• Title/Summary/Keyword: 스트림 서비스

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Design and Evaluation of a Buffering Patching Technique for VOD Systems (주문형 비디오 시스템을 위한 버퍼링 패칭 기법의 설계 및 평가)

  • 하숙정;배인한
    • Journal of KIISE:Computer Systems and Theory
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    • v.30 no.10
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    • pp.523-532
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    • 2003
  • Video on Demand(VOD) services cause high resource consumption in a video seuer, because multimedia are characterized by continuous playback, a high bandwidth requirement, and long playback duration. Patching has been proposed to save the network I/O bandwidth of a video server. To achieve true VOD, patching uses multicasting to share video streams, thereby providing immediate VOD services to users without any service latency. A communication channel is used to either multicast the entire video as a regular channel or multicast only the leading portion of a video as a Patching channel. This paper Proposes a buffering patching technique that divides regular channels, as used in patching, into sub-regular channels and regular channels to shorten the holding time of the channels. In the proposed technique, the last portion of video data, corresponding to the size of the buffering window, is not transferred by sub-regular channels, but rather downloaded and buffered in the user buffer from the latest regular channel. When simulations were performed to compare the performance of the proposed technique with that of conventional patching, the results indicated that the proposed technique was superior in terms of the defection rate, average service latency, and fairness, although the amount of video data buffered at each user station was higher than that with patching.

A Traffic Management Scheme for the Scalability of IP QoS (IP QoS의 확장성을 위한 트래픽 관리 방안)

  • Min, An-Gi;Suk, Jung-Bong
    • Journal of KIISE:Information Networking
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    • v.29 no.4
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    • pp.375-385
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    • 2002
  • The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

Bit-rate control of transcoder considering the activity and direction of motion vector (움직임 벡터의 활동성과 방향성을 고려한 트랜스 부호화기의 비트율 제어)

  • 구성조;장해원;황찬식
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.12C
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    • pp.1209-1216
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    • 2003
  • The transcoding is needed to modify the bit rate of the video bitstream in accordance with the bandwidth of a variety of channels in order to provide users with a realtime multimedia service and more stable and credible VOD(Video On Demand) service on various wire/wireless network. This paper suggests how to guarantee the video quality in the transcoding process, taking the activity and direction of the macroblock into consideration, thereby reducing the calculation. The threshold of the macroblock's activity and direction is applicable to the different videos, with the DCT coefficients from the intermediate data in the decoding process. Moreover, the suggested video transcoding algorism enables us to maintain the proper video quality and reduce the calculation through the video bit rate change experiment in H.263 environment.

A Study of Real-time SVC Bitstream Extraction for QoS guaranteed Streaming (QoS 기반 스트리밍 서비스를 위한 실시간 SVC 비트스트림 추출기에 대한 연구)

  • Kim, Duck-Yeon;Bae, Tae-Meon;Kim, Young-Suk;Ro, Yong-Man;Choi, Hae-Chul;Kim, Jae-Gon
    • Proceedings of the IEEK Conference
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    • 2005.11a
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    • pp.513-516
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    • 2005
  • SVC(Scalable Video coding) is the standardization of MPEG that aims to support multi spatial, temporal, and quality layers. We can support the video service guaranteeing the QoS in varied network circumstance by using SVC bitstream. In this paper, we propose the real-time SVC bitstream extractor that is able to extract the bitstream with varied frame rate and SNR quality in real-time. To do, extraction processing needs to be performed by GOP unit. As well , essential bitstream information for real-time extraction is aquired before doing extraction process. The proposed method is implemented by using JSVM 2.0. Experimental results show that the proposed method is valid.

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A Study on LMMSE Receiver for Single Stream HSDPA MIMO Systems using Precoding Weights (Single Stream HSDPA MIMO 시스템에서 Precoding Weight 적용에 따른 LMMSE 수신기 성능 고찰)

  • Joo, Jung Suk
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.4
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    • pp.3-8
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    • 2013
  • In CDMA-based systems, recently, researches on chip-level equalization have been studied in order to improve receiving performance when supporting high-rate data services. In this paper, we consider a chip-level LMMSE (linear minimum mean-squared error) receiver for D-TxAA (dual stream transmit antenna array) based single stream HSDPA MIMO systems using precoding weights. First, we will derive precoding weights for maximizing the total instantaneous received power. We will also analyze the effects of both transmit delay of precoding weights and mobile velocity on chip-level LMMSE receivers, which is verified through computer simulations in various mobile channel environments.

Minimizing the Total Stretch when Scheduling Flows of Divisible Requests without Interruption (총 스트레치 최소화를 위한 분할 가능 리퀘스트 흐름 스케줄링)

  • Yoon, Suk-Hun
    • The Journal of Society for e-Business Studies
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    • v.20 no.1
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    • pp.79-88
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    • 2015
  • Many servers, such as web and database servers, receive a continual stream of requests. The servers should schedule these requests to provide the best services to users. In this paper, a hybrid genetic algorithm is proposed for scheduling divisible requests without interruption in which the objective is to minimize the total stretch. The stretch of a request is the ratio of the amount of time the request spent in the system to its response time. The hybrid genetic algorithm adopts the idea of seed selection and development in order to improve the exploitation and exploration power of genetic algorithms. Extensive computational experiments have been conducted to compare the performance of the hybrid genetic algorithm with that of genetic algorithms.

Scheme for transmitting Data and TDM based on E-PON (E-PON 기반 데이터 및 TDM 전달을 위한 방안)

  • Jin, Geol;Park, Chun-Kwan
    • Proceedings of the Korean Institute of Intelligent Systems Conference
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    • 2007.04a
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    • pp.465-468
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    • 2007
  • This paper addresses the scheme for transmitting Data and TDM signals based on E-PON. E-PON technology, that combines low-cost Ethernet technology and optical fiber infra-structure, has been appeared as a solution of next generation access network. The transmission speed of E-PON is 1Gbps and symmetric in both direction, such as downstream and upstream. Therefore, it is possible to save the cost through sample network architecture, efficient operation, and low maintenance cost of optical IP Ethernet network. By adding TBMoIP(Time Division Multiplexing over Internet Protocol) module to this E-PON system, and implementing QoS(Quality of Service) control function, this system can provide data and TDM service efficiently.

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Implementation of the Audio CODEC for Digital Audio Broadcasting Service (디지털 오디오 방송 서비스를 위한 오디오 코덱의 구현)

  • 장대영;홍진우
    • Journal of Broadcast Engineering
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    • v.6 no.1
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    • pp.66-71
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    • 2001
  • This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

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Comparison of Collaboration System for e-Science (e-Science 협업 연구를 위한 협업 시스템 비교 분석)

  • Ryu, Jinseung;Kim, Beobkyun;Oh, Chungshick;Jang, Hangjin
    • Proceedings of the Korea Contents Association Conference
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    • 2007.11a
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    • pp.565-568
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    • 2007
  • e-Science is for increasing the research productivity in epochal way using research resources like high performance computers, large capacity data storage, experimental instrument and researchers. For the research activities without being limited by time and space, the collaboration system is prerequisite condition. This paper shows the comparisons and analysis of features of popular collaboration systems in the point of e-Science. We considered that the number of supporting video streams, scalability and cost are main issues for the e-Science collaboration system. By these, we conclude that Access-Grid(AG) is best suitable for the e-Science collaborative research system. This paper shows the current AG status and the future way of AG.

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(A Study on an Adaptive Multimedia Synchronization Scheme for Media Stream Transmission) (미디어 스트림 전송을 위한 적응형 멀티미디어 동기화 기법에 관한 연구)

  • 지정규
    • Journal of the Korea Computer Industry Society
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    • v.3 no.9
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    • pp.1251-1260
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    • 2002
  • Real-time application programs have synchronization constraints which need to be met between media-data. Synchronization method represents feedback method including virtual client-side buffer. This buffer is used in buffer level method. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control. It is the important factor for playback rate and QoS if the buffer level is normal or not. To solve the problems, we can control the start of transmission in multimedia server by appling filtering, control and network evaluation function. Synchronization method is processing for smooth presentation without cut-off while media is playing out. When audio frame which is master media is in high threshold buffer level we decrease play out time gradually, otherwise we increase it slowly.

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