• IEIE Transactions on Smart Processing and Computing
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• v.3 no.6
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• pp.353-357
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• 2014

This paper, presents an efficient hardware architecture of high performance SVC(Scalable Video Coding). This platform uses dedicated hardware architecture to improve its performance. The architecture was prototyped in Verilog HDL and synthesized using the Synopsys Design Compiler with a 65nm standard cell library. At a clock frequency of 266MHz, This platform contains 2,500,000 logic gates and 750,000 memory gates. The performance of the platform is indicated by 30 frames/s of the SVC encoder Full HD($1920{\times}1080$), HD($1280{\times}720$), and D1($720{\times}480$) at 266MHz.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1999.06a
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• pp.63.1-68
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• 1999

MPEG-2 transcoding scheme is proposed as scalable transmission system for a video-on-demand system is described. In this system video contents encoded by MPEG-2 are transmitted to receiver terminals via communication lines that have different bit rate. A transcoding technique that directly converts a bit stream encoded by MPEG-2 into a lower coding rate stream without decoding is proposed. The transcoder called Trampeg includes size reduction and frame drop approach. The experimental results show that an MPEG-2 stream of 0 Mbps is able to be transcoded into a stream of 1.5 Mbps-300 Kbps.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.796-806
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• 2009

SVC (Scalable Video Coding) is a new video coding standard to provide convergence media service in heterogeneous environments with different networks and diverse terminals through spatial-temporal-quality combined flexible scalabilities. This paper presents the performance analysis on packet loss in the delivery of SVC over IP networks and an efficient adaptation method to packet loss caused by buffer overflow. In particular, SVC with MGS (Medium Grained Scalability) as well as spatial and temporal scalabilities is addressed in the consideration of packet-based adaptation since finer adaptation is possible with a sufficient numbers of quality layers in MGS. The effect on spatio-temporal quality due to the packet loss of SVC with MGS is evaluated. In order to minimize quality degradation resulted by packet loss, the proposed adaptation of MGS based SVC first sets adaptation unit of AU (Access Unit) or GOP corresponding to allowed delay and then selectively discards packets in order of importance in terms of layer dependency. In the experiment, the effects of packet loss on quantitative qualities are analyzed and the effectiveness of the proposed adaptation to packet loss is shown.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2007.02a
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• pp.13-17
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• 2007

스케일러블 비디오 코딩(SVC, Scalable Video Coding)은 MPEG(Moving Picture Expert Group)과 VCEG (Video Coding Expert Group)의 JVT(Joint VIdeo Team)에 의해 현재 표준화 되고 있는 새로운 압축 표준 기술이며 시간, 공간 및 화질의 스케일러빌리티를 지원하기 위해 계층 구조를 가지고 있다. 공간적 스케일러빌리티를 위해 기본 계층으로부터 텍스처, 움직임 그리고 잔차신호 정보를 예측하여 사용한다. 그러나 고효율의 압축효과를 얻기 위해 기존의 방식에서는 기본계층에서 얻은 세가지 정보이외에 현재 향상 계층에서 자체적으로 얻은 부호화 정보를 비교하여 최소의 RD(Rate Distortion) 비용을 가지는 정보를 이용하여 부호화 하도록 되어 있다. 하지만 이러한 방식은 향상 계층에서 인터 모드 결정 시 $16\times16,\;16\times8,\;8\times16,\;8\times8,\;4\times4,\;4\times8,\;4\times4$ 블록 모드에 대한 움직임 벡터 예측 및 보상 과정을 거쳐야 하기 때문에 향상 계층에서의 부호화 복잡도는 기본 계층에 비해 상당히 증가하게 된다. 본 논문에서는 기본계층에서 예측한 움직임 벡터 정보를 이용하여 항상 계층에서 모드 결정을 고속화하는 방법에 대해 소개한다. 제안된 방법은 기본 계층에서 예측한 블록모드 중에서 큰 블록인 $16\times16$ 블록에서 움직임 벡터가 (0, 0) 일 경우에 대하여 향상 계층에서는 $16\times16$매크로 블록에 대해서만 움직임 예측 및 보상을 수행함으로써 향상 계층에서 움직임 모드 결정을 조기에 완료하게 된다. 이것은 하위 공간 계층에서 예측한 움직임 벡터 정보가 아주 작을 때는 큰 블록 크기로 모드로 결정되는 일반적인 원리를 이용한 것이고 이 제안 방법을 이용하였을 경우 향상계층에의 모드 결정과정을 고속화함으로써 전체 스케일러빌 비디오 부호하기의 연산량 및 복잡도를 최대 70%까지 감소 시켰다. 그러나 연산량 감소에 따른 비트율의 증가와 화질 열화는 각각 최대 1.32%와 최대 0.11dB로 무시할 수 있을 정도로 작음을 확인 하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2006.11a
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• pp.111-115
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• 2006

스케일러블 비디오 코딩(SVC, Scalable Video Coding)은 MPEG(Moving Picture Expert Group)과 VCEG (Video Coding Expert Group)의 JVT(Joint VIdeo Team)에 의해 현재 표준화 되고 있는 새로운 압축 표준 기술이며 시간, 공간 및 화질의 스케일러빌리티를 지원하기 위해 계층 구조를 가지고 있다. 특히 시간적 스케일러빌리티를 위해 계층적 B-픽처 구조를 채택하고 있다. 스케일러블 비디오 코딩의 기본 계층은 H.264|AVC와 호환적이므로, 모션 예측과 모드 결정과정에서 $16{\times}16,\;16{\times}8,\;8{\times}16,\;8{\times}8,\;8{\times}4,\;4{\times}8$ 그리고 $4{\times}4$와 같은 7개의 서로 다른 크기를 갖는 블록을 사용한다. 스케일러블 비디오 코딩에서 사용되고있는 계층적 B-픽처 구조는 키 픽처인 I와 P 픽처를 제외하고는 한 GOP (Group of Picture)내에서 모두 B-픽처를 사용하므로 H.264|AVC와 비교했을 때 연산량 증가와 함께 부호화 지연도 급격히 증가한다. B-픽처는 양방향 모션 벡터인 LIST0와 LIST1을 사용하고 양방향 모두에서 다중 참조 픽처를 사용하기 때문이다. 본 논문에서는 통계적 가선 검증을 이용하여 스케일러블 비디오 부호화에 적용 가능한 고속 프레임간 모드 결정 알고리듬 대해 소개한다. 제안된 방법은 $16{\times}16$ 매크로 블록과 $8{\times}8$ 서브 매크로 블록에 통계적 가설 감증 기법을 적용하여 실행되며, 현재 블록과 복원된 참조 블록간의 픽셀 값을 비교하여 RD(Rate Distortion) 최적화 기반 모드 결정을 빨리 완료함으로써 고속 프레임간 모드 결정을 가능하게 한다. 제안된 방법은 프레임 간 모드 결정을 고속화함으로써 스케일러블 비디오 부호화기의 연산량과 복잡도를 최대 57%감소시킨다. 그러나 연산량 감소에 따른 비트율의 증가나 화질의 열화는 최대 1.74% 비트율 증가 및 0.08dB PSNR 감소로 무시할 정도로 작다.

• 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 Society of Computer and Information
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• v.5 no.3
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• pp.29-32
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• 2000

Layered video coding schemes allow the video information to be transmitted in multiple video bitstreams to achieve scalability. they are attractive in theory for two reasons. First, they naturally allow for heterogeneity in networks and receivers in terms of client processing capability and network bandwidth. Second, they correspond to optimal utilization of available bandwidth when several video qualify levels are desired. In this paper we propose a scalable video codec architectures with motion estimation, which is suitable for real-time audio and video communication over packet networks. The coding algorithm is compatible with ITU-T recommendation H.263+ and includes various techniques to reduce complexity. Fast motion estimation is Performed at the H.263-compatible base layer and used at higher layers, and perceptual macroblock skipping is performed at all layers before motion estimation. Error propagation from packet loss is avoided by Periodically rebuilding a valid Predictor in Intra mode at each layer.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12C
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• pp.1173-1183
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• 2006

In this paper, we propose an efficient method for improving visual quality of AR-FGS (Adaptive Reference FGS) which is adopted as a key scheme for SVC (Scalable Video Coding) or H.264 scalable extension. The standard FGS (Fine Granularity Scalability) adopts AR-FGS that introduces temporal prediction into FGS layer by using a high quality reference signal which is constructed by the weighted average between the base layer reconstructed imageand enhancement reference to improve the coding efficiency in the FGS layer. However, when the enhancement stream is truncated at certain bitstream position in transmission, the rest of the data of the FGS layer will not be available at the FGS decoder. Thus the most noticeable problem of using the enhancement layer in prediction is the degraded visual quality caused by drifting because of the mismatch between the reference frame used by the FGS encoder and that by the decoder. To solve this problem, we exploit the principle of cyclical block coding that is used to encode quantized transform coefficients in a cyclical manner in the FGS layer. Encoding block coefficients in a cyclical manner places 'higher-value' bits earlier in the bitstream. The quantized transform coefficients included in the ealry coding cycle of cyclical block coding have higher probability to be correctly received and decoded than the others included in the later cycle of the cyclical block coding. Therefore, we can minimize visual quality degradation caused by bitstream truncation by adjusting weighting factor to control the contribution of the bitstream produced in each coding cycle of cyclical block coding when constructing the enhancement layer reference frame. It is shown by simulations that the improved AR-FGS scheme outperforms the standard AR-FGS by about 1 dB in maximum in the reconstructed visual quality.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.4
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• pp.491-508
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• 2010

In peer-to-peer (P2P) on-demand streaming networks, the alleviation of server load depends on reciprocal stream sharing among peers. In general, on-demand video services enable clients to watch videos from beginning to end. As long as clients are able to buffer the initial part of the video they are watching, on-demand service can provide access to the video to the next clients who request to watch it. Therefore, the key challenge is how to keep the initial part of a video in a peer's buffer for as long as possible, and thus maximize the availability of a video for stream relay. In addition, to address the issues of delivering data on lossy network and providing scalable quality of services for clients, the adoption of multiple description coding (MDC) has been proven as a feasible resolution by much research work. In this paper, we propose a novel caching scheme for P2P on-demand streaming, called Dynamic Buffering. The proposed Dynamic Buffering relies on the feature of MDC to gradually reduce the number of cached descriptions held in a client's buffers, once the buffer is full. Preserving as many initial parts of descriptions in the buffer as possible, instead of losing them all at one time, effectively extends peers’ service time. In addition, this study proposes a description distribution balancing scheme to further improve the use of resources. Simulation experiments show that Dynamic Buffering can make efficient use of cache space, reduce server bandwidth consumption, and increase the number of peers being served.

• Proceedings of the Korean Information Science Society Conference
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• 2010.06a
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• pp.196-197
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• 2010