equivalent transformation. Compared to the critical path of the pipeline NLMS implementation, only 2 additions are increased in that of the proposed implementation.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.10
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• pp.80-88
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• 1997

This paper describes a design of a Wavelet Adaptive Filter(WAF) for the removal of the baseline wandering and the minimization of the signal distortion using by wavelet transform and adaptive filter in the ECG signal. WAF consists of two parts. The first part is wavelet transform that decomposes the ECG signal into seven frequency bands using Vaidyanathan and Hoang wavelet. The second part is adaptive filter that uses the signal of seventh low frequency band among the wavelet transformed signals as primary input and a unit impulse sequence as reference input. For the evaluation of the performance of WAF, we used several baseline wandering elimination filters such as commerical standard filter with cutoff frequency of 0.5Hz and general adaptive filter. We made use of MIT/BIH database and real patient data for the evaluation. In conclusion, WAF showed a lower ST segement distortion than standard filter and adaptive filter and has a higher eliminated noise power than standard filter and adaptive filter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.10 no.5
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• pp.258-265
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• 1985

A design method for a single-input single-output discrete time model reference adaptive system is described in this paper. By using the state-variable filters into inputs and outputs in reference model and unknown system, a simple adaptive structure which use all accessible signals can be constructed. Some papers for the adaptive shstem is which thw relative degree of unknown system have one or two have been reported, but the resulting adaptive system are intricate in structures and the design theories for the model reference adaptive system are not generalized. In this paper, for having two or more relative degrees, it has been verified that an adaptive scheme can be obtained by introducing a simple linear filter.

• Proceedings of the KOSOMBE Conference
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• v.1995 no.11
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• pp.101-104
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• 1995

In this paper, we designed a cascade adaptive filter for elimination of the baseline drift and the distortion of the filtered signal. The cascade adaptive filter(CAF) consists of two filters. The first adaptive filter which has the cutoff frequency of 0.3Hz eliminate the noisy signal. The second adaptive filter remove the remnant baseline drift which is not eliminated by the first adaptive filter. Comparing the performance of the CAF with standard filter, recursive notch filter(RNF) and a adaptive impulse correlated filter(AICF), the CAF showed a higher performance in removal of the baseline drift than standard filler, and RNF. Also, considering the distortion of filtered signal, CAF is better than AICF and is comparable to the standard filter.

• International Journal of Control, Automation, and Systems
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• v.1 no.4
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• pp.444-452
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• 2003

An integration system with multi-measurement sets can be realized via combined application of a centralized and federated Kalman filter. It is difficult for the centralized Kalman filter to remove a failed sensor in comparison with the federated Kalman filter. All varieties of Kalman filters monitor innovation sequence (residual) for detection and isolation of a failed sensor. The innovation sequence, which is selected as an indicator of real time estimation error plays an important role in adaptive mechanism design. In this study, the centralized Kalman filter with adaptive measurement fusion is introduced by means of innovation sequence. The objectives of adaptive measurement fusion are automatic isolation and recovery of some sensor failures as well as inherent monitoring capability. The proposed adaptive filter is applied to the GPS/SDINS integration system with an additional sensor. Simulation studies attest that the proposed adaptive scheme is effective for isolation and recovery of immediate sensor failures.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.20 no.5
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• pp.99-106
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• 2006

In this paper, implementation of active mufflers using multiple channel adaptive IIR filter is presented. Usually, recursive LMS(RLMS) algorithms for adaptive IIR filters are highly efficient than filtered-X LMS(FXLMS) algorithms, when the order of both algorithms are the same. However, RLMS algorithms usually diverge before the algorithms arenot yet converged. So, the prefilters are presented to improve the stability by pulling the poles of feedback control transfer function in the beginning of active noise control and returning the original poles after the filters converge. The engine noises of diesel engine automobiles and gasoline engine automobiles are analyzed and the mathematical model of an active muffler is derived. Computer simulations and experiments are performed to show the effectiveness of the proposed systems.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.2
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• pp.211-224
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• 1994

In this paper, a multi-level adaptive intraframe method has been proposed to separate the luminance and chominance components in NTSC composite signal. The control signals are generated by detecting the vertical correlation and transition in the horizontal and diagonal directions. The chrominance component is adaptively processed through vertical and horizontal filters according to the control signals and the luminance component is processed by subtracting the chrominance component from the composite video signal. The several filters have been used at the sampling rate of four times the color subcarrier frequency and computer simulation and SVP(Serial Video Processing) system have been introduced to compare the performance of the conventional methods and that of proposed one.