• The Journal of the Korea Contents Association
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• v.10 no.9
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• pp.36-44
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• 2010

In this paper, we propose video quality control system(VQCS) which can control video service quality through the monitoring of end-to-end available bandwidth for video streaming service like IPTV in NGN convergence network. Various multimedia services such as video, voice and gaming service can be provided by IPTV, and these services require large amounts of bandwidth. At this time, video quality degradation like video jerkiness, block distortion and blurring is caused when network available bandwidth is insufficient. Available bandwidth monitoring method is need to stably control video streaming quality. So, we periodically calculate the amount of the packets in link and measure available bandwidth by using total length field in IP header at terminal. Scalability extractor in network selects suitable video streaming data rate based on available bandwidth and transports video streaming with adaptive data rate to prevent video quality deterioration.

• Journal of Korea Multimedia Society
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• v.13 no.3
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• pp.349-358
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• 2010

Adaptability is an important issue in video streaming over mobile environments, since the clients may request videos with great differences in their workload. In this paper, we propose the issues in limited bandwidth scheduling for efficient MPEG-4 video stream transmission over a mobile or wireless network. In the phase of admission control, the amount of bandwidth allocated to serve a video request is the mean bandwidth requirement of its requested video. The dynamic allocation of bandwidth in the phase of scheduling depends on the playback buffer levels of the clients with an objective to make it more adaptive to the playback situation of individual clients. In the proposed RTA scheduling protocol, more bandwidth may be allocated temporarily to the client whose buffer level is low. By employing the buffer level based scheduling policy, this protocol attempts to maximize the real-time performance of individual playback while minimizing the impact of transient overloading. Extensive simulation experiments have been performed to investigate the performance characteristics of the RTA protocol as comparing with BSBA protocol. This RTA protocol shows the better performance by transferring more frames than BSBA protocol.Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.111-117
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• 2006

Developing a home network and a ubiquitous infrastructure requires various communication techniques and devices with more advanced hardware. With this development, increasing realtime access to multimedia data results in rapid degradation of qualify for multimedia playback. This paper presents a traffic allocation technique based on MPP(Media Preference for Presentation) that can steadily maintain multimedia playback quality by adaptively allocating streaming traffic requested from clients with different playback performances. Media preference is defined in accordance with content popularity and playback performance of client devices. Through experiments when requested stream data exceeds processing ability of a midea server, the proposed allocation technique shows 10% quality improvement comparing to the system without applying the proposed allocation technique.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.5
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• pp.691-708
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• 2010

An important issue of supporting multi-users with diverse quality-of-service (QoS) requirements over wireless networks is how to optimize the systematic scheduling by intelligently utilizing the available network resource while, at the same time, to meet each communication service QoS requirement. In this work, we study the problem of a variety of communication services over multi-media heterogeneous cognitive OFDM system. We first divide the communication services into two parts. Multimedia applications such as broadband voice transmission and real-time video streaming are very delay-sensitive (DS) and need guaranteed throughput. On the other side, services like file transmission and email service are relatively delay tolerant (DT) so varying-rate transmission is acceptable. Then, we formulate the scheduling as a convex optimization problem, and propose low complexity distributed solutions by jointly considering channel assignment, bit allocation, and power allocation. Unlike prior works that do not care computational complexity. Furthermore, we propose the FAASA (Fairness Assured Adaptive Sub-carrier Allocation) algorithm for both DS and DT users, which is a dynamic sub-carrier allocation algorithm in order to maximize throughput while taking into account fairness. We provide extensive simulation results which demonstrate the effectiveness of our proposed schemes.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.50-53
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• 2002

There are increasing needs to deliver the multimedia streaming over heterogeneous networks. When considering network environments and equipment accessed by user, delivery of video streaming must be scalable. There are many kinds of scalable video coding: spatial, temporal, SNR, and hybrid. The SNR scalable and spatial resolution, but different SNR quality with respect to layers. The 1-layer SNR scalable encoder produces SNR scalable video streams with ease. But, there is drift problem. Modified 1-layer approach does not have this problem but coding inefficiency, and is not MPEG-compliant. The present MPEG-compliant 2-layer encoder comes out to reduce coding rate. But it still use only base layer to encode whole layer. In this paper, we propose adaptive MPEG-compliant 2-layer encoder. Using linear combination algorithm, encoder use 1 motion vector to encode the sequences efficiently. By dong this, we can achieve the coding efficiency of SNR scalable coding.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.4
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• pp.1768-1789
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• 2016

In contemporary video streaming systems based on HTTP protocol, video players at the client side are responsible for adjusting video quality to network conditions and user expectations. However, when multiple video clips are streamed simultaneously, an intricate application logic implemented in the video players overlays the TCP mechanism which is responsible for a balanced access to a shared network link. As a result, some video players may not obtain a fair share of network throughput and may be vulnerable to an unstable video bit-rate. Therefore, we propose to simplify the algorithms implemented in the video players, which are responsible for the adjustment of video quality and constrain their functionality only to sending feedback to a server about a state of the player buffer. The main logic of the system is shifted to the server, which is now responsible for bit-rate selection and prioritisation of the video streams transmitted to multiple clients. To verify our proposition, we performed several experiments in a laboratory environment which show that when the server cooperates with the clients, the video players experience fewer quality switches and the system achieves better fairness when allocating network throughput among the video players. However, this comes at the cost of worse utilisation of network bandwidth.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.227-238
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• 2000

This paper, at first, provides analysis on loss pattern of Internet based on real experiments of the current Internet. Then, we propose an effective adaptive QoS management technique, in which measured loss pattern as well as PLR (Packet Loss Ratio) are used to select titrate of temporal scalability. level of FEC and retransmission This selection is also incorporated to the MPEG-4 error resilience tools and error concealment techniques. In order to minimize effect of packet loss, multimedia stream is segmented in the unit of group of Pictures (GOP) and interleaving and FEC are applied to the segment. Proposed algorithms are applied to build a VOD system.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2120-2121
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• 2006

본 논문에서는 네트워크 상태, 사용자 단말 성능, 지원 가능한 비디오 포맷등 사용자 정보들을 서버와 공유하여 원활한 비디오스트리밍이 가능한 클라이언트를 기술한다. 이러한 사용자 정보는 MPEG-21 멀티미디어 프레임워크를 이용하여 서버와 통신하며, 기존의 VoD의 단방향 스트리밍이 아닌 양방향의 스트리밍 구조로 설계하였다. 양방향 스트리밍은 사용자 환경뿐만 아니라 이전에 재생된 비디오 시퀀스 및 사용자 이동성에 따른 적응형 재생이 가능하도록 스트리밍과 다른 통신채널을 이용하여 단말의 정보를 서버에게 보낼 수 있다. 실제 응용과 검증을 위해 본 논문에서는 PC를 이용한 단말 시스템과 PDA를 이용하여 동작시험 결과를 얻었다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.12
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• pp.5981-5997
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• 2017

Advanced Television Systems Committee (ATSC) produced ATSC 2.0 standard, which has been used for a HD/SD digital broadcasting service with MPEG-2 Transport Stream (TS) as a delivery protocol. Recently, users might want to consume multimedia services without being constrained by transport network. However, MPEG-2 TS is not suitable protocol for IP network. Thus, ATSC has enacted ATSC 3.0 standard, which is designed for a hybrid service through a conversion of both broadcasting and broadband channels. The ATSC 3.0 specifies delivery protocols as MPEG Media Transport (MMT) for a broadcasting environment and Dynamic Adaptive Streaming over HTTP (DASH) for a broadband. However, it seems difficult to combine broadcast and broadband for hybrid broadcasting, and it is difficult to synchronize media data between MMT and DASH. This paper suggests various new service scenarios to be brought up by a hybrid broadcasting, ATSC 3.0, and also proposes a stable mechanism for switching contents between different delivery protocols such as MMT and DASH.

• Proceedings of the Korean Information Science Society Conference
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• 2011.06d
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• pp.256-258
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• 2011

유무선 통신기술의 발달로 멀티미디어 스트리밍 서비스의 이용이 증가함에 따라 비디오 스트리밍 기법에 관한 연구가 필요하다. 본 논문에서는 끊김 없는 스트리밍 서비스를 제공하기 위하여 네트워크의 상태와 클라이언트의 버퍼 상태를 고려한 전송률 조절 기법을 제안한다. 제안하는 기법은 네트워크의 상태와 클라이언트의 버퍼 상태를 기반으로 전송 비디오의 프레임 수를 우선순위에 따라 차등적으로 폐기함으로써 전송률을 조절한다. 제안 기법을 적용한 스트리밍 시스템을 구현하고 실험을 통해 끊김 없는 비디오의 전송을 확인할 수 있었다.