• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.102-104
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• 2011

최근 비디오 스트리밍과 대화형 비디오 서비스 등과 같은 광대역 멀티미디어 서비스를 지원하기 위하여 Wimax와 같은 4G 무선네트워크 시스템 기술이 발전해 왔다. 4G 무선네트워크의 OFDM(Orthogonal Frequency Division Multiplexing)과 MIMO(multi Input Multi Output)은 사용자들에게 매우 유연한 QoS(Quality of Service) 서비스를 제공해 줄 수 있다.[1] 이 논문에서는 다양한 네트워크 상황에서 멀티캐스트 그룹에게 효율적인 방법으로 통신 자원을 할당하기 위해 OFDM 방법을 사용 하였다. 이에 본 논문에서는 한 셀(cell) 내의 서로 다른 멀티캐스트의 그룹의 다른 SNR(Signal to noise Ratio)의 사용자 분포에 따른 적응적인 scalable 비디오 멀티캐스트 방식을 제안한다. 더 나은 수신율을 가진 사용자는 최적의 MCS(Modulation and Coding Scheme) 할당을 통해 서로 다른 화질의 scalable 비디오 계층 중 높은 해상도의 비디오를 받을 수 있다. 논문에서는 전체 전송률을 최적화 하는 대신 전송받은 전체 비디오의 평균 화질을 최적화하는 방법을 제안한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.04a
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• pp.1166-1171
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• 2000

기존의 멀티미디어 연구의 실현에 있어 가장 큰 문제라 할 수 있던 성능의 문제가 하드웨어의 급속한 발달로 해결되어 감에 따라 멀티미디어 및 제반 관련기술도 함께 발전되었으며 이에 기반한 multimedia stream에서의 event를 검출하기 위한 다양한 연구들이 진행되어 왔다. 그러나 지금까지의 연구는 주로 전송 및 저장, 검색에 집중되어 연구되어 왔으며 영상인식 등의 Vision관련 연구에서는 멀티미디어 스트리밍 기술과의 연동을 고려하지 않은 연구를 수행함에 따라 검출 가능한 event가 있다고 하더라도 응용영역에 종속적인 인테페이스만을 고려함에 따라 사용자가 이를 기술(記述, description)하거나, 사용자에게 검출 가능한 event를 제시하기 위해 일반화된 방법이 제시되어 있지 않았다. 본 연구에서는 사용자가 검출을 원하는 event를 기술하는 방법과, 시스템에서 검출 가능한 event를 제시하기 위한 방법을 제안하고, 제시되는 방법이 응용영역에 독립적이기 위해 요구되는 사항들과 객체 단위인 이벤트/행위와 처리기 사이의 인터페이스에 관하여 정의한 후 기본적인 동작방식을 제안한다.

• Journal of KIISE:Computer Systems and Theory
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• v.31 no.5_6
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• pp.268-278
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• 2004

In this paper, we describe our experience in the design and implementation of the SMART file system to handle multimedia workload. Our work has three design objectives: (ⅰ) efficient support for sequential workload, (ⅱ) avoiding disk fragmentation, (ⅲ) logical unit based file access. To achieve these three objectives, we develop a file system where a file consists of linked list of Data Unit Group. Instead of tree like structure of the legacy Unix file system, we use single level file structure. Our file system can also access the file based upon the logical unit which can be video frame or audio samples. Data Unit Group is a group of logical data units which is allocated continuous disk blocks. At the beginning of each Data Unit Group, there exists an index array. Each index points to the beginning of logical data units, e.g. frames in the Data Unit Group. This index array enables the random access and sequencial access of semantic data units. SMART file system is elaborately tailored to effectively support multimedia workload. We perform physical experiments and compare the performance of SMART file system with EXT2 file system and SGI XFS file system. In this experiment, SMART file system exhibits superior performance under streaming workload.

• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.3
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• pp.285-289
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• 2004

Recently, due to the high development of the internet, needs for multimedia streams such as digital audio and video is increasing much more. In case of transmitting multimedia streams using the User Datagram Protocol (UDP), it may cause starvation of TCP traffic on the same transmission path, thus resulting in congestion collapse and enormous delay because UDP does not perform TCP-like congestion control. Because of this problem, diverse researches are being conducted on new transmission schemes and protocols intended to efficiently reduce the transmission delay of real-time multimedia streams and perform congestion control. The TCP-friendly congestion control schemes can be classified into the window-based congestion control, which uses the general congestion window management function, and the rate-based congestion control, which dynamically adjusts transmission rate by using TCP modeling equations and the like. In this paper, we suggest the square-root congestion avoidance algorithm with the hybrid TCP-friendly congestion control scheme which the window-based and rate-based congestion controls are dealt with in a combined way. We apply the proposed algorithm to the existing TEAR. We simulate the performance of the proposed TEAR by using NS, and the result shows that it gives better improvement in the stability needed for providing congestion control than the existing TEAR.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.178-180
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• 2003

이동 컴퓨팅 환경에서의 프록시는 트랜스코딩과 캐슁 등의 방법으로 미디어 스트림을 재가공하여 멀티미디어 스트리밍의 연속성을 보장해 준다. 프록시에 대한 기존의 접근들은 대체로 트랜스코딩과 캐슁을 별개의 주제로 다루었었다. 따라서 본 논문에서는 트랜스코딩과 캐슁을 통할하여 연동시킨 프록시 시스템 아키텍처를 구현하였다. 또한 서버와 프록시 사이의 전송 지연. 프록시에서의 미디어 스트림 재가공에 걸리는 지연, 프록시와 클라이언트 사이의 bandwidth 자원 등을 창조하여 adaptive한 트랜스코딩 및 캐슁을 할 수 있도록 해주는 transco-prefix caching에 대해 제안하였다. 그 결과 transco-prefix caching을 통하여 트랜스코딩과 캐슁의 장점들을 동시에 살필 수 있었다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.3
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• pp.476-481
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• 2012

Recently, QoS/QoE technology as a part of the HTTP-based adaptive streaming technology has attracted attention. In this paper, we find out HTTP-based adaptive and dynamic streaming technology. It is based on a dynamic adaptive streaming system over HTTP, which is designed and implemented. The system consist of conversion by the bit rate of MPEG2-TS files, division of segment, creation of MPD(Media Presentation Description) and bitrate's player through analysis of MPD File between servers and clients in network environments of dynamic and adaptive. This diverse network environments, continuous and smooth playback of video will be used in various multimedia fields.

• Proceedings of the Korea Contents Association Conference
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• 2003.05a
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• pp.198-201
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• 2003

Changes in internet and network environment make it possible to provide high-quality content services in real time. As demand for digital content is increased, problems related to intellectual property rights are getting more important. Streaming service like video-on-demand solved this problem by preventing content from being saved. But, as the advent of several tools able to save streamed content, the streamed content is not free from these problems any more. So, with security countermeasure like access control, new technologies to control and manage rights for content are needed. One of the solutions is DRM In this paper, we describe a DRM-based streaming service that can send the ASF stream which is the multimedia file format of Microsoft.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1B
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• pp.25-37
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• 2008

Network streaming technology is a service method which can support several application software via streaming in networks. Although multimedia streaming technology has been developed and widely used, research on the software streaming technology is not yet being progressed actively. In this paper, we design and implement a software streaming technology based on Java language with enhanced performance, compare the existing related systems with it and analyze several characteristics, and perform the performance evaluation. The implemented system has a platform to be used in multiple operating systems in addition to Windows system. It can also be used without installing a software at the client's computer, and has a novel feature that the user's convenience is maximized.

• Journal of KIISE:Information Networking
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• v.37 no.6
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• pp.483-488
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• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.493-505
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• 2003

A new inter-client synchronization framework for multicast media streaming is proposed employing a server-client coordinated adaptive playout control. The proposed adaptive player controls the playback speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. Additionally, the server implicitly helps increasing the time available for retransmission while the clients perform an interactive error recovery mechanism with the assistance of playout control. The network-simulator based simulations show that the proposed framework can reduce the playout discontinuity without degrading the media quality, and thus mitigate the client heterogeneity.