• Journal of Internet Computing and Services
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• v.8 no.2
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• pp.77-87
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• 2007

A typical VOD service allows that a number of remote clients playback a desired video from a large collection of videos stored in one or more video servers. The main bottleneck for a VOD service is the network bandwidth connecting to the VOD server to the client due to the high bandwidth requirements. Many previous researches have shown that VOD server can be greatly improved through the use of multicast, broadcast, or P2P scheme. Broadcast is one of the most efficient techniques because it can transmit a stream to many users without additional network bandwidth. But the broadcast has long latency time. In order to overcome the drawback, in this paper, we propose P2Prefix broadcast scheme that can solve the service latency time, which is the problem of broadcast scheme, by using P2P caching as well as minimizing the client buffer requirement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.932-939
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• 1998

In this paper, we analyze the performance of a multicode-CDMA system which have been proposed for the multimedia communications. The performance of a multicode-CDMA system, providing good spectrum efficiency as well as serving various bit rates, is analyzed with multipath, frequency selective, slowly fading Rayleigh channel. Also the proposed scheme adopting RAKE receiver with MRC(Maximal Ratio Combine) is advantageous to multipath channel. For a practical channel modeling, the JTC(Joint Technical Committee) recommended channel model(JTC(AIR) 23-065R6) is applied to simulation. The proposed schemehas serial-to-parallel convertor which splits input data stream of 2 Mits/s into 20 branches o 100 kbits/s. From the result of simulation, the case of RAKE receiver with 3 fingers to reduce the system complexity required the relatively large $E_{b}/N_O$ of 0 dB~1.5 dB, compared to the case of RAKE receiver with the number of path finger to keep the average error rate to be $1{\times}10^{-3}$ in channel A.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1490-1500
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• 1996

One of the important issues for multimedia communications over ATM netowrks is efficient use of net-work resources since transmitting video at low cost requires high utilization of channel bandwidth. In this paper, we propose an effective bandwidth estimation scheme for MPEG sources in ATM networks. It has been known difficult to allocate effective transmission bandwidth to MPEG sources due to its bursty charactersitics. The MPEG traffic stream can bemodeled using five traffic parameters:service rate, burst load ratio, interburst load ratio, burst period, and interburst period. Using these parameters, a transmission bandwidth for each source can be estimated, and the estimated bandwidth is applied synchronously to reduce the queue size in a buffer. For a simple network modelof a high speed link that multiplexes a number of virtual-circuit connections, simulations on adaptive bandwidth allocation were performed, and the results show that the queueing delay is significantly reduced, when compared to a fixed bandwidth allocation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.6
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• pp.1076-1085
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• 2004

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. And there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for the payload of small size. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of IP/UDP protocols to reduce the overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented in Linux environment. Using the Header compression system designed between a server and clients provides have the advantage of effective data throughput in network. Since the minimum packet size in Ethernet is 64 bytes, the amount of reduction by header compression in practical chatting environment was 6.6 bytes.

• Journal of Korea Game Society
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• v.7 no.2
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• pp.33-40
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• 2007

Smoothing is a transmission plan where variable rate video data is converted to a constant bit rate stream. Among them are CBA, MCBA, MVBA and others. However, these smoothing algorithms produce a transmission plan where extra bandwidth in server is not considered. This may cause difficulty in providing videos to many clients in a server. In this paper, we propose the smoothing algorithm with monotonic transmission rate increase and abrupt transmission rate decrease, in order to provide videos to as many clients in server with limited network bandwidth as possible. In order to show the performance of our proposed algorithm, various evaluation factors were used such as the maximum number of clients, the average number of clients, and so on. Experiments demonstrated that the proposed algorithm outperformed other algorithms in evaluation factors such as the maximum number of clients and the average number of clients.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.571-573
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• 2014

In today's technology streaming service's QoS technologiey is an issue. On quality video streaming service there are some technical issues exist, such as buffering. This submission is "Adaptive dual-streaming system design" which is for the integrity of the data streaming that is sent to TCP and UDP for faster transmission of data to the stream. This system provides real-time incoming video encoding in bitrate of h.264-based H through a process based on the video footage of several server and client-to-TCP and UDP via Adaptive providing streaming services in a network environment. This is an unspecified number of buffers in a network environment and continued through the minimization of various streaming for playback of videos and multimedia will be utilized in the field.

• Journal of KIISE:Software and Applications
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• v.27 no.5
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• pp.529-542
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• 2000

Recent years, the use of multimedia information is rapidly increasing, and the video media is the most rising one than any others, and this field Integrates all the media into a single data stream. Though the availability of digital video is raised largely, it is very difficult for users to make the effective video access, due to its length and unstructured video format. Thus, the minimal interaction of users and the explicit definition of video structure is a key requirement in the lately developing image and video management systems. This paper defines the terms and hierarchical video structure, and presents the system, which construct the clustering-based video hierarchy, which facilitate users by browsing the summary and do a random access to the video content. Instead of using a single feature and domain-specific thresholds, we use multiple features that have complementary relationship for each other and clustering-based methods that use normalization so as to interact with users minimally. The stage of shot boundary detection extracts multiple features, performs the adaptive filtering process for each features to enhance the performance by eliminating the false factors, and does k-means clustering with two classes. The shot list of a result after the proposed procedure is represented as the video hierarchy by the intelligent unsupervised clustering technique. We experimented the static and the dynamic movie videos that represent characteristics of various video types. In the result of shot boundary detection, we had almost more than 95% good performance, and had also rood result in the video hierarchy.

• The KIPS Transactions:PartB
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• v.12B no.6 s.102
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• pp.637-646
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• 2005

Most of the multimedia contents for DBM services art provided as MPEG-2 bit streams. However, they have to be transcoded to H.264 bit streams for practical services because the standard video codec for DMB is H.264. The existing transcoder architecture is Cascaded Pixel-Domain Transcoding Architecture, which consists of the MPEG-2 dacoding phase and the H.264 encoding phase. This architecture can be easily implemented using MPEG-2 decoder and H.264 encoder without source modifying. However. It has disadvantages in transcoding time and DCT-mismatch problem. In this paper, we propose two kinds of transcoder architecture, DCT-OPEN and DCT-CLOSED, to complement the CPDT architecture. Although DCT-OPEN has lower PSNR than CPDT due to drift problem, it is efficient for real-time transcoding. On the contrary, the DCT-CLOSED architecture has the advantage of PSNR over CPDT at the cost of transcoding time.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.1 s.307
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• pp.17-26
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• 2006

This paper proposes a novel hidden Markov model (HMM)-based gesture recognition method and applies it to an HCI to control a computer game. The novelty of the proposed method is two-fold: 1) the proposed method uses a continuous streaming of human motion as the input to the HMM instead of isolated data sequences or pre-segmented sequences of data and 2) the gesture segmentation and recognition are performed simultaneously. The proposed method consists of a single HMM composed of thirteen gesture-specific HMMs that independently recognize certain gestures. It takes a continuous stream of pose symbols as an input, where a pose is composed of coordinates that indicate the face, left hand, and right hand. Whenever a new input Pose arrives, the HMM continuously updates its state probabilities, then recognizes a gesture if the probability of a distinctive state exceeds a predefined threshold. To assess the validity of the proposed method, it was applied to a real game, Quake II, and the results demonstrated that the proposed HMM could provide very useful information to enhance the discrimination between different classes and reduce the computational cost.

• Journal of Digital Convergence
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• v.16 no.2
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• pp.361-366
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• 2018

The purpose of this study is to examine the four aspects of design terminology : dictionary, legal, academic, and popular according to the changes of various media and the stream of times and to establish a basic framework for design terms that can be refined in time. Based on an understanding of the essential meaning of design terms, it provides the basis for establishing the basic direction of the use of design terms that can be used between the design industry and other industries. The research methods are as follows: First, the preliminary meaning of design terms was examined through previous studies. Second, the present state of use of design terms was analyzed using the Internet and various network media. Third, through interviews with experts in each design field, the criteria of contemporary design terms was explored. Ultimately, the direction to establish design terms was derived through these research processes. The result of this study is meaningful to provide the new index of the time and conceptual change of design terms. In the future, it will be necessary to expand the research on design common terms continuously in design-related academia and related industries, and try to establish professional and popular design terms through diverse opinions.