• Journal of Korea Multimedia Society
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• v.15 no.4
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• pp.492-500
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• 2012

In this paper, improved contour region coding method is proposed to accomplish better depth map coding performance. First of all, in order to use correlation between color video and depth map, a structure in SVC is applied to 3DVC. This can reduce bit-rate of the depth map while supporting the video to be transferred via various collection of network. As the depth map is mainly used to synthesize videos from different views, corrupted contour region can damage the overall quality of video. We hereby adapt a new differential quantization method when separating the contour region. The experimental results show that the proposed method can improve video quality by 0.06~0.5dB which translate the bit rate saving by 0.1~1.15%, when compared to the reference software.

• The KIPS Transactions:PartB
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• v.18B no.5
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• pp.249-254
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• 2011

FGS (fine granularity scalability) supporting scalability in MPEG-4 Part 2 is a scalable video coding scheme that provides bit-rate adaptation to varying network bandwidth thereby achieving of its optimal video quality. In this paper, we proposed FGS coding scheme which performs one more bit-plane coding for residue signal occured in the enhancement-layer of the basic FGS coding scheme. The experiment evaluated in terms of video quality scalability of the proposed FGS coding scheme by comparing with FGS coding scheme of the MPEG-4 verification model (VM-FGS). The comparison was conducted by analysis of PSNR values of three tested video sequences. The results showed that when using rate control algorithm VM5+, the proposed FGS coding scheme obtained Y, U, V PSNR of 0.4 dB, 9.4 dB, 9 dB averagely higher and when using fixed QP value 17, obtained Y, U, V PSNR of 4.61 dB, 20.21 dB, 16.56 dB averagely higher than the existing VM-FGS. From results, we found that the proposed FGS coding scheme has higher video quality scalability to be able to achieve video quality from minimum to maximum than VM-FGS.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.189-192
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• 2008

Modern broadcasting/multicasting networks has the heterogeneous nature in terms of terminals and available bandwidth. Such heterogeneity could be coped by scalable video coding (SVC) standard developed recently. More specifically, spatial layers of an SVC bitstream can be consumed by different terminals and SNR (and temporal) scalability can be used to cope with bandwidth heterogeneity. In this work, we tackle the problem of SVC adaptation for different user groups receiving the same broadcast/multicast video, so as to provide a flexible tradeoff between the groups while also maximizing the overall quality of the users. The adaptation process to truncate an SVC bitstream is first formulated as an optimization problem. Then the problem is represented by MPEG-21 DIA description tools, which can be solved by a universal processing. The results show that MPEG-21 DIA is useful to enable automatic and interoperable adaptation in our scenario.

• 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 the Korea Institute of Information Security & Cryptology
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• v.20 no.1
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• pp.129-134
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• 2010

In this paper, we propose a redistribution mechanism of the content that is adapted to devices, which may have different display size and computing capabilities, in home network. The proposed system introduces a mechanism that the encrypted content compressed by H.264/SVC(Scalable Video Coding) scheme which has been standardized recently is provided to the device into a level of content suitable to each device capability. To efficiently superdistribute SVC content, this paper defines three requirements and proposes redistribution mechanism which satisfies these requirements using another licence that it is called 'Ticket'. Our system allows devices to redistribute the content freely in the domain using domain key.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.4092-4104
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• 2017

In this paper, an efficient 4K and 8K UHD(Ultra High Definition) transmission scheme is proposed on the convergence networks with broadcasting and LTE(Long Term Evolution) by using CoMP(Coordinated Multi-Point). A video data is compressed and divided into BL(Base Layer), E(Enhanced layer)1, E2 and E3 by scalable HEVC(High Efficiency Video Coding). The divided layers can be combined by the scalable HEVC such as mobile HD, full HD, 4K and 8K UHD(Ultra High Definition). The divided layers are transmitted through the convergence networks with DVB-T2(Digital Video Broadcasting-$2^{nd}$ Generation Terrestrial) broadcasting system and LTE CoMP. This scheme transmits mobile HD and full HD layers through DVB-T2 broadcasting system by using M-PLP(Multiple-physical Layer Pipes), and adaptively transmits 4K or 8K UHD layer through LTE CoMP with MMT(MPEG Media Transport) server. An adaptive transmitting and receiving scheme in the LTE CoMP system provides 4K or 8K UHD layer to a user according to the user status. The proposed scheme is verified by showing the system-level simulation results which is better BER(bit-error-rate) performance than the conventional scheme. The results show that the proposed scheme provides the stable video contents to the user especially at the cell edge.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.102-108
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• 2007

A high-Performance scalable speech codec generally requires a very low-rate first layer and a fine granule second layer, and this codec can be implemented with the harmonic codec and the MDCT-based transform codec for each layer. In this structure, however. each codec requires independent frequency transform and the time delay of each codec is accumulated. resulting in long time delay for the overall codec. In this paper, new MDCT structure in the second layer is Proposed. where MDCT is forced to share the look-ahead region of the first layer in order to prevent the time delay accumulation and the resulting functional error of MDCT is analyzed and removed after IMDCT The Proposed delayless MDCT requires no additional bits and Provides the equivalent coding performance with the reduced time delay, yielding a meaningful enhancement of the overall codec.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.4
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• pp.1479-1501
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• 2020

Dependable functioning of dynamic networks is essential for delivering ubiquitous services. Faults are the root causes of network outages. The comparison diagnosis model, which automates fault's identification, is one of the leading approaches to attain network dependability. Most of the existing research has focused on stationary networks. Nonetheless, the time-free comparison model imposes no time constraints on the system under considerations, and it suits most of the diagnosis requirements of dynamic networks. This paper presents a novel protocol that diagnoses faulty nodes in diagnosable dynamic networks. The proposed protocol comprises two stages, a testing stage, which uses the time-free comparison model to diagnose faulty neighbour nodes, and a disseminating stage, which leverages a Random Linear Network Coding (RLNC) technique to disseminate the partial view of nodes. We analysed and evaluated the performance of the proposed protocol under various scenarios, considering two metrics: communication overhead and diagnosis time. The simulation results revealed that the proposed protocol diagnoses different types of faults in dynamic networks. Compared with most related protocols, our proposed protocol has very low communication overhead and diagnosis time. These results demonstrated that the proposed protocol is energy-efficient, scalable, and robust.

• The KIPS Transactions:PartC
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• v.17C no.5
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• pp.415-426
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• 2010

CAS is an access control system by which only legal users can access contents. IPTV is a spotlighted system that uses CAS. H.264/SVC is a coding standard that provides a scalable coding method by which users who are in various network environments and have various devices can receive the contents. In this method, the contents are coded in a layered structure to make users choose the quality of the receiving contents. Therefore, contents provider should be able to control users to access only appropriate contents according to their subscriptions. The structure of CAS being employed in many applications is not suitable to control access for H.264/SVC. In this paper, we provide an efficient access control method and a key management method for H.264/SVC contents using CAS.