• The Transactions of the Korean Institute of Electrical Engineers P
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• v.56 no.3
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• pp.136-141
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• 2007

In this paper, a new simple control method for active power filter which can realized the complete compensation of the harmonic currents is proposed. In the proposed scheme, a compensating current reference generator employing resonance model implemented by a DSP(Digital Signal Processor) is introduced. Deadbeat control is employed to control the active power filter. The switching pulse width based SVM(Space Vector Modulation) is adopted so that the current of active power filter is been exactly equal to its reference at the next sampling instant. To compensate the computation delay of digital controller, the prediction of current is achieved by the current observer with deadbeat response.

• Journal of Biomedical Engineering Research
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• v.15 no.3
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• pp.281-288
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• 1994

In this paper, a new decomposition method of the interference EMG signal using MAMDF filtering and digital signal processor. The efficient software and hardware signal processing techniques are employed. The MAMDF filter is employed in order to estimate the presence and likely location of the respective templates which may include in the observed mixture, and high-resolution waveform alignment is employed in order to provide the optimal combination set and time delays of the selected templates. The TMS320C25 digital signal processor chip is employed in order to execute the intensive calculation part of the software. The method is verified through a simulation with real templates which are obtain ed from needle EMG. As a result, the proposed method provides an overall speed improvement of 32-40 times.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.27 no.10
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• pp.909-916
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• 2016

The beacon receiver is an equipment which detects and measures the signal strength of transmitting satellite beacon signal. Beacon signals transmitted by satellites are low power continuous wave(CW) signals without any modulation intended for antenna steering to satellite direction and power control purposes on the earth. The beacon signal detection method using a very narrow band analog filter and RSSI(Received Signal Strength Intensity) has been typically used. However, it requires the implementation to track the frequency at the beacon receiver, thus a beacon frequency variation of the satellite due to temperature changes and long-term operation. Therefore, in this paper, the beacon signal detection receiver is designed by using a very narrow band digital filter array for a faster acquisition and SNR(Signal to Noise Ratio) method detection. For this purpose, by calculating the satellite link budget with the rain attenuation between satellite and ground station, and then extracting the received $C/N_o$ of the beacon signal, this work derives the bandwidth and the array number of the configured digital filter that gives the required C/N.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.45 no.2
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• pp.18-25
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• 2008

The current display systems including TVs are going digital and large-sized, and high visual quality of those systems becomes a very important selling point in the current display system market. Thus, various researches have been carried out for enhancing the visual quality of digital display systems. One of the important digital image(or video) enhancement techniques is sharpness enhancement, and it is generally based on a transient improvement technique that reduces the edge transition time. However, this technique often generates overshoot and undershoot, which cause undesirable pixel-level changes around the transient improved edge. In this paper, we propose a new nonlinear filter-based adaptive shoot elimination method for effectively suppressing the overshoot and undershoot that occur in the transient improvement, so that we can obtain visually sharper and clearer digital images(or videos). The proposed method uses two orthogonal directional min/max nonlinear filters with an adaptive shoot elimination scheme in order to effectively suppress the visually sensitive overshoot and undershoot. Experimental results show that the proposed method suppresses the overshoot and undershoot almost perfectly while maintaining the effect of the transient improvement. The applications of the proposed method include digital TVs, digital monitors, digital cameras/camcoders, portable media players(PMP), etc.

• 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.

• Journal of Digital Convergence
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• v.11 no.7
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• pp.135-140
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• 2013

Noise estimation and detection algorithm to adapt quickly to changing noise environment using the LMS Filter. However, the LMS Filter for noise estimation for a certain period of time and need time to adapt. If the signal changes occur, have the disadvantage of being more adaptive time-consuming. Therefore, noise removal method is proposed to a frame basis AELMS Filter to compensate. In this paper, we split the input signal on a frame basis in noisy environments. Remove the LMS Filter by configuring noise predictions using the mean and variance. Noise, even if the environment changes fast adaptation time to remove the noise. Remove noise and environmental noise and speech input signal is mixed to maintain the unique characteristics of the voice is a way to reduce the damage of voice information. Noise removal method using a frame basis AELMS Filter To evaluate the performance of the noise removal. Experimental results, the attenuation obtained by removing the noise of the changing environment was improved by an average of 6.8dB.

• Journal of Broadcast Engineering
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• v.16 no.1
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• pp.1-13
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• 2011

In this paper, we propose an adaptive in-loop filter to improve the coding efficiency. Recently, there are post-filter hint SEI and block-based adaptive filter control (BAFC) methods based on the Wiener filter which can minimize the mean square error between the input image and the decoded image in video coding standards. However, since the post-filter hint SEI is applied only to the output image, it cannot reduce the prediction errors of the subsequent frames. Because BAFC is also conducted with a deblocking filter, independently, it has a problem of high computational complexity on the encoder and decoder sides. In this paper, we propose the low-complexity adaptive in-loop filter (LCALF) which has lower computational complexity by using H.264/AVC deblocking filter, adaptively, as well as shows better performance than the conventional method. In the experimental results, the computational complexity of the proposed method is reduced about 22% than the conventional method. Furthermore, the coding efficiency of the proposed method is about 1% better than the BAFC.

• Journal of the Korea Institute of Military Science and Technology
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• v.3 no.2
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• pp.121-130
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• 2000

A practical method is proposed to improve hit probability of the digital gun fire control system, when the measured rate of the tracking sensor becomes biased under some operational situation. For ground moving target it is shown that the well-known Kalman filter which uses position measurement only can be optimally used to eliminate the rate bias error. On the other hand, for 3D moving aircraft we present a new algorithm which incorporate FIR-type filter, which uses position and rate measurement at the same time, and the fixed-lag smoother using position measurement only, and show that it has the optimal performance in terms of both estimation accuracy and response time.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.11b
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• pp.1104-1108
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• 2001

It is a difficult and important task to measure the reverberation time of an acoustic room with a short reverberation time. This paper presents a new technique to measure the reverberation time of an acoustic room with low value of BT60. The digital signal processing technique used to do this is the wavelet filter which is very flexible to design the 1/n octave band filter and has no delay problem compared with the conventional IIR digital filter. This method is successfully applied to the measurement of the reverberation time at low frequency band of famous concert halls in Korea.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.6
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• pp.510-520
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• 1988

In this paper, the output error variance due to the differential vcariation of the state-space coefficient [ABCD], which is the coefficient quentization error, is normalized on the variance for cases that infinite wordlength state-space digital filter is realized by the finite one. That is, defining S as the statistical sensitivity and extending controllability gramian, observability gramian, and 2nd order mode analysis method to the state space digital filter, we synthesize the realization structure with the minimum statistical sensitivity and prove the effecency of the minimum statistical sensitivity structure synthesis by the simulation.