• Journal of Internet Computing and Services
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• v.11 no.2
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• pp.143-153
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• 2010

When providing high quality panoramic video across the Internet, mobile communications, and broadcasting areas, it requires a suitable video codec that satisfies both high-compression efficiency and random access functionality. The users must have high-compression efficiency in order to enable video streaming of high-volume panoramic data. Random access allows the user to move the viewpoint and direction freely. In this paper, we propose the parallel processing scheme under cell units in order to improve the performance of streaming service for large screen panoramic video in 10Mbps bandwidths based on H.264/AVC with high compression rate. This improved algorithm divides a screen composed of cells less than $256{\times}256$ in size, encodes it, and decodes it with the cells in the present view. At this point, encoding/decoding is parallel processed by the present cell units. Also, since the cells only included in the present view are packed and transmitted, the possible processing of not extricating blocks is proven by experiment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.12
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• pp.2871-2877
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• 2014

Real-time communication and diversification of multimedia application services using high-capacity data over the Internet demand high speed connection, and real-time and high-speed access internet service such as multimedia make Internet traffic increase rapidly. Meanwhile, QoS (Quality of Service) of internet service is not satisfied. In this situation, ISP (Internet Service Provider) has been required to improve service quality and extend network according to the user's needs. This request includes issue of extending network focused on increasing router and the number of routing table as well as simply extending bandwidth.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.6
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• pp.736-745
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• 2002

In network delivery of compressed video, packets may be lost if the channel is unreliable like Internet. Such losses tend to of cur in burst like continuous bit-stream error. In this paper, we propose an effective error-concealment approach to which an error resilient video encoding approach is applied against burst errors and which reduces a complexity of error concealment at the decoder using data hiding. To improve the performance of error concealment, a temporal and spatial error resilient video encoding approach at encoder is developed to be robust against burst errors. For spatial area of error concealment, block shuffling scheme is introduced to isolate erroneous blocks caused by packet losses. For temporal area of error concealment, we embed parity bits in content data for motion vectors between intra frames or continuous inter frames and recovery loss packet with it at decoder after transmission While error concealment is performed on error blocks of video data at decoder, it is computationally costly to interpolate error video block using neighboring information. So, in this paper, a set of feature are extracted at the encoder and embedded imperceptibly into the original media. If some part of the media data is damaged during transmission, the embedded features can be extracted and used for recovery of lost data with bi-direction interpolation. The use of data hiding leads to reduced complexity at the decoder. Experimental results suggest that our approach can achieve a reasonable quality for packet loss up to 30% over a wide range of video materials.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.3
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• pp.341-357
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• 2010

Forward Error Correction (FEC) techniques have been adopted to overcome packet losses and to improve the quality of video delivery. The efficiency of the FEC has been significantly compromised, however, due to the characteristics of the wireless channel such as burst packet loss, channel fluctuation and lack of Quality of Service (QoS) support. We propose herein an Adaptive Cross-layer FEC mechanism (ACFEC) to enhance the quality of video streaming over 802.11 WLANs. Under the conventional approaches, FEC functions are implemented on the application layer, and required feedback information to calculate redundancy rates. Our proposed ACFEC mechanism, however, leverages the functionalities of different network layers. The Automatic Repeat reQuest (ARQ) function on the Media Access Control (MAC) layer can detect packet losses. Through cooperation with the User Datagram Protocol (UDP), the redundancy rates are adaptively controlled based on the packet loss information. The experiment results demonstrate that the ACFEC mechanism is able to adaptively adjust and control the redundancy rates and, thereby, to overcome both of temporary and persistent channel fluctuations. Consequently, the proposed mechanism, under various network conditions, performs better in recovery than the conventional methods, while generating a much less volume of redundant traffic.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.11
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• pp.2103-2120
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• 2011

In this research, we consider the replication strategy for the applications of video streaming in wireless mesh networks (WMNs). In particular, we propose a closed-form of optimal replication densities for a set of frames of a video streaming by exploiting not only the skewed access probability of each frame but also the skewed loss probability and skewed encoding rate-distortion information. The simulation results demonstrate that our method improves the replication performance in terms of user-perceived quality (UPQ) which includes: 1) minimum average maximum reconstructed distortion for high peak signal-to-noise ratio (PSNR), 2) small reconstructed distortion fluctuation among frames for smooth playback, and 3) reasonable average maximum transmission distance for continuous playback. Furthermore, the proposed strategy consumes smaller storage capacity compared to other existing optimal replication strategies. More importantly, the effect of encoding rate is carefully investigated to show that high encoding rate does not always gain high performance of replication for video streaming.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.257-263
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• 2003

Error control and concealment in video communication is becoming increasingly important because transmission errors can cause single or multiple loss of macroblocks in video delivery over unreliable channels such as wireless networks and the internet. This paper describes a temporal error concealment by postprocessing. Lost image blocks are overlapped block motion compensated (OBMC) using median of motion vectors from adjacent blocks at the decoder. The results show a significant improvement over zero motion error concealment and other temporal concealment methods such as Motion Vector Rational Interpolation or Side Match Criterion OBMC by 1.4 to 3.5㏈ gain in PSNR. We present experimental results showing improvements in PSNR and computational complexity.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.6
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• pp.3165-3181
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• 2019

Spherical videos, which are also called 360-degree videos, have become increasingly popular due to the rapid development of virtual reality technology. However, the large amount of data in such videos is a huge challenge for existing transmission system. To use the existing encode framework, it should be converted into a 2D image plane by using a specific projection format, e.g. the equi-rectangular projection (ERP) format. The existing high-efficiency video coding standard (HEVC) can effectively compress video content, but its enormous computational complexity makes the time spent on compressing high-frame-rate and high-resolution 360-degree videos disproportionate to the benefits of compression. Focusing on the ERP format characteristics of 360-degree videos, this work develops a fast decision algorithm for predicting the coding unit depth interval and adaptive mode decision for intra prediction mode. The algorithm makes full use of the video characteristics of the ERP format by dealing with pole and equatorial areas separately. It sets different reference blocks and determination conditions according to the degree of stretching, which can reduce the coding time while ensuring the quality. Compared with the original reference software HM-16.16, the proposed algorithm can reduce time consumption by 39.3% in the all-intra configuration, and the BD-rate increases by only 0.84%.

• Journal of Internet Computing and Services
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• v.17 no.4
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• pp.19-26
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• 2016

In this paper, the dynamic buffering based smart OTT platform was proposed, and analyzed for adaptive bit-rate video delivery with the optimization of HLS (HTTP Live Streaming). This platform consists of the software platform between sever and client which detects the bandwidth capacity, and adjusts the quality of the streaming for multiple bit-rates resolutions. In order to apply adaptive buffering, two buffers are added to the basic HLS player, and each buffer is responsible for constantly buffering a previous and the next channels relative to the current channel. This adaptive transmitting with smart OTT platform is superior to delivering a static video file at a single buffering, because the video stream of adaptive double buffers can be switched streaming according to client's available network speed. As a result, this proposed smart OTT can be cooperated to the application of HLS server with segmented H.265 MPEG-2 TS video & m3u8 files with its information based on the optimized transmission channel state of live and VOD, and applied to PLC transmission, too.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.4077-4091
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• 2017

This paper presents the design of a broadcasting scenario and system for an 8K-resolution content. Due to an 8K content is four times larger than the 4K content in terms of size, many technologies such as content acquisition, video coding, and transmission are required to deal with it. Therefore, high-quality video and audio for 8K (ultra-high definition television) service is not possible to be transmitted only using the current terrestrial broadcasting system. The proposed broadcasting system divides the 8K content into four 4K contents by area, and each area is hierarchically encoded by Scalable High-efficiency Video Coding (SHVC) into three layers: L0, L1, and L2. Every part of the 8K video content divided into areas and hierarchy is independently treated. These parts are transmitted over heterogeneous networks such as digital broadcasting and broadband networks after going through several processes of generating signal messages, encapsulation, and packetization based on MPEG media transport. We propose three methods of generating streams at the sending entity to merge the divided streams into the original content at the receiving entity. First, we design the composition information, which defines the presentation structure for displays. Second, a descriptor for content synchronization is included in the signal message. Finally, we define the rules for generating "packet_id" among the packet header fields and design the transmission scheduler to acquire the divided streams quickly. We implement the 8K broadcasting system by adapting the proposed methods and show that the 8K-resolution contents are stably received and serviced with a low delay.

• Journal of Korea Multimedia Society
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• v.3 no.5
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• pp.479-494
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• 2000

Although the proliferation of real-time multimedia services over the Internet might indicate its successfulness in dealing with heterogeneous environments, it is obvious, on the other hand, that the internet now has to cope with a flood of multimedia data which consumes most of network communication channels due to a great deal of video or audio streams. Therefore, for the purpose of an efficient and appropriate utilization of network resources, it requires to develop and deploy a new scalable transmission technique n consideration of respective network environment and individual clients computing power. Also, we can eliminate the waste effects of storage device and data transmission overhead in that the same video stream duplicated according to QoS. The purpose of this paper is to develop a technology that can adjust the amount of data transmitted as an MPEG video stream according to its given communication bandwidth, and technique that can reflect dynamic bandwidth while playing a video stream. For this purpose, we introduce a media scalable media decomposer working on server side, and a scalable media composer working o n a client side, and then propose a scalable transmission method and a media sender and a media receiver in consideration of dynamic QoS. Those methods proposed her can facilitate an effective use of network resources, and provide multimedia MPEG video services in real-time with respect to individual client computing environment.