• Journal of Korea Multimedia Society
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• v.9 no.1
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• pp.59-72
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• 2006

The rapid growth of technology for network and terminal devices is able to service multimedia contents in mobile environments. But as the environment of networks and the performance of devices is varied, the same quality service is very difficult to be provided for user in internet and mobile environment. Therefore, multimedia streaming system must provide an adaptive service in considering network and device environment. Video transcoding techniques are the good solution that can provide video streaming service in various network environments adaptively. This paper proposes a dynamic frame-skipping transcoding scheme to provide adaptive streaming service in various network environments. This scheme uses the prediction period for transcoding. A proposed transcoding scheme can reduce time complexity through combination of prediction period and skipping period to control frame-skipping rate in encoder. In simulation results, the performance of proposed scheme is similar to the performance of a traditional dynamic frame-skipping scheme in PSNR value. But the performance of proposed scheme outperforms the performance of traditional scheme in time complexity.

• Journal of Convergence Society for SMB
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• v.6 no.3
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• pp.49-55
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• 2016

This paper proposes an adaptive video signal transmission scheme in order to ensure the QoS (Quality of Service) of user requirements. SVC (Scalable Video Coding) is an effective transmission scheme, because that can transmit video signal according to the video layer's weight. However, in previous works, those adaptive transmission systems which are considered about the various channel environments and user requirements have not been insufficiently studied. So, we propose the SVC signal transmission using inter-layer differential OVSF code allocation scheme in MC-CDMA. The proposed scheme is able to obtain each layer signal's protection order and control the sub-block of MC-CDMA by feedback information from receiver. Therefore, our proposed scheme is possible to provide the video quality for each users according to variable channel environments. The simulation results demonstrate the enhancement of proposed system in terms of BER performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4B
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• pp.202-209
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• 2005

In this paper, a dynamic TCP-friendly rate control (TFRC) is proposed to adjust the coding rates according to the channel characteristics of the wireless-to-wired network consisting of wireless first-hop channel. To avoid the throughput degradation of multimedia flows traveling through wireless lint the proposed rate control system employs a new wireless loss differentiation algorithm (LDA) using packet loss statistics. This method can produce the TCP-friendly rates while sharing the backbone bandwidth with TCP flows over the wireless-to-wired network. Experimental results show that the proposed rate control system can eliminate the effect of wireless losses in flow control of TFRC and substantially reduce the abrupt quality degradation of the video streaming caused by the unreliable wireless link status.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• Journal of KIISE:Information Networking
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• v.35 no.3
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• pp.173-182
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• 2008

Promoting quality of streaming service in wireless networks has attracted intensive research over the years. Instable wireless channel condition causes high transmission delay and packet loss, due to fading and interference. Therefore, they lead to degrade quality of video streaming service. The IEEE 802.11 Working Group is currently working on a new standard called IEEE 802.11e to support quality of service in WLANs. And several schemes were proposed in order to guarantee QoS. However, they are not adaptable to network condition. Accordingly, they suffered video quality degradation, due to buffer overflow or packet loss. In this paper, to promote quality of video streaming service in WLANs, we propose a cross-layer architecture based on IEEE 802.11e EDCA model. Our cross-layer architecture provides differentiated transmission mechanism of IEEE 802.11e EDCA based on priority of MPEG-4 video frames and adaptively controls the transmission rate by dropping video frames through the efficient bandwidth estimation based on distinction of each AC. Through the simulation, proposed scheme is shown to be able to improve end-to-end qualify for video streaming service in WLANs.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.10
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• pp.722-733
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• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• Journal of Broadcast Engineering
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• v.27 no.6
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• pp.914-922
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• 2022

In this paper, we construct a commercial 4K UHD(Ultra High Definition) streaming device that utilizes a 5G mobile communication network as a transport channel and conduct a streaming performance test. It uses RTP(Realtime Transport Protocol) which has transmission quality monitoring capability as a transmission protocol to apply adaptive streaming. In addition, it provides the function to adjust the encoding rate of the video signal so that encoding can be optimized for the change in the bandwidth of the transmission channel. Through the performance test, it is confirmed that the H.265 encoding rate for 4K UHD signal is 48.69Mbps, the average glass-to-glass delay time is 293.60ms, and the average time difference between video and audio for lip sync is 120ms. With the result of performance test, it is shown that the streaming device is applied to 4K UHD Streaming device over 5G mobile communication network.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.20-29
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• 2014

Recently, LTE is preparing to make a new leap forward LTE-A all over the world. As LTE privides high speed service, the role of mobile phones seems to change from voice to data service. According to Cisco, global mobile data traffic will increase nearly 11-fold between 2013 and 2018. Mobile video traffic will reach 75% by 2018 from 66% in 2013 in Korea. However, voice service is still the most important role of mobile phones. Thus, controllability of throughput and low BLER is indispensable for high-quality VoIP service among various type of traffic. Although the maximum AMR-WB, 23.85 Kbps is sufficient to a VoIP call, it is difficult for the LTE which can provide tens to hundreds of MB/s may not keep the certain level VoIP QoS especially in the cell-edge area. This paper proposes a new scheduling algorithm in order to improve VoIP performance after analyzing various scheduling algorithms. The proposal is the technology which applies more priority processing for VoIP than other applications in cell-edge area based on two-tier scheduling algorithm. The simulation result shows the improvement of VoIP performance in the view point of throughput and BLER.

• The KIPS Transactions:PartB
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• v.15B no.6
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• pp.575-584
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• 2008

According to the network condition, the communication network overload could be occurred when media transmitting. Many researches are being carried out to lessen the network overload, such as the filtering, load distributing, frame dropping and many other methods. Among these methods, one of effective method is frame dropping that reduces specified video frames for bandwidth diminution. B frames are dropped and then I, P frames are dropped according to dependency among the frames in frame dropping. This paper proposes a scheme for protecting copyrights by encryption, when we apply frame dropping to reduce bandwidth of media following MPEG-4 file format. We designed two kinds of frame dropping: first one stores and then sends the dropped files and the other drops frames in real-time when transmitting. We designed three kinds of encryption methods in which DES algorithm is used to encrypt MPEG-4 data: macro block encryption in I-VOP, macro block and motion vector encryption in P-VOP, and macro block and motion vector encryption in I, P-VOP. Based on these three methods, we implemented a digital right management solution for MPEG-4 data streaming. We compared the results of dropping, encryption, decryption and quality of video sequences to select an optimal method, and there is no noticeable difference between the video sequences recovered after frame dropping and the ones recovered without frame dropping. The best performance in encryption and decryption of frames was obtained when we apply the macro block and motion vector encryption in I, P-VOP.