• Information and Communications Magazine
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• v.30 no.5
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• pp.51-58
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• 2013

본 논문에서는 차세대 지상파 DMB 방송 기술인 AT-DMB(Advanced Terrestrial Digital Multimedia Broadcasting) 기술의 내용과 AT-DMB 기반의 지역한정 데이터방송 서비스 기술에 대해 기술하고, 실제 필드에서 수행된 실험방송 내용에 대해서도 소개한다. AT-DMB 기술은 기존의 T-DMB 방송과 역호환성(backward compatibility)을 유지하면서 전송용량을 T-DMB 대비 최대 2배까지 증대시킨 기술이다. 추가확보된 전송용량에 새로운 채널을 할당하거나 MPEG-4 SVC(Scalable Video Coding) 기술을 이용하여 기존 T-DMB의 QVGA(320x240) 화질 대신 VGA(640x480)급의 고화질을 제공할 수 있다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.02a
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• pp.159-162
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• 2008

이 연구는 Ka대역 위성방송 서비스를 제공할 때 발생하는 강우감쇠문제를 해결하기 위하여 MPEG-4 SVC 신호를 이용하는 방안에 대하여 고찰하였다. Ka대역 위성방송시스템은 DVB-S2 VCM 모드를 고려하였으며, JSCC (Joint Source Channel Coding) 기법을 이용하여, SVC신호를 Ka대역 위성방송시스템에 적용하였다. SVC신호는 Spatial Scalability, SNR Scalability 및 Temporal Scalability로 구분되어서, PSNR값의 변화에 따른 비트율 변화정도를 분석하였다. 비트율 변화율이 가장 큰 Spatial Scalability를 적용한 SVC신호가 Ka대역 위성방송서비스의 강우감쇠 문제를 해결하기 위한 방안으로 제안되었으며, 이에 대한 분석이 수행되었다.

• 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 the Institute of Electronics Engineers of Korea CI
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• v.43 no.6 s.312
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• pp.60-70
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• 2006

Currently, Scalable Video Coding (SVC) is being standardized. By using scalability of SVC, QoS managed video streaming service is enabled in heterogeneous networks even with only one original bitstream. But current SVC is insufficient to dynamic video conversion for the scalability, thereby the adaptation of bitrate to meet a fluctuating network condition is limited. In this paper, we propose dynamic full-scalability conversion method for QoS adaptive video streaming in H.264/AVC SVC. To accomplish full scalability dynamic conversion, we propose corresponding bitstream extraction, encoding and decoding schemes. On the encoder, we newly insert the IDR NAL to solve the problems of spatial scalability conversion. On the extractor, we analyze the SVC bitstream to get the information which enable dynamic extraction. By using this information, real time extraction is achieved. Finally, we develop the decoder so that it can manage changing bitrate to support real time full-scalability. The experimental results showed that dynamic full-scalability conversion was verified and it was necessary for time varying network condition.

• Journal of Broadcast Engineering
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• v.12 no.6
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• pp.596-610
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• 2007

In order to reduce the complexity of SVC encoding, we introduce a fast mode decision method in the enhancement layers of spatial scalability by selectively performing the inter-layer residual prediction of SVC. The Inter-layer residual prediction coding in Scalable Video Coding has a large advantage of enhancing the coding efficiency since it utilizes the correlation between two residuals from a lower spatial layer and its next higher spatial layer. However, this entails the dramatical increase in the complexity of SVC encoders. The proposed method is to analyze the characteristics of integer transform coefficients for the subtracted signal for two residuals from lower and upper spatial layers. Then it selectively performs the inter-layer residual prediction coding and rate-distortion optimizations in the upper spatial enhancement layer if the SAD values of residuals exceed adaptive threshold values. Therefore, by classifying the residuals according to the properties of integer-transform coefficients only with SAD of residuals between two layers, the SVC encoder can perform the inter-layer residual coding selectively, thus significantly reducing the total required encoding time. The proposed method results in reduction of the total encoding time with 51.5% in average while maintaining the RD performance with negligible amounts of quality degradation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11B
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• pp.1151-1168
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• 2009

Improving packet loss does not necessarily coincide with the improvement in user perceivable QoS because each frame carries different degree of importance. We propose Significance-aware packet scheduling (SAPS) to maximize user perceivable QoS. SAPS carries out two fundamental issues of packet scheduling: "What to transmit" and "When to transmit?" To adapt to the available bandwidth, it is necessarily to transmit the subset of the data packets if the entire set of packets can not be transmitted. "Packet Significance" quantifies the importance of the frame by elaborately incorporating frames' dependency. Greedy approach is used in selecting packets and transmission schedule is determined based on the Packet Significance. The proposed scheme is tested using publicly available MPEG-4 video clips. Decoding engine is embedded in the simulation software and user perceivable QoS is exposeed in termstermiSNR. Throughout the simulation based experiment, the performance of the proposed scheme is compared two other schemes: Size-based packet scheduling and Bit-rate based best effort packet scheduling. SAPS successfully incorporates the semantics of a packet and improves user perceivable QoS significantly. It successfully provides unequal protection to more important packets.

• 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.

• 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.