• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.9
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• pp.1947-1954
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• 2012

The first frame is encoded in intra mode which generates a larger number of bits. In addition, the first frame is used for the inter mode encoding of the following frames. Thus the initial QP for the first frame affects the first frame as well as the following frames. Traditionally, the initial QP is determined among four constant values only depending on the bpp. In the case of low bit rate video coding, the initial QP value is fixed to 40 regardless of the output bandwidth. Although this initialization scheme is simple, yet it is not accurate enough. An accurate initial QP prediction scheme should not only depends on bpp but also on the complexity of the video sequence and the output bandwidth. In the proposed scheme, we determine the initial QP according to the ratio of the first frame to the total bits allocated to a GOP. To estimate the QP of the allocated bits, Rate-QP model is used. It is shown by experimental results that the new algorithm can predict the optimal initial QP more accurately and generate the PSNR performance better than that of the existing JVT algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.12B
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• pp.1329-1335
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• 2009

In this paper, we propose a new medium access control protocol referred to as DCTW (Dual Channel Transmission Scheme for wireless body area networks). Wireless body area networks (WBANs) requires prioritization mechanism for life-critical data to transmit the data as early as possible. The proposed DCTW exploits a narrow band for transmitting life-critical data while it uses a broadband channel to transmit normal data. Since the narrow band is dedicated to life-critical data, the DCTW can effectively reduce the delay of life-critical data transmission. Through extensive simulation, we show the DCTW outperforms other existing schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2305-2312
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• 1998

This paper analyzed the delay characteristics of speech packets in virtual cellular network(VCN). The probability distribution of packet delay is obtained using the markov chain model when periodic speech packets are transmitted by slotted-ALOHA protocol. The effects of probility of capture and retransmission policy on the performance were also analyzed. At first, the probability cumulative function of packet delay is calculated from the probability of capture as a function of location of mobile terminal. In order to investigate the effects of backoff delay, we defined a parameter NPr, where N is the period (frame size) of the speech packets and Pr is the retransmission probability for each speech packet. We also obtained the 1% outage delay for various frame size N.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.7
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• pp.696-708
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• 1992

In this study, we have several objectives. First, In spite of a fixed bandwidth waste of reservation channel, we maximize the efficiency of transmission channel by the multiple access algorithm that performs channel reservation and data transmission independently, eliminating the frame structure of transmission channel and reserving it by the slot unit . Second, In order to improve the entire system performance, we accommodate the variation of traffic at each earth station more effectively, and accomplish the stable delay characteristics and the equlity of service #or users. For this purpose, we design the satellite channel that consists of reservation channel and transmission channel which are logically separate and operate Independently. We also design a demand assignment multiple access algorithm based on the satellite channel structure.

• Journal of Broadcast Engineering
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• v.23 no.5
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• pp.682-692
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• 2018

In this paper, we propose an ARQ packet error control scheme using multiple threads in delivering massive capacity of multimedia based on MMT(MPEG Media Transport) protocol. On the sending side, each frame that constitutes an image is packetized into MMT packets based on MMT protocol. The header of the packet stores the sequence number of the frames contained in the packet and the time of presentation information. The payload of the packet stores the direct information that comprises the frame. The generated MMT packet is transmitted to the IP network. The receiving side checks if any error has occurred in the received packet. For any identified error, it controls the error through ARQ error control scheme and reconfigure the frame according to the information stored in the header of the received packet. At this point, a multi-threading based transport design is constructed so that each thread takes over a single frame, which increases the transmission efficiency of massive capacity multimedia. The efficiency of the multi-threading transport method is verified by solving the problems that might arise when using a single-thread approach if packets with errors are retransmitted.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1889-1899
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• 2015

In this paper, we propose a multiple slot reservation scheme to transmit data rapidly in the satellite random access channel. In the R-CRDSA (Reservation scheme with Contention Resolution Diversity Slotted ALOHA), each satellite terminal can use only one slot in a frame. Therefore, many slots are wasting in the low traffic load and the satellite terminals which have large date needs many frame to transmit their data although there are unused slots. In the multiple slot reservation scheme, each satellite terminal transmits a packet with their data size and reserves many slots in the light of data size and slot reservation status. Therefore, each satellite terminal transmits their data faster than R-CRDSA. This is because they can reserve many slots in a frame. Furthermore, we simulate proposed scheme and validate the performance of proposed scheme.

• Journal of the Korea Society of Computer and Information
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• v.7 no.2
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• pp.95-100
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• 2002

By feedback channel, the decoder reports the addresses of corrupted macroblocks induced by transmission errors back to the encoder With these negative acknowledgements, the encoder can make the next frame having propagated errors by using forward dependency based on GOBs and MBs of the frame happening transmission errors. The encoder can precisely calculate and track the propagated errors by examining the backward motion dependency for each of four comer pixels in the current encoding frame until before-mentioned the next frame. The error-propagation effects can be terminated completely by INTRA refreshing the affected macroblocks. Such a fast algorithm further reduce the computation and memory requirements. The advantages of the low computation complexity and the low memory requirement are Particularly suitable for real-time implementation.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.11a
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• pp.63-65
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• 2016

DVB (Digital Video Broadcasting)-S2 (Satellite - Second Generation) 표준은 현재 위성방송 시스템으로 가장 많이 사용되고 있는 표준이나, 추가적인 성능향상과 보다 다양한 응용분야에 적용하기 위해 DVB-S2 기술을 확장한 DVB-S2x (Satellite - Second Generation Extension) 시스템이 제정되었다. 그 중에서도 612,540 심볼 길이의 수퍼프레임 (Super-frame) 구조를 선택적으로 도입하여 긴 데이터 길이에 대해 스크렘블링 (Scrambling)을 적용하는 방법과 PLS (Physical Layer Signaling) 코드와 같은 참조 데이터 필드를 반복적으로 사용하는 방법 등을 사용할 수 있도록 하였다. 이를 통해 동일채널 간섭 (co-channel interference)에 대한 강인성을 증가시키고, 매우 낮은 SNR (Signal to Noise Ratio) 환경에서의 수신기 성능 향상 효과를 제공하게 된다 본 논문에서는 버스트 슈퍼프레임 전송 기반의 DVB-S2x 수신기를 위한 동기부를 설계하고 구조를 제안한다. 슈퍼프레임의 포맷은 DVB-S2x Annex E 의 규격 중 2 번 포맷을 이용하였으며, 2 번 슈퍼프레임 포맷은 버스트 (burst) 기반의 전송 방식에 용이한 측면이 있다. 동기부는 크게 버스트 검출부, 주파수 복구부, 신호 이득 조절부 그리고 심벌 타이밍 복구부로 구성된다.

• Proceedings of the KAIS Fall Conference
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• 2005.05a
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• pp.214-216
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• 2005

본 논문에서는 가변형식메시지체계에서 COMSEC 프레임동기 정보의 전송영향을 분석하였다. 실험결과 비트오류율 $10^{-1}\~10^{-5}$ 환경에서 COMSEC 프레임 동기 정보의 강인한 특성을 고찰하였으며 암호통신을 위해 사용되는 프레임 동기정보가 동기검출 및 오검출 측면에서 영향정도를 분석하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.157-159
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• 2001

인터넷 화상전화는 저렴한 가격으로 통신이 가능하게 하는 기술로 최근에 주목을 받고 있다. 인터넷은 제한적인 대역폭과 가변적인 손실을 나타내므로, 인터넷을 통해 전송되는 화상은 오류 전파에 의해 품질이 크게 떨어지게 된다. 일반적으로 화상은 움직임 예측, 보상기법을 이용하여 압축되기 때문에 한 프레임에서 발생한 오류가 그 프레임을 참조하는 프레임으로 전파되는 데 이를 오류 전파라고 한다. 기존의 오류 전파를 방지하는 기법에는 Error Tracking, NEWPRED, PFEC등이 있다. 본 논문에서는 인터넷 화상 전화에서 화상의 움직임이 적은 특성을 고려하여 Motion Vector의 중복전송을 이용한 PRMS기법을 제안한다. PFEC에 비해 부호화 효율을 높이면서 MB단위로 참조 프레임을 선택하여 오류 전파를 효과적으로 방지하는 기법이다. 실험을 통하여 PRMS기법이 기존치 PFEC나 NEWPRED에 비해 성능이 우수함을 보인다.