• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.4
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• pp.23-33
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• 2009

A shot change detection technique is an important technique for effective management of video data, thus it requires an adaptive algorithm for various video sequences to detect an accurate shot change frames. In this paper, we propose an adaptive shot change detection algorithm using histogram mean of frames within extension sliding window. Our algorithm calculates a frame feature value using histogram and defines an adaptive threshold using an average of histogram mean of frames within the extension sliding window and determines a shot change by comparing the feature value and the threshold. We obtained better detection rate than the conventional methods maximally by 15% in the experiment with the same test sequence. We verified real-time operation of shot change detection in the hardware platform with low performance by implementing it on TVUS HM-900 PLUS model of Homecast. The Proposed algorithm can be useful in the hardware platform such as portable multimedia player(PMP) or cellular phone with low CPU performance.

• Journal of Internet of Things and Convergence
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• v.8 no.1
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• pp.53-58
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• 2022

Smoothing is a transmission plan that converts video data stored at a variable bit rate into a constant bit rate. In the study of [6-7], when a data rate increase is required, the frame with the smallest increase is set as the start frame of the next transmission rate section, when a data tate decrease is required. the frame with the largest decrease is set as the start frame of the next transmission rate section, And the smoothing algorithm was proposed and performance was evaluated in an environment where network traffic is not considered. In this paper, the smoothing algorithm of [6-7] evaluates the adaptive CBA algorithm and performance with minimum frame rate, average frame rate, and frame rate variation from 512KB to 32MB with E.T 90 video data in an environment that considers network traffic. As a result of comparison, the smoothing algorithm of [6-7] showed superiority in the comparison of the minimum refresh rate.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1618-1626
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• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• Journal of KIISE:Information Networking
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• v.37 no.6
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• pp.483-488
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• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.6
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• pp.341-349
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• 2014

Dynamic adaptive streaming over HTTP (DASH) is the most recent and promising technology to support high quality streaming services. In dynamic adaptive streaming over HTTP (DASH), a client consecutively estimates the available network bandwidth and decides the transmission rate for the forthcoming video chunks to be downloaded. In this paper, we propose a novel rate adaptation algorithm called quality of experience QoE-enhanced adaptation algorithm over DASH (QAAD), which preserves the minimum buffer length to avoid interruption and minimizes the video quality changes during the playback. We implemented a DASH test bed and conducted extensive experiments. Experimental results demonstrate that under fluctuating network conditions, QAAD provides seamless streaming with stabilized video quality while the previous buffer-aware algorithm (i.e., QDASH[9]) frequently changes the video quality and undergoes the interruption.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.3
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• pp.476-481
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• 2012

Recently, QoS/QoE technology as a part of the HTTP-based adaptive streaming technology has attracted attention. In this paper, we find out HTTP-based adaptive and dynamic streaming technology. It is based on a dynamic adaptive streaming system over HTTP, which is designed and implemented. The system consist of conversion by the bit rate of MPEG2-TS files, division of segment, creation of MPD(Media Presentation Description) and bitrate's player through analysis of MPD File between servers and clients in network environments of dynamic and adaptive. This diverse network environments, continuous and smooth playback of video will be used in various multimedia fields.

• Proceedings of the Korean Information Science Society Conference
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• 2011.06a
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• pp.385-387
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• 2011

본 논문에서는 이기종 무선망에서 제공되는 비디오 스트리밍 서비스의 품질을 보장하기 위해, 핸드오버 상황에 따라 변화하는 네트워크 상태에 적응적인 스트리밍 품질 적응 기법을 제안한다. 제안하는 기법은 패킷 페어 방식을 이용하여 네트워크의 가용대역폭을 측정하고 핸드오버 상황 공지 메시지를 이용하여 핸드오버 상황을 빠르게 인지함으로써 핸드오버로 인한 네트워크 가용대역폭 변화에 능동적으로 대응하며 네트워크 상태에 적응적으로 비디오 스트림의 품질을 조절한다. 실험을 통해 제안하는 기법이 핸드오버 상황에서 비디오 패킷 손실을 감소시켜 비디오의 재생 품질을 보장할 수 있음을 확인하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2011.06d
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• pp.234-237
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• 2011

최근 인터넷을 통한 멀티미디어 스트리밍 서비스 이용이 증가함에 따라 스트리밍 서비스 품질을 향상 시키기 위한 연구들이 진행되고 있다. 하지만 기존 연구들은 네트워크 상태에 적응적으로 전송률을 조절함으로써 네트워크의 효율성 및 안전성을 개선하는데 초점을 맞추고 있어서 사용자 측면의 체감품질에 대한 고려가 부족하다. 본 논문에서는 멀티미디어 스트리밍 서비스의 사용자 체감품질 (QoE)을 향상시키기 위한 적응적인 전송률 및 비디오 품질 조절 기법을 제안한다. 제안하는 기법은 네트워크의 가용대역폭과 클라이언트의 버퍼 상태를 기반으로 전송률 및 비디오 품질을 조절함으로써 멀티미디어 재생의 불연속성과 잦은 비디오 품질 변화에 의한 사용자 체감품질을 개선하였다. 시험 결과를 통해서 제안하는 기법이 부드럽게 (Smoothly) 비디오 품질을 조절하는 것을 보였다.

• Journal of KIISE:Computing Practices and Letters
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• v.6 no.5
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• pp.505-512
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• 2000

In this paper, an adaptive multimedia synchronization scheme for VOD (Video On Demand) services in internet environments is proposed. This scheme considers the characteristics of MPEG (Moving Picture Expert Group) system stream. Consequently, the intra-synchronization is handled at the pack layer, and the inter-synchronization is handled at the packet layer. The proposed scheme can cope adaptively with variation of packet loss, jitter and client's playback capacity. If there are variations of the packet loss or client's playback capacity, the server will change the transmission rate by selective picture skip. The client can then adjust and control the playback time according to the variation of the network jitter. Our experimental results show that the proposed scheme can quickly adapt to the network condition, and can guarantee a better quality of service than the other existing schemes.

• Journal of Digital Contents Society
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• v.13 no.4
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• pp.567-574
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• 2012

In this paper, we propose a buffering scheme to improve the performance by adaptively adjusting the length of the buffering period according to the degree of data duplication among peers that belong to each specific playback period in P2P live streaming systems. To do so, we first assign all the peers into many groups depending on their playback time positions. We then determine the length of their buffering periods according to the number of peers belonging to each group. That is, we increase the length of the buffering periods when much buffered data are duplicated among peers within a group while we decrease it when few buffered data are duplicated. By simulation experiments, we show that our proposed buffering scheme can improve the performance significantly compared to the fixed length buffering scheme that have been employed in the existing mesh-based P2P systems.