• Proceedings of the KIEE Conference
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• 2004.05a
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• pp.68-70
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• 2004

In this paper we develop an adaptive interface between video compression and transission protocols to handle QoS fluctuations that are common to mobile communication systems. We consider various generic design alternatives for QoS adaptation and identify 'QoS negotiation' as the most promissing. This method gives the best possibilities to obtain system-wide efficiency. To handle the indued system complexity we apply a design philosophy (called ARC) that separates implementation dependencies by introducing QoS interfaces between system modules. In the ARC phlosophy the implementation details are hidden in the subsystems. To assure efficient adaptation, the QoS must be negotiated between modules. We select the QoS parameters that are both necessary and sufficient for efficient negotiation between the video encoder and protocol modules. We describe the relation between the QoS parameters at the interface and the internal parameters of common video coding methods and protocol elements. Furthermore, we describe a negotiation procedure that allows a system-wide optimum to emerge.

• Proceedings of the Korea Information Processing Society Conference
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• 2004.05a
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• pp.139-142
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• 2004

본 논문에서는 MPEG-21 Digital Item Adaptation(DIA)에 의한 워터마크 신호의 영향을 실험하고 분석한다. MPEG-21 DIA에서는 다양한 소비환경에 맞게 멀티미디어 컨텐츠를 변할 수 있는 기능들을 제공하고 있다. 그러나 컨텐츠 변환기능들은 저작권 보호를 위해 컨텐츠에 삽입된 워터마크신호를 홰손시킬 수 있으므로, DIA 환경에서 워터마킹기술을 사용하기 위해서는 워터마킹기술에 대한 DIA의 영향을 분석할 필요가 있다. 본 논문에서는 일반적으로 널리 알려진 대표적인 워터마킹기술을 이용하여 MPEG-21 DIA에서 정의하고 있는 각각의 적응변환기능에 대한 워터마크의 강인성을 실험하여, 그 결과를 바탕으로 DIA 환경에서 워터마킹기술을 적용할 때 필요한 요구사항을 분석하였다.

• Journal of KIISE:Software and Applications
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• v.35 no.4
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• pp.207-218
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• 2008

With the growth of the Internet, multimedia streaming becomes an important means to deliver video contents over the Internet and the amount of the streaming multimedia contents is also getting increased. However, it becomes difficult to guarantee the quality of service in real-time over the IP network environment with instantaneously varying bandwidth. In this paper, we propose an optimal adaptation framework for streaming contents over the Internet in the sense that the perceptual quality of the multi-angie content with multiple visual objects is maximized given the constraints such as available bandwidth and transcoding cost. In the multi-angle video service framework, the user can select his/her preferred alternate views among the given multiple video streams captured at different view angles for a same event. This enhanced experience often entails streaming problems in real-time over the network, such as instantaneous bandwidth changes in the Internet. In order to cope with this problem, we assume that multi-angle video contents are encoded at different bitrates and the appropriate video streams are then selected or transcoded for delivery to meet such bandwidth constraints. For the user selective consumption of the various bitstreams in the multi-angle video service, the bitstream in each angle can be encoded in various bitrate, and the user can select a sub-bitrstream in the given bitrstreams or transcode the corresponding content in order to deliver the optimally adapted video contents to the instantaneously changing network condition. Therefore, we define the transcoding cost which means the time taken for transcoding the video stream and formulate a unified optimization framework which maximizes the perceptual quality of the multiple video objects in the given constraints such as the transcoding cost and the network bandwidth. Finally, we present plenty of the experimental results to show the effectiveness of the proposed method.

• Journal of KIISE:Software and Applications
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• v.33 no.2
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• pp.220-230
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• 2006

As the computing technique and mobile devices get developed, the demand of multimedia contents for mobile environment has been increased. The multimedia contents provided on PDA has so far been limited to the materials such as video and audio. MPEG-4 is the international standard used for supporting to properly save, communicate multimedia in formation such as video, audio, image, text and two-dimensional object, which can present various, multimedia contents by using adaptation techniques. However, since most MPEG-4 contents are not used for PDA but desktop, it seems not to be played on PDA where needs low power consumption, limited memory capability and GUI, and so on. In this paper, we propose the adaptation techniques, which can present the MPEG-4 contents on PDA, using scene composition with MPEG-4. The proposed scheme consists of three subparts: physical adaptation, variation adaptation and resource adaptation. Physical adaptation adjusts the physical difference of between the authoring environment and playback environment. Event adaptation part transforms events used for desktop to the events used for playback on PDA. The resource adaptation enables efficiency of playback to promote by using the essential information table on BIFS parser. As the proposed scheme is applied to MPEG-4 player, we see that the MPEG-4 contents are efficiently played on PDA.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.385-389
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• 2009

Multi-hop wireless mesh networks (WMNs) suffer from significant packet losses due to insufficient available bandwidth and high channel error probability. To conquer packet losses, end-to-end (E2E) error control schemes have been proposed. However, in WMNs, E2E error control schemes are not effective in adapting to the time-varying network condition due to large delay. Thus, in this paper, we propose a network-adaptive error control for video streaming over WMNs that flexibly operates E2E and hop-by-hop (HbH) error control according to network condition. Moreover, to provide lightweight support at intermediate nodes for HbH error control, we use path-partition-based adaptation. To verify the proposed scheme, we implement it and evaluate its transport performance through MPEG-2 video streaming over a real IEEE 802.11a-based WMN testbed.

• International Journal of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.166-172
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• 2020

Video streaming application such as YouTube is one of the most popular mobile applications. To adjust the quality of video for available network bandwidth, a streaming server provides multiple representations of video of which bit rate has different bandwidth requirements. A streaming client utilizes an adaptive bit rate scheme to select a proper video representation that the network can support. The download behavior of video streaming client player is governed by several parameters such as maximum buffer size. Especially, the size of the maximum playback buffer in the client player can greatly affect the user experience. To tackle this problem, in this paper, we propose the maximum buffer size optimization according to available network bandwidth and buffer status. Our simulation study shows that our proposed buffer size optimization scheme successfully mitigates playback stalls while preserving the similar quality of streaming video compared to existing ABR schemes.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1419-1422
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• 2003

Multimedia contents what we call Digital Items include various types of resources such as music, speech, text, video, graphics, and so on. The current Adaptation QoS described in the ISO/IEC 21000-7 CD-Part 7: Digital Item Adaptation, however, lacks adaptation methods for audio signals. The goal of this paper is to provide adaptation methods that are necessary to deal with audio signals. Two operations are introduced in order to adapt audio items. One method is to make use of the functionality of Fine Grain Scalability, and the other is intended to drop the channel of audio output channel. This paper provides a DIA description tool that associates the operators with the corresponding values of the constraint and the utility. Furthermore, the operations are evaluated and compared to alternative solutions.

• Journal of Communications and Networks
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• v.1 no.2
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• pp.134-142
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• 1999

We have developed fully functional Video Server and Client applications which can transmit, receive, decompress and display compressed video over various networks. Our video trans-port allows dynamic rate control feedback, loss detection, and repair requests from Clients to the Server. Our experiments show how feedvack-before-degradation scheme for rate adaptation maintains good display frame-rate for video playback. We show how the playback degradation(reduction in display frame-rate) oc-curs and what happens if corrective measures are not taken to im-prove the situation. The degradation is attributed to the increased internal kernel buffering which consumes scarce CPU resources. We demontrate with our experimental results that ServerNet, with improved hardware delivery guarantees, can significantly reduce host CPU resource consumption while serving video streams. We present the maximum number of streams which can be served for each of ATM and ServerNet interconnects. The appropriate user-level packet size for the video server are also determined for each case.

• Journal of Communications and Networks
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• v.14 no.3
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• pp.286-295
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• 2012

To meet the increased demand of video-on-demand (VoD) services, peer-to-peer (P2P) mesh-based multiple video approaches have been recently proposed, where each peer is able to find a video segment interested without resort to the video server. However, they have not considered the constraint of the server's upload bandwidth and the fairness between upload and download amounts at each peer. In this paper, we propose a novel P2P VoD streaming system, named peer cache adaptation (PECAN) where each peer adjusts its cache capacity adaptively to meet the server's upload bandwidth constraint and achieve the fairness. For doing so, we first propose a new cache replacement algorithm that designs the number of caches for a segment to be proportional to its popularity. Second, we mathematically prove that if the cache capacity of a peer is proportional to its segment request rate, the fairness between upload and download amounts at each peer can be achieved. Third, we propose a method that determines each peer's cache capacity adaptively according to the constraint of the server's upload bandwidth. Against the proposed design objective, some selfish peers may not follow our protocol to increase their payoff. To detect such peers, we design a simple distributed reputation and monitoring system. Through simulations, we show that PECAN meets the server upload bandwidth constraint, and achieves the fairness well at each peer. We finally verify that the control overhead in PECAN caused by the search, reputation, and monitoring systems is very small, which is an important factor for real deployment.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.433-435
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• 2001

인터넷상에서 화상전화를 구현할 경우, 대용량의 멀티미디어 데이터의 전송은 네트웍 congestion의 원인이 될 수 있다. 이러한 congestion이 일어날 경우, congestion avoidance 알고리즘을 적용하는 TCP 데이터는 스스로 전송률을 줄이게 되므로, congestion 정책을 사용하지 않는 UDP 패킷과 같은 데이터와 함께 전송될 경우, TCP 데이터에 불리하게 된다. 이때, UDP 패킷 데이터에 TCP와 유사한 방법의 congestion avoidance 알고리즘을 적용하여 이를 해결할 수 있는데, 이것은 TCP-friendly Adaptation 알고리즘이다. 본 논문에서는 인터넷 화상전화의 구현에 대해 기술하고 인터넷 환경에서 화성전화를 사용할 때에 congestion을 control 하기 위해서, 그 출력 대역폭을 네트웍 상태에 따라 TCP와 유사한 방식으로 조절하는 TCP-friendly Adaptation 알고리즘을 적용한다.