• Journal of the Institute of Electronics Engineers of Korea SD
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• v.45 no.2
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• pp.28-33
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• 2008

This paper presents a continuous-time equalizer adopting a clock loss tracking technique for digital display interface. This technique uses bottom hold circuit to detect the incoming clock loss. The generated loss signal is directly fed to equalizer filters, building adaptive feed-forward loops which contribute the stability of the system. The design was done in $0.18{\mu}m$ CMOS technology. Experimental results summarize that eye-width of minimum 0.7UI is achieved until -33dB channel loss at 1.65Gbps. The average power consumption of the equalizer is a maximum 10mW, a very low value in comparison to those of previous researches, and the effective area is $0.127mm^2$.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.280-287
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• 2016

In this paper, we propose a normalization algorithm that can be applied to adaptive filters for multi-channel active noise control. The FxLMS (Filtered-x Least Mean Square) algorithm for the single-channel active noise control can be normalized in the same way as the NLMS (Normalized Least Mean Square) algorithm, whereas in case of the multi-channel active noise control, the single-channel normalization for the FxLMS algorithm cannot be extended to the normalization for the multi-channel FxLMS algorithm straightforwardly. First, we adopt a generalized normalization algorithm for the multi-channel FxLMS algorithm based on the principle of minimal disturbance and then, proposed a normalized algorithm considering only diagonal elements to avoid computation for matrix inversion. We carried out performance comparisons of the proposed algorithm with other algorithms without normalization. It is shown that the proposed algorithm presents better convergence characteristics under non-stationary environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.2C
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• pp.181-192
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• 2003

Blind equalization of communication channel is important because it does not need training signal, nor does it require a priori channel information. So, we can increase the bandwidth efficiency. The linear prediction error method is perhaps the most attractive in practice due to the insensitive to blind channel equalizer length mismatch as well as for its simple adaptive implementation. Unfortunately, the previous one-step prediction error method is known to be limited in arbitrary decision delay. In this paper, we propose method for fractionally spaced blind equalizer with arbitrary decision delay using one-step forward prediction error filter from second-order statistics of the received signals for SIMO channel. Our algorithm utilizes the forward prediction error as training signal and computes the best decision delay from all possible decision delay. Simulation results are presented to demonstrate the performance of our proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.9B
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• pp.908-913
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• 2009

Mobile phones became important household appliance that cannot be without in our daily lives. And the short messaging service (SMS) in these mobile phones is 1.5 to 2 times more than the voice service. However, the spam filtering functions installed in mobile phones take a method to receive specific number patterns or words and recognize spam messages when those numbers or words are present. However, this method cannot properly filters various types of spam messages currently dispatched. This paper proposes a more powerful and more adaptive spam filtering system using SVM and thesaurus. The system went through a process of isolating words from sample data through pro-processing device and integrating meanings of isolated words using a thesaurus. Then it generated characteristics of integrated words through the chi-square statistics and studied the characteristics. The proposed system is realized in a Window environment and the performance is confirmed through experiments.

• The Journal of the Acoustical Society of Korea
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• v.33 no.6
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• pp.375-382
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• 2014

In many practical active noise control applications, the primary noise contains multiple closely-spaced harmonics. A narrowband ANC system consists of adaptive filters excited by a composite reference signal, which is the set or sum of sinusoids. This paper analyzes and shows that the convergence speeds of the direct form, parallel form, and simplified parallel form narrowband ANC systems are affected by the fundamental frequency and frequency separation between two adjacent sinusoids in the reference signal. This paper also proposes the new simplified parallel form narrowband ANC system whose convergence speed is independent on the frequency of the reference signal. Computer simulations are conducted to verify the analysis presented in the paper and to compare the proposed narrowband ANC system with the conventional narrowband ANC system.

• Journal of Biomedical Engineering Research
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• v.20 no.3
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• pp.291-297
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• 1999

Source consislency filtering (SCF) is very effective at removing nOIse when only one or a few leads among multi-lead ECG signals are contaminated. When the noise at one or only a few leads are statistically uncorrelated with signals at other leads, SCF seleclIvely removes the noise with a neglIgIble amount of distortion in the original signal waveform. In order to enhance the understanding of this new method, we describe the lheory and implementational details of SCF in this paper. Numerical implementation and test results of SCF on a multi-lead ECG dalabase show that SCF is a new kind of adaptive filters utilizmg spatial as well as temporal information in multi-c.hannel signals origmatmg from a single source. We also describe the limitations and future improvements in using SCF.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.58-66
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• 2012

Even after establishment of H.264/AVC standard, the video coding experts group (VCEG) of ITU-T has researched on development of promising coding techniques to increase coding efficiency based on the key technology area (KTA) software. Recently, the joint collaboration team video coding (JCT-VC) which was composed of the VCEG and the motion picture experts group (MPEG) of ISO/IEC is developing a next-generation video standard namely HEVC intended to gain twice efficiency than H.264/AVC. An adaptive interpolation technique, one of various next-generation techniques, reported higher coding efficiency. However, it has high computational complexity and does not deal with various error characteristics for videos. In this paper, we investigate characteristics of interpolation filters and propose an effective fixed interpolation filter bank including diverse properties of error. Experimental results is shown that the proposed method achieved bitrate reduction by 0.7% and 1.3% compared to fixed directional interpolation filter (FDIF) of the KTA and the directional interpolation filter (DIF) of the HEVC test model, respectively.

• Journal of Broadcast Engineering
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• v.26 no.2
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• pp.143-154
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• 2021

In this paper, we propose a method of compressing digital hologram standardized data provided by JPEG Pleno. In numerical reconstruction of digital holograms, the addition of random phases for visualization reduces speckle noise due to interference and doubles the compression efficiency of holograms. Holograms are composed of completely complex floating point data, and due to ultra-high resolution and speckle noise, it is essential to develop a compression technology tailored to the characteristics of the hologram. First, frequency characteristics of hologram data are analyzed using various wavelet filters to analyze energy concentration according to filter types. Second, we introduce the subband selection algorithm using energy concentration. Finally, the JPEG2000, SPIHT, H.264 results using the Daubechies 9/7 wavelet filter of JPEG2000 and the proposed method are used to compress and restore, and the efficiency is analyzed through quantitative quality evaluation compared to the compression rate.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.43 no.6
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• pp.507-515
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• 2015

This paper proposes a filtering method for effective state vector estimation of highly maneuvering target. It is needed to hit the point called 'sweet spot' to increase the kill probability in missile interception. In paper, a filtering method estimates the length of a moving target tracked by a frequency modulated continuous wave (FMCW) radar. High resolution range profiles (HRRPs) is generated from the radar echo signal and then it's integrated into proposed filtering method. To simulate the radar measurement which is close to real, the study on the properties of scattering point of the missile-like target has been conducted with ISAR image for different angle. Also, it is hard to track the target efficiently with existing Kalman filters which has fixed measurement noise covariance matrix R. Therefore the proposed method continuously updates the covariance matrix R with sensor measurements and tracks the target. Numerical simulations on the proposed method shows reliable results under reasonable assumptions on the missile interception scenario.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.41 no.1
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• pp.9-18
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• 2004

This paper presents an efficient image compression method for memory reduction in multimedia processor which can be simply implemented in hardware and provides high performance. The multimedia processor, which includes processing of high-resolution images and videos, requires large memories: they are external frame memories to store frames and internal line memories for implementing some linear filters. If we can reduce those memories by adopting a simple compression method in multimedia processor, it will strengthen its cost competitiveness. There exist many standards for efficiently compressing images and videos. However, those standards are too complex for our purpose and most of them are 2-D block-based methods, which do not support raster scanned input and output. In this paper, we propose a low-complexity compression method which has good performance, can be implemented with simple hardware logic, and supports raster scan. We have adopted 1${\times}$8 Hadamard transform for simple implementation in hardware and compression efficiency. After analyzing the coefficients, we applied an adaptive thresholding and quantization. We provide some simulation results to analyze its performance and compare with the existing methods. We also provide its hardware implementation results and discuss about cost reduction effects when applied in implementing a multimedia processor.