• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.380-389
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• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of KIISE:Information Networking
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• v.36 no.1
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• pp.34-42
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• 2009

In this paper, we propose a practical hybrid transport error control scheme to provide SVC video streaming service over error-prone IP networks. Many error control mechanisms for various video coding standards have been proposed in the literature. However, there is little research result which can be practically applicable to the multilayered coding structure of SVC(the scalable extension of H.264/AVC). We present a new hybrid transport error control scheme that efficiently combines layered Forward Error Correction(FEC) and Automatic Repeat Request(ARQ) for better packet-loss resilience. In the proposed hybrid error control, we adopt ACK-based ARQ instead of NACK-based ARQ to maximize throughput which is the amount of effective data packets delivered over a physical link per time unit. In order to prove the effectiveness of the proposed hybrid error control scheme, we adopt NIST-Net network emulator which is a general-purpose tool for emulating performance dynamics in IP networks. It is shown by simulations over the NIST-Net that the proposed hybrid error control scheme shows improved packet-loss resilience even with much less number of overhead packets compared to various conventional error control schemes.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.177-178
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• 2008

지상파 방송(DMB: Digital Multimedia Broadcasting)은 음성, 영상, 데이터와 같은 다양한 멀티미디어 신호를 디지털 방식으로 변조하여 이동 중에 방송을 청취할 수 있는 차세대 디지털방송 서비스이다. 그러나 지상파 DMB 전송 고도화망에서는 계층 변조(Hierarchical Modulation)전송 기법을 통하여 추가의 전송대역폭을 확보할 수 있다. 또한 스케일러블 비디오 코딩(Scalable Video Coding)부호화 방식을 이용하여 고전송효율/고품질의 이동 멀티미디어 방송서비스를 제공할 수 있는 고품질 AT(Advanced Terrestrial)-DMB 시스템이 가능하다. 이러한 고품질 AT-DMB의 개발에 있어서 여러 방식들이 제시됨에 따라 시뮬레이터를 통한 다중화 시스템의 분석이 필요하다. 본 논문에서는 고품질 AT-DMB가 가능한 스케일러블 비디오 방식을 JSVM8.8을 사용하여 구현하였으며, 다중화 시스템의 실험을 하였다. 또한 시뮬레이터를 통하여 복호된 계층 간의 화질 차이와 엔지니어를 위해 비트스트림의 분석화면 및 PSNR을 제공 하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.1C
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• pp.48-55
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• 2009

In scalable video coding, MCTF plays an important role for time-scalability and SNR-scalability. But there is image quality decreasing as MCTF level is increased because time interval of each frame is extended so that is hard to find suitable motion vector. In this paper, we propose an algorithm to prevent image quality from decreasing with unsuitable motion vector during MCTF update process using JND. We adapt JND to find errors within blocks of image and set a threshold which is used to add high frequency components during update process. We can overcome time-gap between frames and achieve better image quality through the proposed algorithm.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.1188-1191
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• 2011

IPTV 방송에서 인터넷 기반 멀티캐스트는 일대다 또는 다대다 통신을 위한 차세대 중요한 서비스로 주목 받고 있다. 멀티캐스트는 네트워크 또는 애플리케이션 레벨에서 서비스할 수 있다. IP 멀티캐스트는 소스노드에서 라우터로 데이터그램을 보내면 라우터가 이를 복제하여 수신노드들에게 전달해 주는 네트워크레벨 서비스로 네트워크 자원을 효율적 사용할 수 있다. 그러나 네트워크에 IP 멀티캐스트 라우터가 설치되어야 하는 등 여러 문제로 인해 널리 사용되지 못하고 있다. 따라서 대안으로 애플리케이션 레벨에서의 오버레이 멀티캐스트가 주목 받고 있다. 오버레이 멀티캐스트는 종단호스트가 라우터처럼 동작하는 것으로 비록 IP 멀티캐스트에 비해서 링크 사용율과 지연 값이 높아질 수 있지만, IP 멀티캐스트의 현실적인 적용의 어려움을 해결할 수 있는 장점을 가지고 있다. 본 논문에서는 IP 멀티캐스트가 제공되지 않는 네트워크에서 효율적인 SVC(Scalable Video Coding) 멀티미디어 방송서비스와 SSENS(SVC Stream Exchange Network Server) 라우팅을 위한 MST(Minimum Spanning tree)를 목적으로 하는 오버레이 멀티캐스트 트리를 구현하는 알고리즘을 설계한다. 제안된 알고리즘의 성능 분석을 위해 시뮬레이션을 통해 패킷 손실 측면에서 Prim 알고리즘에 비해 평균 패킷 손실율이 40% 가까이 향상됨을 증명하였다.

• Journal of KIISE:Information Networking
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• v.35 no.6
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• pp.549-556
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• 2008

To achieve flexible visual content adaption for multimedia communications, the ISO/IEC MPEG & ITU-T VCEG form the JVT to develop SVC amendment for the H.264/AVC standard. JVT uses inter-layer prediction as well as inter prediction and intra prediction that are provided in H.264/AVC to remove the redundancy among layers. The main goal consists of designing inter-layer prediction tools that enable the usage of as much as possible base layer information to improve the rate-distortion efficiency of the enhancement layer. But inter layer prediction causes the computational complexity to be increased. In this paper, we proposed an efficient residual prediction. In order to reduce the computational complexity while maintaining the high coding efficiency. The proposed residual prediction uses modified interpolation that is defined in H.264/AVC SVC.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.860-872
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• 2018

In this Paper, we propose 8K UHD (Ultra High Definition) encapsulation method for efficient transmission and reception based on MMT (MPEG Media Transport). Broadcasting services for 8K UHD allow users to feel the maximized reality. However, present technology is difficult to provide 8K UHD in broadcasting networks, because the 8K UHD bitrate is too high to be transmitted in the current broadcasting networks. Research for transmitting 8K UHD is underway. In some researches, a receiver is implemented with four 4K UHD display instead of a 8K UHD display. In order to transmit 8K UHD within the limited transmission bitrate of broadcasting network, 8K UHD contents encoded by SHVC (Scalable High Efficiency Video Coding) and then transmitted over heterogeneous network. For using the broadcasting and communication networks, MMT standard is used. MMT is IP based transmission protocol as the next generation transmission protocol. According to the MMT standard, video stream encapsulated and transmitted in MMTP (MMT Protocol) packet. IP-based broadcasting and communication networks can be used to transmit simultaneously, and the receiver can synchronize and play it. We propose an encapsulation method that can efficiently transmit and receive 8K UHD. The proposed method increases a payload rate and decreases an initial delay at the receiver. We show that the efficiency of the proposed method is verified by experimental tests.

• Journal of Broadcast Engineering
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• v.20 no.2
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• pp.310-339
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• 2015

In this paper, the possibility of a terrestrial fixed 4K UHD (Ultra High Definition) and mobile HD (High Definition) convergence broadcasting service through a single channel employing the FEF (Future Extension Frame) multiplexing technique in DVB (Digital Video Broadcasting)-T2 (Second Generation Terrestrial) systems is examined. The performance of such a service is also investigated. FEF multiplexing technology can be used to adjust the FFT (fast Fourier transform) and CP (cyclic prefix) size for each layer, whereas M-PLP (Multiple-Physical Layer Pipe) multiplexing technology in DVB-T2 systems cannot. The convergence broadcasting service scenario, which can provide fixed 4K UHD and mobile HD broadcasting through a single terrestrial channel, is described, and transmission requirements of the SHVC (Scalable High Efficiency Video Coding) technique are predicted. A convergence broadcasting transmission system structure is described by employing FEF and transmission technologies in DVB-T2 systems. Optimized transmission parameters are drawn to transmit 4K UHD and HD convergence broadcasting by employing a convergence broadcasting transmission structure, and the reception performance of the optimized transmission parameters under AWGN (additive white Gaussian noise), static Brazil-D, and time-varying TU (Typical Urban)-6 channels is examined using computer simulations to find the TOV (threshold of visibility). From the results, for the 6 and 8 MHz bandwidths, reliable reception of both fixed 4K UHD and mobile HD layer data can be achieved under a static fixed and very fast fading multipath channel.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.939-945
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• 2008

Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.