• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1330-1338
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• 2015

In wireless communication network, the amount of packet data for the video streaming in the playout buffer of the receiver is changed with time according to network condition. If the amount of packet data is less than a specific buffer amount, the buffer underflow problem is generated. On the contrary, if the amount of packet data is more than a given buffer amount, the buffer overflow problem is occurred. When the playout of the video streaming is processed, these buffer underflow and overflow problems cause stop and skip phenomenons and then provide the discontinuity of playout. Therefore, to solve the buffer underflow and overflow problems of the video streaming in wireless communication network, This paper analyzes the combined effect of Token Bucket scheme, which controls the bursty traffic, and AMP(Adaptive Media Playout) scheme, which adaptively changes the playout speed of receiver. Through simulation, we found that the combination of Token Bucket and AMP schemes provides the superiority in terms of the occurrence number of buffer underflow and overflow, the stop duration time and the number of removed frames generated by underflow and overflow, and PSNR.

• The KIPS Transactions:PartA
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• v.16A no.5
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• pp.327-338
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• 2009

Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose a network adaptive prefetching technique, PAP, for guarantee of constant QoS. The PAP prefetches frames by increasing the frame transmission rate while the available network bandwidth is high. The PAP uses the prefetched frames to guarantee the QoS while the available network bandwidth is low and increases the playout interval to prevent buffer underflow. The experiment result shows that the proposed PAP could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10B
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• pp.899-915
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• 2003

Delivering rich and continuous media contents robustly over a wide range of network conditions of the wired/wireless Internet is a highly challenging task. To address this challenges, the continuous media applications at the edge of network has become more and more adaptive while the best-effort Internet is slowly progressing towards improved networking services. That is, the role of network adaptive media delivery, which dynamically links the quality demand of application contents to the underlying networking services, has become more crucial. In this paper, we will first review the required network adaptation functionalities seen from the application side: congestion control / rate control, error control, and synchronization / adaptive playout. Then, we start the coverage of networking support issues that helps the realization of network adaptive media streaming - from network support and protocol support toward consolidated support via middleware. Finally, we propose a dynamic network adaptation framework that efficiently leverages its awareness of both media application (including contents) and underlying networking support.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.649-655
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• 2000

Real-time application programs have constraints which need to be met between media-data. These constraints represents the delay time ad quality of service between media-data to be presented. In order to efficiently describe the delay time and quality of service, a new synchronization mechanism is needed. Proposed paper is a dynamic synchronization that minimized the effects of adaptive transmission delay time. That is, the method meets the requirements of synchronization between media-dat by handling dynamically the adaptive waiting time resulted from variations of delay time. In addition, the mechanism has interval adjustment using maximum delay jitter time. This paper decreases the data loss resulted from variation of delay time and from loss time of media-data by means of applying delay jitter in order to deal with synchronization interval adjustment. Plus, the mechanism adaptively manages the waiting time of smoothing buffer, which leads to minimize the gap from the variation of delay time. The proposed paper is suitable to the system which requires the guarantee of high quality of service and mechanism improves quality of services such as decrease of loss rate, increase of playout rate.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.953-960
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• 2002

Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.2
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• pp.449-454
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• 2008

As demand for the application in multimedia networks is increasing rapidly, it is important to provide these services in Mobile Environments (ME). Obtaining to multimedia services which satisfy synchronization constraints in ME and improving the delay time and Quality of Service(QoS) between media streams to be presented, new specification model has to be proposed. In this paper we propose Mobile Synchronization Model(MSM) as a new specification model for describing efficiently the QoS and the delay time. Also, we propose the multimedia synchronization algorithm where is a dynamic synchronization method that minimizes the effects of adaptive transmission delay time. The proposed model has higher the guarantee of QoS such as the loss rate and the playout rate than it of the previous work.