• Journal of Broadcast Engineering
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• v.13 no.3
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• pp.289-298
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• 2008

We propose an SNR-scalable coding algorithm for three-dimensional mesh sequences based on singular value decomposition (SVD). SVD achieves a coding gain by representing a mesh sequence with a small number of basis vectors and singular values. First, we introduce a bit plane coding scheme and derive a quantitative relationship between each bit plane and the reconstructed image quality. Using the relationship, we develop a rate-distortion (RD) optimized coding algorithm. Moreover, we propose prediction techniques to exploit the spatio-temporal correlations in real mesh sequences. Simulation results demonstrate that the proposed algorithm provides significantly better RD performance than conventional SVD coders.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3166-3168
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• 2000

This paper proposed variable threshold dual rate ADPCM coding method which is modified from the standard ADPCM of ITU G.726 for speech quality improvement. The speech quality of variable threshold dual rate ADPCM is better than single rate ADPCM at noisy environment without increasing the complexity by using ZCR(Zero Crossing Rate). In this case, ZCR is used to divide input signal samples into two categories(noisy & speech). The samples with higher ZCR is categorized as the noisy region and the samples with lower ZCR is categorized as the speech region. Noisy region uses higher threshold value to be compressed by 16Kbps for reduced bit rates and the speech region uses lower threshold value to be compressed by 40Kbps for improved speech quality. Comparing with the conventional ADPCM, which adapts the fixed coding rate. the proposed variable threshold dual rate ADPCM coding method improves noise character without increasing the bit rate. For real time applications, ZCR calculation was considered as a simple method to obtain the background noise information for preprocess of speech analysis such as FFT and the experiment showed that the simple calculation of ZCR can be used without complexity increase. Dual rate ADPCM can decrease the amount of transferred data efficiently without increasing complexity nor reducing speech quality. Therefore result of this paper can be applied for real-time speech application such as the internet phone or VoIP.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.31-35
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• 1996

This paper presents a real time algorithm for monitoring of the arrythmia of ECG signal. A real time monitoring, following by detecting a QRS complex, is the most important. Using 2-dimensional time-delay coordinates which are reconstructed by the phase portrait plotting special trajectory, we detect QRS complexes. In this study, arrythmias are detected by matching the past standard template with tile present pattern when changing abruptly In order to matching with each other, we propose modified chain coding algorithm which applies vetor table consisting of eight orthonormal code(=binary code) to the phase portraits. This algorithm using logical function increases the weight if exceeding to the threshold determinded by correlation value and the distance from a straight line(y=x). Evaluating the performance of the proposed algorithm, we use standard MIT/BIH database. The results are fellowing, 1) Improve the speed of matching template than that of cross-correlation ever has been used. 2) Because the proposed algorithm is robust to varing fiducial point, it is possible to monitor the ECG signal with irregular RR interval. 3) In spite of baseline wandering owing to the low frequency noise, monitoring performance is not reduced.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2003.06a
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• pp.181-184
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• 2003

Recently, the fieldbus becomes an indispensable component for many automated systems. In the fieldbus system, realtime data containing sensor values and control commands has a tendency to rapidly lose its value as time elapses after its creation. In order to deliver these data in time, the fieldbus network should be designed to have short delay compared to the maximum allowable delay. Because the communication delay is affected by performance parameters such as target rotation timer of token passing protocol, it is necessary to select proper parameter settings to satisfy the real-time requirement for communication delay. This paper presents the timer selection method for Profibus token passing networks using genetic algorithm (GA) to meet the delay requirements.

• Journal of Korea Multimedia Society
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• v.18 no.2
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• pp.149-157
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• 2015

The 3D-AVC standard aims at improving coding efficiency by applying new techniques for utilizing intra, inter and view predictions. 3D video scenes are rendered with existing texture video and additional depth map. The depth map comes at the expense of increased computational complexity of the encoding process. For real-time applications, reducing the complexity of 3D-AVC is very important. In this paper, we present a fast intra mode decision algorithm to reduce the complexity burden in the 3D video system. The proposed algorithm uses similarity between texture video and depth map. The best intra prediction mode of the depth map is similar to that of the corresponding texture video. The early decision algorithm can be made on the intra prediction of depth map coding by using the coded intra mode of texture video. Adaptive threshold for early termination is also proposed. Experimental results show that the proposed algorithm saves the encoding time on average 29.7% without any significant loss in terms of the bit rate or PSNR value.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.979-982
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• 2000

In this paper we suggest the adaptive QoS-based VOD system design, which can diagnose and reflect the variable network situation. This system is based on the changeable Forward Error Correction(FEC) coding and re-transmission to enhance system quality by using the measured real-time network value such as loss, delay and jitter.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.35-38
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• 2000

Low bit rate image/video coding is essential for many visual communication applications. When bit rates become low, most compression algorithms yield visually blocking artifacts that highly degrade the perceptual quality, and post-processing approaches provide one attractive solution. In this paper, we provide simple and very efficient blocking artifacts reduction method using constrained random value addition. Our method is simple enough to be used for real-time post-processing of video.

• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.1
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• pp.99-104
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• 2014

Nowadays, various researches fulfill in many communication engineering area for B4G (Beyond Forth Generation). Next LTE-A (Long Term Evolution Advanced), MU-MIMO (Multi-User Multi Input Multi Output) method raises to upgrade throughput performance. However, the method of user selection is not decided because of many types and discussions in MU-MIMO system. Many existing methods are powerful for enhancing performance but have various restrictions in practical implementation. Fairness problem is primary restriction in this area. Existing papers emphasis algorithm to increase sum-rate but we introduce an algorithm about dealing with fairness problem for real commercialization implementation. Therefore, this paper introduces new user selection method in MU-MIMO system. This method overcomes a fairness problem in SUS (Semiorthogonal User Selection) algorithm. We can use the method to get a similar sum-rate with SUS and a high fairness performance. And this paper uses a hybrid method with SC-SUS (Superposition Coding SUS) algorithm and SUS algorithm. We find a threshold value of optimal performance by experimental method. We show this performance by computer simulation with MATLAB and analysis that results. And we compare the results with another paper's that different way to solve fairness problem.

• Journal of Convergence for Information Technology
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• v.10 no.5
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• pp.1-7
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• 2020

This paper proposes a link adaptation method for Underwater Internet of Things (IoT), which reduces power consumption of sensor nodes and improves the throughput of network in underwater IoT network. Adaptive Modulation and Coding (AMC) technique is one of link adaptation methods. AMC uses the strong correlation between Signal Noise Rate (SNR) and Bit Error Rate (BER), but it is difficult to apply in underwater IoT as it is. Therefore, we propose the machine learning based AMC technique for underwater environments. The proposed Modulation Coding and Scheme (MCS) prediction model predicts transmission method to achieve target BER value in underwater channel environment. It is realistically difficult to apply the predicted transmission method in real underwater communication in reality. Thus, this paper uses the high accuracy BER prediction model to measure the performance of MCS prediction model. Consequently, the proposed AMC technique confirmed the applicability of machine learning by increase the probability of communication success.

• Proceedings of the KIEE Conference
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• 2003.07d
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• pp.2480-2482
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• 2003

As make a simulator including user interface functions like start & stop, load parameters, record and save, view 3D display has a real-like length and numerical value of sizes, represent real-shape of inner and outer part of robot, make the possible fast and slow selective observation as a adjust a step, receiving the images through the image device which attached in robot, so make a motion tester simulator of humanoid robot which coded by windows based GUI(Graphic User Interface) program with a MMI(Man Machine Interface) function that user can watch the environment which included robot and use a images. For implement this, we use a design data that converted data which made by use a CAD for Laser RP(Rapid Prototyping) progress into C coding for simulator programming. Using OpenGL, an API of graphic, it has a efficiency and detail of graphic operation. To make and test animation data, it has the option of save and resume in animation.