• Journal of KIISE:Software and Applications
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• v.29 no.7
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• pp.436-449
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• 2002

This paper describes on our experience of developing the Database Connector as an interconnection method between multimedia database, and the streaming framework. It is possible to support diverse and mature multimedia database services such as retrieval and join operation during the streaming if an interconnection method is provided in between streaming system and multimedia databases. The currently available interconnection schemes, however have mainly used the file systems or the relational databases that are Implemented with separated form of meta data, which deafs with information of multimedia contents, and streaming data which deals with multimedia data itself. Consequently, existing interconnection mechanisms could not come up with many virtues of multimedia database services during the streaming operation. In order to resolve these drawbacks, we propose a novel scheme for an interconnection between streaming framework and multimedia database, called the Inter-Process Communication (IPC) based Database connector, under the assumption that two systems are located in a same host. We define four transaction primitives; Read, Write, Find, Play, as well as define the interface for transactions that are implemented based on the plug-in, which in consequence can extend to other multimedia databases that will come for some later years. Our simulation study show that performance of the proposed IPC based interconnection scheme is not much far behind compared with that of file systems.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10B
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• pp.857-865
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• 2004

Recently the interest in digital home services and its related technology has increased rapidly. Multimedia data transmission between digital devices that are connected to home network is essential function. This paper presents the design and implementation of multimedia streaming system based on IEEE1394 in Linux environment. The key features of proposed system are as follows. First, it can transmit and receive not only DV format data but also MPEG2-TS format data through IEEE1394 bus. Second, it uses device drivers that handling IEEE1394 and IEC61883 protocol transparently to reduce the complexity of communication program. Third, it applied capacity adaptation technique considering various specifications of PCs to get smooth streaming for HD data. These characteristics are tested on testbed of IEEE1394 based home network. The proposed media streaming technique can be used as a home media sever in Linux environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.711-723
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• 2002

A finer granular scalable (FGS) version of ISO/IEC MPEG-4 video streaming is investigated in this work with the prioritized stream delivery over loss-rate differentiated networks. Our proposed system is focused on the seamless integration of rate adaptation, prioritized packetization, and simplified differentiation for the MPEG-4 FGS video streaming. The proposed system consists of three key components: 1) rate adaptation with scalable source encoding, 2) content-aware prioritized packetization, and 3) loss-based differential forwarding. More specifically, a constant-quality rate adaptation is first achieved by optimally truncating the over-coded FGS stream based on the embedding rate-distortion (R-D) information (obtained from a piecewise linear R-D model). The rate-controlled video stream is then packetized and prioritized according to the loss impact of each packet. Prioritized packets are transmitted over the underlying network, where packets are subject to differentiated dropping and forwarding. By focusing on the end-to-end quality, we establish an effective working conditions for the proposed video streaming and the superior performance is verified by simulated MPEG-4 FGS video streaming.

• Journal of KIISE:Information Networking
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• v.36 no.3
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• pp.191-197
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• 2009

One of popular applications in the Internet is the multimedia streaming services such as IPTV or You Tube in which supporting quality of service (QoS) is an important issue. One of widely adopted rate control scheme is TCP-friendly rate control (TFRC) which shows comparable performance with TCP in term of throughput and lower variation of throughput over time. On the other hand, devices with multiple interfaces are emerging in the market. However, it has not proposed to exploit multiple interfaces simultaneously for multimedia streaming services with TFRC. In this paper, we propose a multimedia streaming algorithm with TFRC which exploits multiple interfaces to guarantee the quality of service. We show that the proposed scheme shows better performance than those with a single interface in terms of throughput and communication costs.

• Journal of KIISE
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• v.44 no.5
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• pp.545-552
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• 2017

Because of the explosive growth of mobile devices and development of network technologies, HTTP (Hypertext Transfer Protocol) adaptive streaming has become a new trend in video delivery to provide efficient multimedia streaming services. As interest in personalized broadcasting grows, the study of interactive multimedia has been actively pursued. The interactive multimedia service is a method of playing media according to a scenario selected by the user. Providing the interactive multimedia service with the existing HTTP adaptive streaming causes switching delay and buffer underflow according to the point in time when the user selects the scenario while the client streams the interactive multimedia and therefore decreases the user QoE (Quality of Experience). In this paper, we propose an adaptive streaming scheme for interactive multimedia service in HTTP adaptive streaming to provide seamless playback. We design the architecture and prefetching scheme for interactive multimedia streaming.

• Journal of the Korea Convergence Society
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• v.12 no.12
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• pp.197-211
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• 2021

The emergence of knowledge-sharing live streamers provides more diversified content to the live streaming platform. Analysis of the factors affecting the intention to use knowledge sharing live streaming users can allow the live streaming platform to understand better the adoption characteristics of users who follow this type of content. Help platform operators provide better services and help live streaming platforms innovate. Based on the TAM model, this research uses questionnaire surveys and structural equation models to construct a conceptual model of the influencing factors of users' intentions in the knowledge sharing live streaming and conduct an empirical analysis on the influencing factor models. The results of data analysis show that a significant influence of users' attitudes of knowledge sharing live streaming is perceived usefulness, followed by flow experience; perceived value has a positive impact on users' attitudes and intention to use, and the positive influence of users attitude significantly affect the user's intention.

• Journal of Broadcast Engineering
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• v.25 no.4
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• pp.560-568
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• 2020

In recent years, as network technology develops, the usage of streaming services by users is increasing. However, the complexity of streaming services is also increasing due to various terminal environments. Even when streaming the same content, it is necessary to re-encode the content according to the type of service. In order to solve the complexity and latency of the streaming service, Moving Picture Experts Group (MPEG) has standardized the Common Media Application Format (CMAF). In addition, as content transmission using a communication network becomes possible, the Republic of Korea's Ultra High Definition (UHD) broadcasting standard has been enacted as a hybrid standard using a broadcasting network and a communication network. The hybrid service enables various services such as transmitting additional information of contents or providing user-customized contents through a communication network. The Republic of Korea's UHD transmission standard utilizes MPEG Media Transport (MMT), and Mobile MMT is an extension of MMT to provide mobile network-specific functions. This paper proposes a method of inserting CMAF contents suitable for various streaming services using signaling messages of MMT and Mobile MMT. In addition, this paper proposes a model for content insertion system in heterogeneous network environment using broadcasting and communication networks, and verifies the validity of the proposed technology by checking the result of content insertion.

• Journal of Korea Society of Industrial Information Systems
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• v.6 no.2
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• pp.47-53
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• 2001

Real-time multimedia transmit services like internet broadcasting, web-based education, chatting, conference, circumference observation etc. by using internet are given a great deal of weight on most recently internet services. In these way, various internet services are presented and there applications are also gradually spreaded because of improvement MPEG and streaming processing techniques. But currently internet service is a development of system which has independent function of above services. And a research and development of system which have a multipurpose internet service are insufficiency. In this paper, we design and develop system(RealCam SUN A/V) for transmitting images and voices based on MPEG4 streaming. As a result of our research, it made possible to transmit images with 15 frame/sec in 160×120 frame size at 56Kbps and 30 frame/sec in 640×480 frame size at 512Kbps. And also we can apply this system as multipurpose function to circumference observation and internet broadcasting etc., by transmitting voices, images and control signal simultaneously.

• Journal of Information Technology Services
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• v.14 no.1
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• pp.99-113
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• 2015

Adaptive bitrate (ABR) streaming technology has become an important and prevalent feature in many multimedia delivery systems, with content providers such as Netflix and Amazon using ABR streaming to increase bandwidth efficiency and provide the maximum user experience when channel conditions are not ideal. Where such systems could see improvement is in the delivery of live video with a closed loop cognitive control of video encoding. In this paper, we present streaming camera system which provides spatially and temporally adaptive video streams, learning the user's preferences in order to make intelligent scaling decisions. The system employs a hardware based H.264/AVC encoder for video compression. The encoding parameters can be configured by the user or by the cognitive system on behalf of the user when the bandwidth changes. A cognitive video client developed in this study learns the user's preferences (i.e. video size over frame rate) over time and intelligently adapts encoding parameters when the channel conditions change. It has been demonstrated that the cognitive decision system developed has the ability to control video bandwidth by altering the spatial and temporal resolution, as well as the ability to make scaling decisions