• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권8호
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• pp.3856-3872
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• 2018

HTTP-based adaptive streaming (HAS) has recently been widely deployed on the Internet. In the HAS system, a video content is encoded at multiple bitrates and the encoded video content is segmented into small parts of fixed durations. The HAS client requests a video segment and stores it in the playout buffer. The rate adaptation algorithm employed in HAS clients dynamically determines the video bitrate depending on the time-varying bandwidth. Many studies have shown that an efficient rate adaptation algorithm is critical to ensuring quality-of-experience in HAS systems. However, existing algorithms have problems estimating the network bandwidth because bandwidth estimation is performed on the client-side application stack. Without the help of transport layer protocols, it is difficult to achieve accurate bandwidth estimation due to the inherent segment-based transmission of the HAS. In this paper, we propose an alternative approach that utilizes the playout buffer occupancy rather than using bandwidth estimates obtained from the application layer. We start with a queueing analysis of the playout buffer. Then, we present a buffer-aware rate adaptation algorithm that is solely based on the mean buffer occupancy. Our simulation results show that compared to conventional algorithms, the proposed algorithm achieves very smooth video quality while delivering a similar average video bitrate.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 II
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• pp.959-962
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• 2003

At the processing capabilities and operating frequency of embedded system are growing, so is the needed data bandwidth to fully utilize the processing capability. The ability to transfer huge amount of data between the embedded core and external devices is required for efficient system operation. In this paper, the data communication architecture for the mixed-clock system is proposed. The dynamic priority adaptation algorithm for bus arbitration is proposed for bandwidth guarantee. The communication architecture that incorporates the proposed arbitration algorithm adapts the priority of communication components dynamically based on the information from FIFO. The experiments show that the measured bandwidth of each component traces the required bandwidth well compared to the other arbitration algorithms

• 한국통신학회논문지
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• 제39B권6호
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• pp.341-349
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• 2014

Dynamic adaptive streaming over HTTP (DASH)는 최근 주목받고 있는 스트리밍 기술이다. DASH 클라이언트는 연속적으로 가용 네트워크 대역폭을 측정하고, 이를 기반으로 향후 다운받을 비디오 세그먼트의 비트율을 결정한다. 본 고에서는 최소 재생 버퍼 레벨을 보존하여 가변적인 네트워크 환경에서도 사용자 체감 품질 저하를 야기하는 재생 끊김 현상, 재생 중 비디오 품질의 급격한 변화 등을 효율적으로 방지하는 비트율 적응 알고리즘인 QoE-enhanced adaptation algorithm over DASH (QAAD)를 제안한다. 또한 DASH 테스트베드 상에 QAAD와 재생 버퍼를 고려한 기존의 알고리즘 (즉, QDASH[9])을 구현하여 실험적으로 성능을 비교 및 분석하였다. 네트워크 환경이 급변하는 환경에서 실험을 수행한 결과, QDASH는 재생 끊김 현상 및 비디오 화질의 잦은 변화가 발생하는 반면, QAAD는 재생 끊김 없이 안정적인 비디오 화질을 제공하는 것을 확인할 수 있었다.

• 한국멀티미디어학회논문지
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• 제3권4호
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• pp.399-406
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• 2000

다자간 화상회의에서 발생하는 네트워크 혼잡상태에 동적으로 적응하기 위한 연구로는 송신자기반기법과 수신자기반 기법이 있는데, 전자는 수신자들의 평균 손실률에 따른 전송률로 가용 대역폭이 높은 수신자의 대역폭 낭비가 발생하며, 후자는 수신자들의 가용 대역폭에 따른 레이어(layer)를 동적으로 나누는 방법이 부족하다. 이를 위해 본 논문에서는 수신자의 네트워크 상태에 따라 정상상태와 혼잡상태로 구분하는 두 개의 멀티캐스트 그룹을 형성하고, 수신자가 손실률에 따라 동적으로 그룹을 선택함으로써 대역폭에 적합한 품질의 서비스를 받을 수 있도록 하는 대역폭 관리 알고리즘을 설계한 후 이를 구현하였다. 실험을 통한 결과를 살펴볼 때 수신자는 손실률에 따라 적합한 그룹을 선택함으로써 대역폭 낭비가 발생하는 문제점을 개선할 수 있었다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(1)
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• pp.17-20
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• 2000

TCP is ill-suited to real-time multimedia applications. Its bursty transmission, and abrupt and frequent wide rate fluctuations cause high delay jitters and sudden quality degradation of multimedia applications. Deploying non congestion controlled traffic results in extreme unfairness towards competing TCP traffic. Therefore, they need to be enhanced with congestion control schemes that not only am at reducing loss ratios and improve bandwidth utilization but also are fair towards competing TCP connections. This paper proposes a differentiated rate adaptation algorithm based on loss and round trip time. Rate in a sender quickly responds to loss ratio and holds steady state. Additionally, this algorithm reduces loss ratio by loss prediction in a receiver.

• 한국정밀공학회:학술대회논문집
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• 한국정밀공학회 2003년도 춘계학술대회 논문집
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• pp.438-442
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• 2003

For an efficient sliding mode control system stability and chattering avoidance should be guaranteed. A continuation method using boundary layer is well known as one solution for this. However since not only model uncertainties and disturbances but also control task itself is variable. it is practically impossible to set controller parameters - control discontinuity, control bandwidth, boundary layer thickness - in advance. In this paper first an adaptation law of control discontinuity is introduced to assure system stability and then fuzzy logic based tuning algorithm of design parameters is applied based on monitored performance indices of tracking error, control chattering, and model precision. In the end maximum control bandwidth not exciting unmodeled dynamics and minimum control discontinuity, boundary layer thickness making system stable and free of chattering are found respectively. This eliminates control chattering and enhances control accuracy as much as possible under given control situation. In order to demonstrate the validity of the proposed algorithm safe headway maintenance control for autonomous transportation system is simulated. The control results show that the proposed algorithm guarantees system stability all the time and tunes control parameters consistently and in consequence implements an efficient control in terms of both accuracy and actuator chattering.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제3권6호
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• pp.628-646
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• 2009

Users want to browse and search various web contents with mobile devices which can be used anywhere and anytime without limitations, in the same manner as desktop. But mobile devices have limited resources compared to desktop in terms of computing performance, network bandwidth, screen size for full browsing, and etc, so there are many difficulties in providing support for mobile devices to fully use desktop-based web contents. Recently, mobile network bandwidth has been greatly improved, however, since mobile devices cannot provide the same environment as desktop, users still feel inconvenienced. To provide web contents optimized for each user device, there have been studies about analyzing code to extract blocks for adaptation to a mobile environment. But since web contents are divided into several items such as menu, login, news, shopping, etc, if the block dividing basis is limited only to code or segment size, it will be difficult for users to recognize and find the items they need. Also it is necessary to resolve interface issues, which are the biggest inconvenience for users browsing in a mobile environment. In this paper, we suggest a personalized adaptation system that extracts item blocks from desktop-based web contents based on user interests, layers them, and adapts them for users so they can see preferred contents first.

• 정보처리학회논문지C
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• 제13C권7호
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• pp.933-938
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• 2006

최근 들어 폭발적으로 증가하고 있는 다양한 형태의 멀티미디어 서비스들은 서로 다른 대역폭과, 다양한 QoS의 보장을 요구한다. 따라서, 한정된 대역폭을 가지는 네트워크의 특성으로 인해 효율적인 대역폭 관리에 대한 관심이 증가하고 있다. 본 논문에서는 네트워크상에서 서로 다른 멀티미디어 서비스를 효율적으로 지원하면서 동시에 우선순위가 높은 서비스의 QoS를 보장할 수 있는 적응적 대역폭 관리 방법을 제안하였다. 네트워크 자원의 효율적인 관리를 위해 제안된 적응적 대역폭 관리기법은 현재 네트워크 상태를 기반으로 하는 온라인 기법으로 적용되어 서로 상충되는 네트워크의 성능 메트릭간의 적절한 균형을 이루도록 설계되었다. 컴퓨터 시뮬레이션을 통하여 대역폭 관리를 위해 제안된 기존의 타 기법들과의 비교, 분석을 수행하여 제안된 방법이 다양한 네트워크 트래픽 상황에서 우수한 성능을 가지는 것을 확인할 수 있었다.

• 한국통신학회논문지
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• 제34권5B호
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• pp.517-525
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• 2009

인터넷을 통하여 다른 서비스에 비해 상대적으로 많은 대역폭을 필요로 하는 비디오 스트리밍 서비스를 제공할 경우, 다른 플로우들과의 형평성을 제공하면서 비디오 서비스의 품질을 보장해야한다. 이를 위하여 본 논문에서는 스케일러블 비디오 압축 스트림에 대한 멀티채널 TCP-friendly 스트리밍 기법을 제안하였다. 제안 구조는 스케일러블 비디오 코딩으로 압축된 비디오 정보를 기본계층(baseline layer)은 TCP를 이용하여 손실 없이 전송하고 상위계층(enhancement layer)는 TFRC (TCP friendly rate control)을 이용하여 전송하도록 하였다. 이때, TFRC로 전송되는 상위계층 레이어에 대한 전송률은 수신측 재생버퍼의 상태와 전송 미디어의 계층별 부호화율을 고려하여 적응적으로 결정되도록 제안하였다. 시뮬레이션 결과를 통하여, 이러한 적응적 전송률 제어를 통하여 제안 스트리밍 방식이 기존 방식에 비해 네트워크 상황변화에 신속히 적응하고 혼잡상황에서도 일정수준의 서비스 품질을 보장함을 확인 할 수 있다.

• Journal of Communications and Networks
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• 제14권3호
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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.