• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.2
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• pp.88-94
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• 2013

In this paper, an efficient signal interference control technique to improve the convergence speed of LMS algorithm is introduced. The convergence characteristics of the proposed algorithm,whose coefficients are multiply adapted in a symbol time period by recycling the received data,are analyzed to prove theoretically the improvement of convergence speed. According as thestep-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the average squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• International Journal of Internet, Broadcasting and Communication
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• v.8 no.4
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• pp.19-25
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• 2016

There are many free applications that need users to sign up before they can use the applications nowadays. It is difficult to choose a suitable password for your account. If the password is too complicated, then it is hard to remember it. However, it is easy to be intruded by other users if we use a very simple password. Therefore, biometric-based approach is one of the solutions to solve the issue. The biometric-based approach includes keystroke dynamics on keyboard, mice, or mobile devices, gait analysis and many more. The approach can integrate with any appropriate machine learning algorithm to learn a user typing behavior for authentication system. Preprocessing phase is one the important role to increase the performance of the algorithm. In this paper, we have proposed ensemble-by-session (EBS) method which to operate the preprocessing phase before the training phase. EBS distributes the dataset into multiple sub-datasets based on the session. In other words, we split the dataset into session by session instead of assemble them all into one dataset. If a session is considered as one day, then the sub-dataset has all the information on the particular day. Each sub-dataset will have different information for different day. The sub-datasets are then trained by a machine learning algorithm. From the experimental result, we have shown the improvement of the performance for each base algorithm after the preprocessing phase.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.5 s.347
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• pp.121-127
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• 2006

It is well known the fact that turbo codes has better performance as the number of iteration and the interleaver size increases in the AWGN channel environment. However, as the number of iteration and the interleaver size are increased, it is required much delay and computation for iterative decoding. Therefore, it is important to devise an efficient criterion to stop the iteration process and prevent unnecessary computations and decoding delay. In this paper, it proposes the efficient stop criterion algorithm for turbo codes using the maximum sign change of LLR. It is verifying that the proposal variable iterative decoding controller can be reduced the average iterative decoding number compared to conventional schemes with a negligible degradation of the error performance.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.48-55
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• 2000

Recently due to rapid improvement of integrated circuit design environment, size of IC design is to become large to possible design System on Chip(SoC) that one chip with multi function enclosed. Also cause to this rapid change, consumption market is require to spend smallest time for new product development. In this paper to propose a methodology can design a large size IC for save time and applied to design of MPEG Layer 2 decoder to can use audio receiver in digital broadcast system. The digital broadcast audio decoder in this paper is pointed to save hardware size as optimizing algorithm. MPEG Layer 2 decoder algorithm is include MAC to can have an effect on hardware size. So coefficients are using sign digit expression. It is for hardware optimization. If using this method can design MAC without multiplier. The designed audio decoder is using 14,000 gates hardware size and save 22% (4000 gates) hardware usage than using multiplier. Also can design MPEG Layer 2 decoder usable digital broadcast receiver for short time.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.1788-1793
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• 2003

The present of state is situation that is realized by necessity of maintenance of public security about great many information is real condition been increasing continually in knowledge info-age been situating in wide field of national defense, public peace, banking, politics, education etc. Also, loss or forgetfulness, and peculation by ID for individual information and number increase of password in Internet called that is sea of information is resulting various social problem. By alternative about these problem, including Biometrics, several authentication systems through sign(Signature), Smart Card, Watermarking technology are developed. Therefore, This paper shows that extract factor that efficiency can get into peculiar feature in physical features for good fingerprint recognition algorithm implementation with old study finding that take advantage of special quality of these fingerprint.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.389-392
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• 2002

In this paper, we implemented system to detect obstacle in that develop a guidance robot for the visually impaired through sound. We used ultra sonic sensor to detect obstacle. We supposed the algorithm that classifies distance and direction of obstacle using information that produce correct warning negative sign according to direction and distance of obstacle. According to the experiment, a reagent could detect obstacle without sight information.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2E
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• pp.39-44
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• 1996

The performance of two-channel adaptive noise canceller is ofter degraded by the weights perturbation due to the speech signal. In this paper, an adaptive noise canceller employing a speech detector and two adaptation algorithms which are switched according to the speech detector is proposed. When highly correlated speech signal is detected, the tap weights of the adaptive filter are adapted by the sign algorithm. On the other hand, the weights are adapted by the NLMS algorithm when silence is detected or when the characteristics of the noise propagation channel is changed. The employed speech detector utilizes the power ratio of the input and the output of an adaptive linear prediction-error filter. According to the computer simulation, the proposed method yields better performance than conventional ones.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6C
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• pp.571-577
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• 2009

This paper proposes some new improved versions of node-wise VCRBP algorithm for low-density parity-check (LDPC) codes, called forced-convergence node-wise VCRBP algorithm and sign based node-wise VCRBP, both of which significantly reduce the decoding complexity and latency, with only negligible deterioration in error correcting performance.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1219-1222
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• 1998

This paper analyzes the dynamic spars algebraic codebook used to model a residual signal and proposes a new algebraic codebook structure as well as a searching process with improved performance. The proposed algorithm improves the disadvantage of algebraic codebook without increased computation. First, this paper makes it possibel to select various pulse amplitudes differently from the conventional method which looks up the sign bit simply. In addition, two pulses are made to be selected on the same track. For speech quality on the telephone line 5.6kbps speech coder using the proposed algorithm was equivalent to the 6.3kbps MP-MLQ in the viewpoint of subjective speech quality. However, speech degradation was caused a little compared to the MP-MLQ where MNRU 1=15dB.

• Journal of Korea Multimedia Society
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• v.19 no.11
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• pp.1862-1870
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• 2016

A respiration is one of the most useful techniques for vital checking and an abnormal respiration is often the earliest sign of critical illness. Detection of respiration is based on the photo-plethysmography (PPG) with photodiode technique. Because PPG sensor using photodiode can be easily miniaturized, it is suitable for wearable devices. A system to measure respiration rate based on PPG signal is implemented and for a reliable measurement an improved algorithm in accuracy using PPG signal pattern is proposed in this paper. As results regarding to three types of respirations (regular interval, free interval, and weak respiration) the proposed algorithm showed error rate of 0.047, 0.067, and 0.122 respectively.