• Journal of Broadcast Engineering
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• v.13 no.1
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• pp.119-127
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• 2008

Transmission of video over Internet or wireless network requires coded stream capable of adapting to dynamic network conditions instantly. To meet this requirement, various scalable video coding schemes have been developed, among which the Scalable Video Coding (SVC) extension of the H.264/AVC is the most recent one. In comparison with the scalable profiles of previous video coding standards, the SVC achieves significant improvement on coding efficiency performance. For adapting to dynamic network bandwidth, the SVC employs inter-layer switching between different temporal, spatial or/and fidelity layers, which is currently supported with instantaneous decoding refresh (IDR) access unit. However, for real-time adaptability, the SVC has to frequently employ the IDR picture, which dramatically decreases the coding efficiency. Therefore, an extension of SP picture from the AVC to the SVC for an efficient inter-layer switching is investigated and presented in this paper. Simulations regarding the adaptability to dynamic network bandwidth are implemented. Results of experiment show that the SP picture added SVC provides an average 1.2 dB PSNR enhancement over the current SVC while providing similar adaptive functionality.

• KIPS Transactions on Computer and Communication Systems
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• v.1 no.1
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• pp.29-34
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• 2012

Wireless multicast service can be used for video streaming service to save the network resources by sending the same popular multimedia contents to a group of users at once. For better multimedia streaming multicast service, we propose a quality-aware rate adaptation (QARA) scheme for scalable video multicast in rate adaptive wireless networks. In QARA, transmission rate is determined depending on the content's type and users' channel conditions. First, the base layer is transmitted by a low rate for high reliability. That means we provide basic service quality to all users. On the contrary, the transmission rate for enhancement layer is adapted by using channel condition feedback from a randomly selected node. So, the enhancement layer frames in a multimedia content is sent with various transmission rates. Therefore, each node can be provided with differentiated quality services. Consequently, QARA is capable of serving heterogeneous population of mobile nodes. Moreover, it can utilize network resources more efficiently. Our simulation results show that QARA outperforms utilization of the available transmission rate and reduces the data transmission time.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.696-706
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• 2008

In this paper we show the performance of the network-adaptive high-definition scalable video streaming using QWLAN board with IEEE 802.11e MAC monitoring and control. Realtime collected MAC parameters are used to determine which video data is extracted for the predicted available bandwidth. To achieve performance, extraction through R-D is proposed instead of the standard video packet extraction. It is shown through experiments that streaming video quality can be enhanced by fast adaptation to network conditions by using the proposed method.

• 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 Broadcast Engineering
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• v.10 no.4 s.29
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• pp.548-556
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• 2005

In wireless network environments, wireless channels are characterized by time-varying fading and interference conditions, which may lead to burst packet corruptions and delay variation. This can cause severe quality degradation of streaming media. To guarantee successful transmission of media over the hostile wireless networks, where channel conditions are highly fluctuating, a flexible and network-adaptive transport method is required. Thus, we propose a network-adaptive transport error control consisting of packet-level interleaved FEC and delay-constrained ARQ, which acts as an application-level transport method of streaming media to alleviate burst packet losses while adapting to the changing channel condition in wireless networks. The performances of the proposed network-adaptive transport error control, general error control schemes, and hybrid schemes are evaluated by a developed simulator at the transport-level and video quality of streaming media. Simulation results show that the proposed mechanism provides the best overall performance among compared other schemes in terms of the transport-level performance of error control and the performance of video quality for streaming media.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.50-53
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• 2002

There are increasing needs to deliver the multimedia streaming over heterogeneous networks. When considering network environments and equipment accessed by user, delivery of video streaming must be scalable. There are many kinds of scalable video coding: spatial, temporal, SNR, and hybrid. The SNR scalable and spatial resolution, but different SNR quality with respect to layers. The 1-layer SNR scalable encoder produces SNR scalable video streams with ease. But, there is drift problem. Modified 1-layer approach does not have this problem but coding inefficiency, and is not MPEG-compliant. The present MPEG-compliant 2-layer encoder comes out to reduce coding rate. But it still use only base layer to encode whole layer. In this paper, we propose adaptive MPEG-compliant 2-layer encoder. Using linear combination algorithm, encoder use 1 motion vector to encode the sequences efficiently. By dong this, we can achieve the coding efficiency of SNR scalable coding.

• KIPS Transactions on Computer and Communication Systems
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• v.3 no.7
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• pp.231-240
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• 2014

Video transcoding techniques provide an efficient mechanism to make a video content adaptive to the capabilities of a variety of clients. However, it is hard to provide an appropriate quality-of-service(QoS) to the clients owing to heavy workload on transcoding operations. In light of this fact, this paper presents the dual transcoding method in order to guarantee QoS in streaming services by maximizing resource usage in a transcoding server equipped with both CPU and GPU computing units. The CPU and GPU computing units have different architectural features. The proposed method speculates workload of incoming transcoding requests and then schedules the requests either to the CPU or GPU accordingly. From performance evaluation, the proposed dual transcoding method achieved a speedup of 1.84 compared with traditional transcoding approach.

• Journal of Korea Multimedia Society
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• v.13 no.3
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• pp.349-358
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• 2010

Adaptability is an important issue in video streaming over mobile environments, since the clients may request videos with great differences in their workload. In this paper, we propose the issues in limited bandwidth scheduling for efficient MPEG-4 video stream transmission over a mobile or wireless network. In the phase of admission control, the amount of bandwidth allocated to serve a video request is the mean bandwidth requirement of its requested video. The dynamic allocation of bandwidth in the phase of scheduling depends on the playback buffer levels of the clients with an objective to make it more adaptive to the playback situation of individual clients. In the proposed RTA scheduling protocol, more bandwidth may be allocated temporarily to the client whose buffer level is low. By employing the buffer level based scheduling policy, this protocol attempts to maximize the real-time performance of individual playback while minimizing the impact of transient overloading. Extensive simulation experiments have been performed to investigate the performance characteristics of the RTA protocol as comparing with BSBA protocol. This RTA protocol shows the better performance by transferring more frames than BSBA protocol.Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• The KIPS Transactions:PartC
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• v.15C no.6
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• pp.547-556
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• 2008

Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose a network adaptive prefetching technique, NAP, for guarantee of constant QoS. The NAP prefetches frames by increasing the frame transmission rate while the available network bandwidth is high. The NAP uses the prefetched frames to guarantee the QoS while the available network bandwidth is low. The experiment result shows that the proposed NAP could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.493-505
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• 2003

A new inter-client synchronization framework for multicast media streaming is proposed employing a server-client coordinated adaptive playout control. The proposed adaptive player controls the playback speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. Additionally, the server implicitly helps increasing the time available for retransmission while the clients perform an interactive error recovery mechanism with the assistance of playout control. The network-simulator based simulations show that the proposed framework can reduce the playout discontinuity without degrading the media quality, and thus mitigate the client heterogeneity.