• KSII Transactions on Internet and Information Systems (TIIS)
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• v.16 no.4
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• pp.1249-1265
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• 2022

Video streaming has become one of the most popular applications for mobile devices. The network bandwidth required for video streaming continues to exponentially increase as video quality increases and the user base grows. Multi-Path TCP (MPTCP), which allows devices to communicate simultaneously through multiple network interfaces, is one of the solutions for providing robust and reliable streaming of such high-definition video. However, mobile video streaming over MPTCP raises new concerns, e.g., power consumption and cellular data usage, since mobile device resources are constrained, and users prefer to minimize such costs. In this work, we propose a mobile video streaming framework over MPTCP (mDASH) to reduce the costs of energy and cellular data usage while preserving feasible streaming quality. Our evaluation results show that by utilizing knowledge about video behavior, mDASH can reduce energy consumption by up to around 20%, and cellular usage by 15% points, with minimal quality degradation.

• Journal of Advanced Information Technology and Convergence
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• v.10 no.2
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• pp.33-48
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• 2020

By Digital Transformation, new technologies like ML (Machine Learning), Big Data, Cloud, VR/AR are being used to video streaming technology. We choose ML to provide optimal QoE (Quality of Experience) in various network conditions. In other words, ML helps DASH in providing non-stopping video streaming. In DASH, the source video is segmented into short duration chunks of 2-10 seconds, each of which is encoded at several different bitrate levels and resolutions. We built and compared the performances of five prototypes after applying five different machine learning algorithms to DASH. The prototype consists of a dash.js, a video processing server, web servers, data sets, and five machine learning models.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 1994.10a
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• pp.348-353
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• 1994

Magnetic Servo Levitation (MSL) has been proposed as a method to drive a fast-tool servo system. This paper discusses some fundamental control and modeling issues in the development of a long-range high-bandwidth fast-tool servo based on MSL. A resursive linear model is developed to describe the system's dynamics linear model is developed to describe the system's dynamics, and further used to discuss controller design. For a given controller architecture, the performance of two controllers is then compared, one based on an approximation to the inverse plant dynamics, the second based on a adaptive neural network.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.891-898
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• 2003

This paper analyzes and proposes a scheme that improves the performance of the RTP that is developed to support the end-to-end transmission function and QoS monitor function for real-time multimedia data transmission. Although the existing RTP module supports real-time transmission, it has some problems in guaranteeing QoS parameters. To solve this problem, we propose a new Selective Repeat Adaptive Rate Control (SRARC). The SRARC can support QoS by referring to the data transmission status from the client and then classifying the network status into three levels. It selectively transmits multimedia data and dynamically controls transmission rates based on such information as bandwidth, packet loss rate, and latency that can be calculated in data transfer phase. To verify the SRARC, we implement it in real local area networks and compare the QoS parameters of the SRARC with those of the SR and RTP By the experimental results, the SRARC shows better performance in the aspects of bandwidth usage rate, packet loss rates, and transmission delays than the existing RTP schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11A
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• pp.1827-1840
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• 2001

In digital data transmission/storage systems, the compensation for channel distortion is conducted normally using a training sequence that is known a priori to both the sender and receiver. The use of the training sequences results in inefficient utilization of channel bandwidth. Sometimes, it is also impossible to send training sequences such as in the burst-mode communication. As such, a great deal of attention has been given to the approach requiring no training sequences, which has been called the blind equalization technique. On the other hand, to utilize the limited bandwidth effectively, the concept of partial response (PR) signaling has widely been adopted in both the high-speed transmission and high-density recording/playback systems such as digital microwave, digital subscriber loops, hard disk drives, digital VCRs and digital versatile recordable disks and so on. This paper is concerned with blind adaptive equalization of partial response channels whose transfer function zeros are located on the unit circle, thereby causing some problems in performance. Specifically we study how the problems of blind channel equalization associated with the PR channels can be improved. In doing so, we first discuss the existing methods and then propose new structures for blind PR channel equalization. Our structures have been extensively tested by computer simulation and found out to be encouraging in performance. The results seem very promising as well in terms of the implementation complexity compared to the previous approach reported in literature.

• Journal of KIISE:Information Networking
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• v.30 no.4
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• pp.461-473
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• 2003

Multimedia applications are much more sensitive to QoS(Quality of Service) than text based ones due to their data continuity. In order to provide a fast moving MH(Mobil Host) using multimedia application with a consistent QoS,an efficient call admission mechanism is in need. This paper proposes the 2SCA(2-Step Call Admission) scheme based on cal admission scheme using pripority to guarantee the consistent QoS for mobile multimedia applications. A calls of MH are classified new calls, hand-off calls, and QoS upgrading calls. The 2SCA is composed of the basic call admission and advanced call admission; the former determines the call admission based on bandwidth available in each cell and the latter determines the call admission by applying DTT(Delay Tolerance Time), PQeueu(Priority Queue), and UpQueue(Upgrade Queue) algorithm according to the type of each call blocked at the basic call admission stage. In order to evaluate the performance of our mechanism, we measure the metrics such as the dropping probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The result shows that the performance of our mechanism is superior to that of existing mechanisms such as CSP(Complete Sharing Policy), GCP(Guard Channel Policy) and AGCP(Adaptive Guard Channel Policy).

• Journal of the Korea Academia-Industrial cooperation Society
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• v.22 no.1
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• pp.61-73
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• 2021

Metaheuristic search methods have been developed to solve problems with a range of purpose functions in situations lacking information and time constraints. In this study, the Hybrid Vision Correction Algorithm (HVCA), which enhances the performance of the Vision Correction Algorithm (VCA), was developed. The HVCA has applied two methods to improve the performance of VCA. The first method changes the parameters required by the user for self-adaptive parameters. The second method, the CGS structure of the Exponential Bandwidth Harmony Search With a Centralized Global Search (EBHS-CGS), was added to the HVCA. The HVCA consists of two structures: CGS and VCA. To use the two structures, a method was applied to increase the probability of selecting the structure with the optimal value as it was performed. The optimization problem was applied to determine the performance of the HVCA, and the results were compared with Harmony Search (HS), Improved Harmony Search (IHS), and VCA. The HVCA improved the number of times to find the optimal value during 100 repetitions compared to HS, IHS, and VCA. Moreover, the HVCA reduced the Number of Function Evaluations (NFEs). Therefore, the performance of the HVCA has been improved.

• Journal of IKEEE
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• v.13 no.4
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• pp.29-36
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• 2009

The packet loss and the disconnection during handoff are the most critical problems which degrade the video quality in wireless video streaming. To cope with these problems, we propose an efficient video streaming method in this paper, which does not only dynamically adjust the video transmission rate based on the raptor forward error correction (FEC) level, but also minimize the error propagation during handoff. Firstly, the channel bandwidth of the wireless broadband internet, called WiBro, is estimated by analyzing channel parameters including the carrier to interference and noise ratio (CINR) and the handoff. Secondly, the streaming server adjusts the next transmission rate according to the estimated channel bandwidth and the raptor FEC level to avoid packet error. Also, the encoder performs the intra refresh method that inserts an intra frame (I-frame) right after handoff to reduce the error propagation effectively. Experimental results indicate that the proposed method can improve the performance of the video streaming over WiBro network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.8
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• pp.890-895
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• 2016

As the wireless communication technique is developing rapidly, the use of smart devices is increasing. Due to gradually increasing data traffic, a new area, more than 6GHz of bandwidth to increase capacity of the network, has been studied. Millimeter Wave(MmWave) communications utilizes the bandwidth above 6GHz, which makes it possible to achieve one gigabit per second data rate. To overcome the path loss due to the smaller wavelength, the mass of the antenna arrangement is used. This paper presents an algorithm that maximizes the spectral efficiency of the system in the pre-coding process using a hybrid beamforming. Also it is suggested with the optimization of the number of beams that maximizes the spectral efficiency was maximized by the propose method.

• The Journal of the Korea Contents Association
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• v.12 no.3
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• pp.77-83
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• 2012

In this paper, a low-complexity embedded compression (EC) Codec algorithm for the wavelet video coder is applied to reduce excessive external memory requirements. The EC algorithm is used to achieve a fixed compression ratio of 50 % under the near-lossless-compression constraint. The EC technique can reduce the 50 % memory requirement for intermediate low-frequency coefficients during multiple discrete wavelet transform stages compared with direct implementation of the wavelet video encoder of this paper. Furthermore, the EC scheme based on a forward adaptive quantization and fixed length coding can save bandwidth and size of buffer between DWT and SPIHT to 50 %. Simulation results show that our EC algorithm present only PSNR degradation of 0.179 and 0.162 dB in average when the target bit-rate of the video coder are 1 and 0.5 bpp, respectively.