• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.7
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• pp.1617-1625
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• 2010

Recently, the standardization of SVC technology which can provide adaptive video quality in diverse service environments has been completed. This paper proposes a layered encryption technique which takes into account the prioritized layer characteristics of SVC and control constant encryption complexity satisfying the target computational complexity. In particular, it analyzes the importance of NAL unit in the SVC video layer and suggests a method to apply appropriate encryption complexity proper for it. The effectiveness of the proposed method is proved through the comparison of time required for encryption and the reconstructed video quality using imperfect decryption key.

• Proceedings of the Korean Information Science Society Conference
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• 2008.06d
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• pp.224-229
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• 2008

기존의 웹 서비스용 프록시는 숫자와 문자형, 이미지와 같은 크기가 비교적 작은 데이터 전송에는 성공 적인 방법이지만, 스트리밍 데이터의 특성으로 인하여 스트리밍 데이터 전송에는 적합하지 않다. 그래서 대안으로 CDN 같은 방법도 사용되고 있지만 새로운 인프라를 구성해야 하므로 엄청난 비용이 든다. 따라서 기존의 프록시 자원을 활용한 스트리밍 프록시 방법에 대한 많은 연구가 진행 중이다. 이러한 방법들은 스트리밍 데이터 전체를 캐싱하는 데는 많은 문제점들이 있기 때문에 컨텐츠를 보다 작은 단위로 나누어 캐싱하는 방법들이 제안되었다. 컨텐츠를 나누는 방법으로는 각 연구별 다른 기준으로 나누어지는데 본 논문에서는 이미 코딩이 되어 있는 신호를 다른 신호로 바꾸는 트랜스코딩에 프록시를 더해서 균등한 크기로 세그먼트를 나누는 방법의 단점인 데이터의 교체수가 너무 많은 점을 개선하기 위한 혼합 세그먼트 기반의 트랜스코딩 프록시를 제안하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.779-781
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• 2013

As the demand for various multimedia service increases the technology that utilizes image as information transfer method develops rapidly. Though average filter, median filter and weight filter etc. have been proposed to remove various noises that are added to images, the existing methods are short of noise removal and edge reservation performance. Therefore, in this paper an algorithm, in which noise is decided at the first hand, and then it is processed through modified median filter and adaptive weighted average filter, is proposed to effectively remove the complex noise that has been added to an image. And it was compared with existing methods through simulation and PSNR(peak signal to noise ratio) has been used as a criterion.

• Journal of KIISE:Information Networking
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• v.32 no.4
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• pp.525-534
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• 2005

This paper is concerned with the query slipping and its prevention for trajectory-based matchmaking service in wireless sensor networks. The problem happens when a query propagating along a subscribe trajectory moves through a publish trajectory without obtaining desired information, even though two trajectories intersect geometrically. There follows resubmission of the query or initiation of another subscribe trajectory Thus, query slipping results in considerable time delay and in the worst, looping in the trajectory or query flooding the network. We address the problem formally and suggest a solution. First, the area where nodes are distributed is logically partitioned into smaller grids, and a grid-based multicast next-hop selection algorithm is proposed. Our algorithm not only attempts to make the trajectory straight but also considers the nodal density of recipient nodes and the seamless grid-by-grid multicast. We prove that the publishing and subscribing using the algorithm eventually eliminate the possibility of the slipping. It toms out that our algorithm dissipates significantly less power of neighbor nodes, compared to the non grid-based method, as greedy forwarding, and the fixed- sized grid approach, as GAF (Geographical Adaptive Fidelity)

• Journal of Broadcast Engineering
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• v.19 no.2
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• pp.174-183
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• 2014

This paper is to propose a scalable video coding scheme for adaptive digital hologram video service for various reconstruction environments. It uses both the light source information and digital hologram at both the sending side and the receiving side. It is a resolution-scalable coding method that scales the resolution, that is, the size of the reconstructed image. The method compresses the residual data for both the digital hologram and the light source information. For the digital hologram, a lossy compression method is used, while for the light source information, a lossless compression method is used. The experimental results showed that the proposed method is superior to the existing method in the image quality at the same compression ratio. Especially it showed better performance than the existing method as the compression ratio becomes higher.

• Journal of KIISE:Computer Systems and Theory
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• v.32 no.11_12
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• pp.617-628
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• 2005

The services for ubiquitous computing environments have to provide automatically user-specific adaptive services in dynamically changed environments with many informations provided by both a user and his/her environment. Workflows used in business and distributed computing environments support service automation by connecting many tasks with rules or orderings of tasks. Therefore we must specify context information on transition condition to support context-aware services by adapting a workflow to ubiquitous computing environments. In this paper, we present a structural context model to specify context information on transition constraints of the workflow. And we propose an uWDL Ubiquitous Workflow Description Language) based on web services, which provides web service interfaces which are standardized and independent on heterogeneous and various platforms, protocols, and languages. The uWDL tan be interpreted and executed by a workflow engine, and provide users autonomic services based on context-awareness.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• The KIPS Transactions:PartD
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• v.13D no.6 s.109
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• pp.839-846
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• 2006

Context-aware adaptive service for overcoming the limitations of wireless devices and maintaining adequate service levels in changing environments is becoming an important issue. However, most existing studies concentrate on an adaptation module on the client, proxy, or server. These existing studies thus suffer from the problem of having the workload concentrated on a single system when the number of users increases md, and as a result, increases the response time to a user's request. Therefore, in this paper the adaptation module is dispersed and arranged over the client, proxy, and server. The module monitors the contort of the system and creates a proposition as to the dispersed adaptation system in which the most adequate system for conducting operations. Through this method faster adaptation work will be made possible even when the numbers of users increase, and more stable system operation is made possible as the workload is divided. In order to evaluate the proposed system, a prototype is constructed and dispersed operations are tested using multimedia based learning content, simulating server overload and compared the response times and system stability with the existing server based adaptation method. The effectiveness of the system is confirmed through this results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.380-389
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• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• 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).