• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.1
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• pp.56-67
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• 2013

Because H.264/SVC can meet the needs of different networks and user terminals, it has become more and more popular. In this paper, we focus on the spatial resolution scalability of H.264/SVC and propose a blind video watermarking algorithm for the copyright protection of H.264/SVC coded video. The watermark embedding occurs before the H.264/SVC encoding, and only the original enhancement layer sequence is watermarked. However, because the watermark is embedded into the average matrix of each macro block, it can be detected in both the enhancement layer and base layer after downsampling, video encoding, and video decoding. The proposed algorithm is examined using JSVM, and experiment results show that is robust to H.264/SVC coding and has little influence on video quality.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.2
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• pp.71-75
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• 2013

Recently proliferating heterogeneous multimedia service environments should be able to deal with many different transmission speeds, image sizes, or qualities of video. However, not many existing video compression standards satisfy those necessities. To satisfy the functional requirements, the standardization of the H.264/AVC Scalable Extension (SE) technique has been recently completed. It is an extension of the H.264/AVC which can encode several image sizes and qualities at the same time as a single bitstream. To perform optimum mode decision, motion estimation is performed for all MB modes, and the RD costs are compared to identify an MB mode with the smallest RD cost. This increases computational complexity of H.264/AVC SE encoding. In this paper, we propose an early skip mode detection scheme to reduce candidate modes and suggest an algorithm of fast mode decision utilizing reference modes according to the mode history.

• Journal of Broadcast Engineering
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• v.17 no.5
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• pp.765-780
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• 2012

In this paper, we propose a fast mode decision method that determines the coding unit depth for enhancement layers to improve an encoding speed of a scalable video encoder based on HEVC. To decide the coding unit depth of the enhancement layer, firstly, the coding unit depth of the corresponded coding unit in the basement layer is employed. At this stage, the final CU depth is decided by calculating the rate-distortion costs of one lower depth to one upper depth of the referenced depth. The proposed method can reduce a computational load since it does not calculate the rate-distortion costs for all the depths of a target CU. We found that the proposed algorithm decreases encoding complexity of 26% with approximately 1.4% bit increment, compared with the simulcast encoder of the HM 4.0.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.8
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• pp.3121-3135
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• 2015

Owing to the increase in consumption of multimedia content and the improvement of device capacity, user demand for high-quality content has increased. However, it is difficult to transport such large amounts of content over the existing broadcasting network with limited bandwidth. To provide high-definition broadcasting, some studies suggest methods of transporting multimedia over heterogeneous networks after encoding content hierarchically. MPEG Media Transport (MMT), standardized by Moving Picture Experts Group (MPEG), is a solution that enables large-volume media transport over heterogeneous networks such as digital broadcasting networks and packet-switched networks. In the case of delivering a scalable encoded video over different networks, synchronization of each stream is an important issue. MMT defines a synchronization scheme, but does not contain sufficient functions to implement it. In this paper, we propose a synchronization scheme for media streams that are encoded hierarchically, divided into layers, and transported over heterogeneous networks. We implement our scheme using MMT and HTTP, and experimented using three encapsulated video streams with different durations. As a result, we show that the proposed scheme can reduce the waiting time to display high-quality video, relative to Dynamic Adaptive Streaming over HTTP-Scalable Video Coding (DASH-SVC) by requesting segments of enhanced layers after calculating the transmission time. Additionally, we find out that the selection of durations have a relation to the characteristics of the video.

• 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 Digital Contents Society
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• v.17 no.2
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• pp.89-96
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• 2016

SVC(Scalable Video Coding), which is one form among video encoding technologies, makes video streaming with the various frame rate, resolution, and video quality by combining three different scalability dimensions: temporal, spatial, and video quality scalability. As the above SVC-encoded video streaming consists of one base layer and several enhancement layers, and a wireless AP(Access Point) chooses and sends a suitable layer according to the received power from the receiving terminals in the changeable wireless network environment, the receiving terminals supporting SVC are able to receive video streaming with the appropriate resolution and quality according to their received powers. In this paper, after the performance analysis for the received power, packet loss rate, PSNR(Required Peak Signal to Noise Ratio), video quality level and amount of received video data based on the number of SVC layers was performed, an efficient method for selecting the number of SVC layer satisfying the RSNR and minimizing the amount of received video data is proposed.

• 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.5
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• pp.425-436
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• 2009

In recent years, the user demands have increased for multimedia service of high quality over the broadband convergence network. These rising demands for high quality multimedia service led the popularization of various user terminals and large scale display equipments, which needs a variety type of QoS (Quality of Service). In order to support demands for QoS, numerous research projects are in progress both from the perspective of network as well as end system; For example, at the network perspective, QoS guaranteeing by improving of internet performance such as Active Queue Management, while at the end system perspective, SVC (Scalable Video Coding) encoding scheme to guarantee media quality. However, existing AQM algorithms have problems which do not guarantee QoS, because they did not consider the essential characteristics of video encoding schemes. In this paper, it is proposed to solve this problem by deploying the TS- AQM (Temporal Scalability Active Queue Management) which employs the differentiated packet dropping for dependency of the temporal level among the frames, based on SVC encoding characteristics by exploiting the TID (Temporal ID) field of the SVC NAL unit header. The proposed TS-AQM guarantees multimedia service quality through video decoding reliability for SVC streaming service, by differentiated packet dropping when congestion exists.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2075-2078
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• 2002

Internet became the most popular network in spite of its weakness in realtime multimedia service. Many experts believe that the Internet has the potential to become the main multimedia distribution network of the near future. Currently, it does not provide any (BoS guarantees and, even when it does, guaranteed quality delivery of video may turn out to be too expensive. Unavoidable packet losses and delay jitter caused by congestion in a best effort delivery environment require use of intelligent transport techniques for effective video delivery. According to market needs of better quality of service (QoS) fur realtime multimedia services over Internet, they have been standardizing RSVP, IntServ, and DiffServ This paper combines the benefits of QoS mechanisms such as RSVP/IntServ with scalable video encoding. We propose that more important bit stream is given more priority such that limited network resources are guaranteed far the stream. Various prioritizing approaches are proposed and compared to normal approach by using Network Simulator. The calculated QoS parameters such as packet loss rate are used to calculate degree of degradation in video quality. In this Paper, proposed methods can be implemented adaptively to Von protocol, such as H.323, SIP.

• Journal of Broadcast Engineering
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• v.12 no.4
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• pp.360-372
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• 2007

We propose a fast inter-layer mode decision method by utilizing coding information of base layer upward its enhancement layer inscalable video coding (SVC), also called MPEG-4 part 10 Advanced Video Coding Amendment 3 or H.264 Scalable Extension (SE) which is being standardized. In this paper, when the motion vectors from the base layer have zero motion (0, 0) in inter-layer motion prediction or the Integer Transform coefficients of the residual between current MB and the motion compensated MB by the predicted motion vectors from the base layer are all zero, the block mode of the corresponding block to be encoded at the enhancement layer is determined to be the $16{\times}16$ mode. In addition, if the predicted mode of the MB to be encoded at the enhancement layer is not equal to the $16{\times}16$ mode, then the rate-distortion optimization is only performed on the reduced candidated modes which are same or smaller partitioned modes. Our proposed method exhibits the complexity reduction in encoding time up to 72%. Nevertheless, it shows negligible PSNR degradation and bit rate increase up to 0.25dB and 1.73%, respectively.