• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.996-999
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• 2014

Early Smart TV market had their interest in the ability of HW, but these days Smart TV market begin to concentrate on it's services, contents and applications of SW. However, because in Smart TV application, there is no major difference with the application in smart phones and tablets. So the utilization rate in Smart TV application is evidently low. That's the reason why many people did not feel the need to use smart TV applications. In order to solve this problem, this paper propose to develope an application with the cloud computing technique [clould computing] that is intended to provide an usability through the interlocking of the individual PC and smart phone. After uploading user's photos from smart phone or PC to smart TV, this application will help not only to create user's own video but also to add various kind of video effects that user want. Therefore, in this paper, I expect the activation of smart TV applications.

• Journal of Internet Computing and Services
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• v.17 no.1
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• pp.55-64
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• 2016

Existing Android applications for streaming video in real time are dependent on the codec, which composes the encoding function, and the version of Android operating system. Also, for streaming video in real time, most applications should be connected with a separate desktop PC. To overcome these disadvantages, we propose a new application, which records and streams video in real time. Specifically, the proposed application uses the flash video file format, which is the common media file format supported by various versions of Android operating system. Through experiments, we show that it is possible for the proposed application to record the video screens more than 20 frames per second and to stream it in real time while using the existing video encoding methods.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.2
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• pp.181-185
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• 2008

Layered video multicast such as RLM Receiver-driven layered multicast) is a premising technique for delivering streaming video to a set of heterogeneous receivers over ALM(Application Layer Multicast) as well as over IP multicast. However, this approach may suffer from unnecessary fluctuation of video quality due to overlapped and failed join-experiments. Though a shared teaming scheme was introduced to resolve these problems, it may cause high control overhead and slow convergence problem when used with ALM. In this paper, we propose a new shared learning scheme for ALM-based layered video multicast which reduces control overhead and convergence latency while keeping the number of fluctuation reasonably small. The simulation results show that the proposed scheme performs better than an ALM-based layered video multicast with shared learning in terms of control overhead and convergence latency.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.192-194
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• 2005

Acceptance of the international standards for video compression, such as H.261, MPEG-1 and MPEG-2, along with the developments in video codec hardware, has created an explosion of application. Among these, the long time quest for long-distance digital video transmission causes an increasing interest in transporting compressed video over networks which are nontraditional for this purpose, including asynchronous transfer mode networks, the Internet, and cellular and wireless channels. Transmission of compression video over packet network is improved for error resilience. And layered video coding techniques improves error resilience. We present a efficient method of scalable video coding for low bandwidth.

• 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.

• ETRI Journal
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• v.22 no.1
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• pp.20-29
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• 2000

In this paper, we present a design of video and audio single chip encoder/decoder for portable multimedia application. The single-chip called as video audio signal processor (VASP) consists of a video signal processing block and an audio single processing block. This chip has mixed hardware/software architecture to combine performance and flexibility. We designed the chip by partitioning between video and audio block. The video signal processing block was designed to implement hardware solution of pixel input/output, full pixel motion estimation, half pixel motion estimation, discrete cosine transform, quantization, run length coding, host interface, and 16 bits RISC type internal controller. The audio signal processing block is implemented with software solution using a 16 bits fixed point DSP. This chip contains 142,300 gates, 22 Kbits FIFO, 107 kbits SRAM, and 556 kbits ROM, and the chip size is $9.02mm{\times}9.06mm$ which is fabricated using 0.5 micron 3-layer metal CMOS technology.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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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.

• Proceedings of the KSRS Conference
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• 2003.11a
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• pp.396-398
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• 2003

Recently as users' requests for geographic information systems are being more various, several new functions are being added to existing geographic information systems. One of them is linkage of spatial geographic data and video / image data to offer more realistic information of geographic objects to users. Geographic information systems implementing this function are called video geographic information system. In this paper, we design and implement a video geographic information system for providing map data, video data and link information of them to web applications. We 1) design system architecture of a video geographic information system, 2) analyze some processes to construct link information of map data and video data, 3) design database schema to store map data, video data, and link information, and 4) design some XML/GML schema used to query and retrieve these data for web applications.

• ETRI Journal
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• v.27 no.5
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• pp.491-496
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• 2005

An application specific processor for an H.264 decoder with a configurable embedded processor is designed in this research. The motion compensation, inverse integer transform, inverse quantization, and entropy decoding algorithm of H.264 decoder software are optimized. We improved the performance of the processor with instruction-level hardware optimization, which is tailored to configurable embedded processor architecture. The optimized instructions for video processing can be used in other video compression standards such as MPEG 1, 2, and 4. A significant performance improvement is achieved with high flexibility. Experimental results show that we could achieve 300% performance for the H.264 baseline profile level 2 decoder.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.149-152
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• 2003

In this paper, we propose a scheme that applies to the VoIP application based on H.323 protocol to enhance the bandwidth efficiency. We multiplex the audio and video stream. In this scheme, audio frame is carried with video stream. And we applies not only multiplexing but also (in header compressing to the real audio/video stream to increase the bandwidth efficiency. With the multiplexing and RTP header compressing, we gain the bandwidth efficiency. In the finite network environment, We can assign bandwidth to other users who want to use other service. and other VoIP users. If we can apply the real time network situation to the our VoIP application, we can get more efficient performance.