• 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.

• Proceedings of the KIEE Conference
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• 2008.04a
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• pp.83-85
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• 2008

현행 지상파 DMB(Digital Multimedia Broadcasting)에서는 최대 CIF급 해상도의 동영상을 384 Kbps 정도의 비트율로 압축하여 비디오 CD급의 화질을 제공한다. 따라서 큰 화면을 통해 DMB 방송을 시청할 경우 해상도 부족 등으로 인한 화질 열화를 느낄 수 있다. 이와 같은 문제를 해결하기 위한 방안으로 계층적으로 부호화된 비트스트림을 서로 다른 채널로 전송하는 고품질 AT-DMB 시스템이 연구되고 있다. 더불어, 계층변조 기술을 이용하여 추가적인 전송대역폭을 확보하고, 계층적 비디오 부호화 기술을 이용하여 고품질 이동 멀티미디어 방송을 제공하는 고품질 T-DMB 방식이 개발되고 있다. 이와 같은 AT-DMB 방식으로 사용자에게 안정된 고품질 비디오 서비스를 제공하기 위해서는 계층적 비디오 부호화를 고려한 효율적인 AV다중화 및 동기화 구조에 대한 연구가 필수적으로 요구 된다. 본 논문에서는, 이와 같은 고품질 AT-DMB 시스템을 위해서 각 계층 ES(Elementary stream)을 MPEG-4 over MPEG-2 시스템과 Eureka-147 DAB 시스템을 통해 서로 다른 채널로 전송한다. 계층별 전송으로 인한 시간 지연에 대처하여 수신입력부에서는 각각의 계층으로 전송된 ES들이 동기화 되어 복호화 되도록 SVC의 계층적 비트스트림 구조와 적응적으로 동작되는 다중화 시스템구조를 제작하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.517-525
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• 2009

We propose a quality guaranteed scalable video streaming service over the Internet using a new rate adaptation algorithm. Because video data requires much more bandwidth rather than other types of service, therefore, quality of video streaming service should be guaranteed while providing friendliness with other service flows over the Internet. To successfully provide this, we propose a framework for providing quality-guaranteed streaming service using two-channel transport layer and rate adaptation of scalable video stream. In this framework, baseline layer for scalable video is transmitted using TCP transport for minimum qualify service. Enhancement layers are delivered using TFRC transport with layer adaptation algorithm. The proposed framework jointly uses the status of playout buffer in the client and the encoding rate of layers in media contents. Therefore, the proposed algorithm can remarkably guarantee minimum quality of streaming service rather than conventional approaches regardless of network congestion and the encoding rate variation of media content.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.454-456
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• 2001

차세대 인터넷에 있어 새로운 응용 서비스의 제공은 반드시 필요한 문제이다. 기존에 일반 데이터 서비스를 포함하여, 음성, 비디오 서비스를 실시간으로 제공할 필요성이 증가되고 있다. 이를 해결하기 위해서는 계층 별로, 비디오 데이터의 압축 방식, 네트워크 자원의 예약, 세션 관리, 호 혀용 제어, 패킷 스케쥴링, 버퍼 관리 등 여러 가지 기술들이 필요하다. 본 논문에서는 실시간 비디오 서비스를 효과적으로 사용자에게 제공하기 위한 시스템 구조와 요구사항, 구현 방법을 다루었다. 특히 용용 계층을 고려한 스케쥴링과 버퍼 관리 기법의 필요성을 MPEG 비디오 데이터를 사용한 실험을 통해 제시하였다. 결과적으로 응용 계층을 고려한 스케쥴링과 버퍼 관리 기법이 서비스 품질(PSNR)을 향상시킬 수 있었으며, 실시간 비디오 서비스를 지원하는데 효과적으로 이용될 수 있다는 것을 보였다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.260-263
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• 2005

본 논문에서는 차세대 유무선 광대역 통합망에서 실시간 멀티미디어의 응용 계층 품질을 평가하기 위한 실시간 음성 전송 플랫폼 라이브러리를 개발하였다. 개발된 라이브러리는 실시간 음성 통신시 사용자가 요구할 수 있는 다양한 품질을 제공하기 위해 전송률과 압축 방식에 따른 음성 코덱들을 포함하고 있다. 본 라이브러리는 다양한 환경에도 동일 입출력, 다중스레드, 인터넷 통신 등이 가능한 PWLIB를 기반으로 한다. 음성 데이터는 RTP/UDP/IP를 이용하여 패킷화되고 RTCP를 이용하여 전송 품질이 모니터링된다. 개발된 음성 통신 라이브러리를 이용하여 간단한 음성 통신 시스템을 구현하였으며, 음성 코덱별로 네트워크 상에서 송수신 테스트를 진행하였다. 본 라이브러리는 비디오 코덱 및 시그널링 및 네트웍 자원 예약 프로토콜과 결합되어 멀티미디어 통신 단말 개발에 사용 될 수 있다.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.36C no.9
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• pp.76-84
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• 1999

EZW proposed by Shapiro is based on a zerotree constructed in a way that a parent coefficient in a subband is related to four child coefficients in the next finer subband of similar orientation. This fixed treeing based on 1-to-4 parent-child is suitable to exploti hierachical correlations among subbands but not to exploit spatial correlations within a subband. A new treeing by Joo, et al. is suggested to simulatneously exploit those two correlatins by extending parent-child relationship in a flexible way. The flexible treeing leads to increasing the number of symbols and lowering entorpy comparing to the fixed treeing, and therefore a better compression can be resulted. In this paper, we suggest two techniques to suppress the increasing of symbols. First, a probing bit is generated to avoid redundant scan for insignivicant coefficients. Second, since all subbands do not always require the same kind of symbol-set, produced symbols are re-symbolized into binary codes according to a pre-defined procedure. Owing to those techniques, all symbols are generated as binary codes. The binary symbols can be entropy-coded by an adaptive arithmetic coding. Moerover, the binary symbol stream can give comparatively good performances without help of additional entropy coding. Our proposed coding scheme is suggested in two modes: binary coding mode and arithmetic coding mode. We evaluate the effectivenessof our modifications by comparing with the original EZW.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.393-398
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• 2002

The conventional bit allocation scheme for MPEG-1 audio encoding searches the subband with minimum MNR(mask-to-noise ratio) repetitively until its operation is completed, which occupies most of its total computational complexity. In this paper, as a computationally efficient approximation of it, we propose a new bit allocation scheme with a simple subband search and compare it with the existing schemes[1][2] in terms of the computational complexity and sound quality. For the performance comparison, we used the pop music signal contained in SQAM(sound quality assess material) CD from EBU. Simulation results show that the computational complexity of the proposed method is about 42% of that of the existing one in [1] and the sound quality difference in terms of MNR between the two schemes is within the 0.2 ㏈, for the case of using the layer II at the bit rate of 128 kbps.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7C
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• pp.921-930
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• 2004

In this paper, a robust video transmission system is proposed. To effectively prevent the corruption of video stream and its propagation in spatial and temporal domains, a version of turbo code, so-called as byte-aligned variable-length turbo code, is applied. Protection performance of the proposed turbo code is first evaluated by applying it to GOB-based variable-size ITU-T H.263+ video packets, where the protection level is statically controlled based on the joint source-channel criteria. This protection is then extended to support the adaptation of code ratio to best match the time-varying channel condition. The time-varying Rayleigh fading channel is modelled considering the correlation of the fading channel. The resulting performance comparison with the static turbo code as well as the conventional RCPC code clearly demonstrates the possibility of the proposed adaptation approach for the time-varying correlated Rayleigh-fading channel.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.5
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• pp.136-146
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• 2012

As information technology used for simulation and virtual reality developed, there is a growing interest in animation technologies which will effectively deliver simulation results to users. Various efforts have been made to improve animation performance, like playback quality and speed, input-output speed and storage space reduction. However, earlier studies generally focused on image compression frame by frame. To significantly improve storage space and playback speed, animation data should be vectorized. Also, spatial and temporal duplication have to be removed. In this study, animation data structure was improved fundamentally through establishment of hierarchy and vectorization. Also Spatial and temporal duplication of animation data was removed through vectorization based on train route. As a result, storage space was reduced, input-output speed and playback speed were considerably improved. According to the test, additional Patternization which followed vectorization brought reduction of over 80% in storage space and input-output speed was quadrupled. Patternization technology can be used as a proper storage method of animation data, and can provide user-specific animation by small data transmission.