• The Journal of the Korea Contents Association
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• v.7 no.12
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• pp.303-310
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• 2007

Compressed video bitstreams are intended for real-time transmission over communication networks. Because video compression algorithms eliminate the temporal, spatial, and statistical redundancies, the coded video bitstreams are very sensitive to transmission errors. We propose an error resilient video coding technique to limit the effect of error propagation in low bit-rate video coding. The success of error resilient coding techniques relies on how accurately the transmission errors can be detected. To detect the transmission error, we propose a very simple error detection technique based on data hiding Next, we conceal the corrupted MB data using intra MB refresh and motion compensation with the estimated motion vector and compare the simulation results. This method will be useful in video communication in error Prone environment such as WCDMA networks.

• Proceedings of the IEEK Conference
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• 2002.06c
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• pp.47-50
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• 2002

In this paper, the efficient transmission technique of real time video data over 3G wireless networks is presented. To understand the transmission characteristic of 3G networks on real time video data, we implemented the data transport structure of 3G wireless networks. Also this research is based on current 3G wireless network specification of 3GPP, 3GPP2 standard organization to evaluate the result over real 3G wireless network environment. The retransmission by radio link layer results in the delay factor. To implement video data transmission efficiently, we Propose to use both no-retransmission and forward error correction method.

• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.3
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• pp.1-5
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• 1983

The effects of modal noise of analog video transmission systems using semiconductor laser diode is investigated. The system linearity degradation due to modal noise is examined for various fiber types. It was concluded that in alalog video transmission systems using multimode fiber, modal noise is so serious that reduction of coherency is essential to the development and that single mode libers are adequate for high quality analog video transmission systems.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.66-69
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• 2000

In this paper, we propose a progressive transmission of a video using a 2$\times$2$\times$2 DCT First of all, the video data is transformed into multiresolution represented video data using a 2$\times$2$\times$2 DCT. Then. it is represented by a 3-D EZT(Embedded Zero Tree) coding fur the progressive transmission with a bit-rate scalability. The proposed progressive transmission algorithm needs much less computations and buffer memories than the higher-order convolution based wavelet filter. Also, since the 2$\times$2$\times$2 DCT requires independent local computations, parallel processing can be applied.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.169-172
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• 2001

In this paper, we describe a method for real-time transmitting video streams based on RTP In order to guarantee synchronous video streams, we propose a method to grasp network situations by analyzing end-to-end network traffic. In addition, we present an algorithm for satisfying QoS requirements of real-time multimedia transmission and maintaining continuity of transmission. It describes a buffering method for overcoming bandwidth limitations and an analyzing method based on RTCP for grasping network traffic situation to resolve the problem of real-time transmission of video streams.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• 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.

• Journal of Digital Contents Society
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• v.13 no.1
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• pp.111-118
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• 2012

The burst is to require abruptly high transmission rate in case of transmitting pre-stored variable bit rate video data, and it causes to be inefficient use of network resource, resource reservation. To avoid these problems, smoothing is transmission plan where variable rate video data is converted to a constant bit rate stream. These smoothing algorithms include CBA, MCBA, MVBA and others. To evaluate amount of burst reduction in the existing CBA, MCBA, MVBA algorithm, this paper compares the burst-related-factors of transmission plan in smoothing algorithms with original video sources which were stored Variable Bit Rate. There are maximum frame bytes, maximum GOP bytes, transmission rate variability per frame, transmission rate variability per GOP in burst-related evaluation factors. Experimental result shows burst-related factors of smoothing algorithms which were used for experiment lower than that of pre-stored video data, except special case.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.8
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• pp.510-516
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• 2005

Rapid advances in telecommunication make emergency telemedicine possible that specialist offers medical care to an emergency case in moving vehicle. Although there were many telemedicine projects delivering the image or video of patient over several wireless networks, none of them considered effective solutions for optimizing video transmission over error-prone environments, such like wireless links. To alleviate the effect of channel errors on compressed video bit-stream, this paper analyzed the error resilient features of MPEG-4 standard and measured the quality of transmitted MPEG-4 encoded video over commercially available CDMA2000 1xEV-DO networks, transmitting different IP packet sizes and RM positions. we propose an error resilient transmission methods for emergency telemedicine over real 3G network.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2007.11a
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• pp.325-329
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• 2007

The mobile and multimedia service on wireless network have been leaded from the improvement of telecommunication techniques. A typical multimedia service, a video transmission usually requires larger bandwidth than voice transmission. Many channel allocation algorithms for Orthogonal Frequency Division Multiple Access (OFBMA) to use resources more efficiently, Previous channel allocation algorithms have developed with an assumption that the data traffic is constant bit rate (CBR). However, existing algorithms are not suitable to video traffic because it usually generates a variable bit rate (VBR) traffic. In this paper, we proposed a new channel allocation algorithm called a queue-based channel allocation. it is more suitable to transmit the video traffics. Also, a problem are notified in case of realtime generated video traffic and a corresponding heuristic solution was proposed.