• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.9-15
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• 2001

Very low bit rate video coding uses the inter-picture video coding method for high compression. The inter-picture video coding is coded based on the information of the previous frames so any packet loss can lead to reduce the image quality on the transmission. In this paper, we proposed the multi-stream generation method for inter-media synchronization of very low bit rate video based on TCP for reliable transmission. The proposed approach performs a reliable transmission via a TCP based protocol. This method incorporates multi-streams in order to enhance the robustness of delivery and can withstand against network jitter. Moreover, the client bandwidths are fully utilized in a highly efficient way.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.11
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• pp.9-14
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• 1999

In this paper, a method of one line transmission of 622Mbps to interface a low speed part with a high speed part is introduced instead of H-BUS method of the 10 line transmission of 77.76Mbps in the 10Gbps optical transmission system. For this method, a reframing method to align the received data of 622Mbps transmission to STM(Synchronous Transfer Mode)-64 frame of SDH(Synchronous Digital Hierarchy) is described. Reframing is designed with VHDL and applied in the 10G-S4 ASIC of T14U board of 10Gbps optical transmission system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.175-178
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• 2007

According to recently presented QoS architecture by 3GPP, a traffic conditioner may be deployed to provide conformance of the negotiated QoS. A real-time frame-layer rate control method which can be applied to the traffic conditioner is proposed. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. The proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.13 no.3
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• pp.38-46
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• 2014

Wi-Fi STAs in power saving mode described in the Wi-Fi specification can deteriorate the system performance such as throughput, time delay when they receive data frames from a Wi-Fi AP. This problem is caused by data frame reception failure due to the channel state, and it increases re-transmissions. In this paper, we propose a modified power saving mode of Wi-Fi system in order to solve the problem. Also, we evaluate its performance in terms of throughput and time delay by simulation.

• Proceedings of the Korean Information Science Society Conference
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• 2007.06b
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• pp.344-349
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• 2007

무선 통신 환경에서 실시간 화상 통신을 위해 제안된 3G-324M 프로토콜은 회선 교환 통신을 사용함으로 인해 제한된 대역폭을 이용해야만 하므로 전송 시 여러 가지 제약이 발생하게 된다. 특히 송수신의 스케줄링을 효율적으로 하지 못할 경우 H.223 프로토콜에서 매 타임 슬롯마다 전송 가능한 최대 크기의 데이터를 전송 버퍼에 채워주지 못해 전송 지연 시간이 발생하게 된다. 본 논문에서는 이러한 문제들을 해결하기 위한 통신 스케줄링 기법을 제시한다. 이 통신 스케줄링 기법은 화상 프레임의 종류에 따라 크기가 크게 변한다는 사실에 기초해 H.223 프로토콜의 실행 비율을 임시적으로 증가시켜 지연시간이 늘어나는 것을 최소화하고 이로 인해 내부 버퍼 사용량을 줄일 수 있도록 한다. 또 수신 버퍼에 처리해야 할 데이터가 많은 경우, 임시로 H.223 프로토콜의 실행 비율을 증가시켜 불필요한 수신 지연 시간이 발생하지 않도록 한다. 실험은 내부 버퍼의 사용량은 제안한 통신 스케줄링 기법이 다른 통신 스케줄링 기법들에 비해 효율적으로 관리 되며 패킷의 손실률, 수신 단말기에서의 지연이 줄어드는 것을 보여준다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.12
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• pp.108-116
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• 2009

In this paper, we propose a Ethernet based audio distribution system with a new L1/L2 hybrid switching scheme, and evaluate its performance. The proposed scheme not only offers guaranteed low latency and jitter characteristics that are essentially required for the distribution of high-quality uncompressed audio traffic, and but also provide an efficient transmission of data traffic on the Ethernet environment. The audio distribution system with a proposed scheme consists of a master node and a number of relay nodes, and all nodes are mutually connected as a daisy-chain topology through up and downlinks. The master node generates an audio frame for each cycle of 125us, and the audio frame has 24 time slotted audio channels for carrying stereo 24 channels of 16-bit PCM sampled audio. On receiving the audio frame from its upstream node via the downlink, each intermediate node inserts its audio traffic to the reserved time slot for itself, then relays again to next node through its physical layer(L1) transmission - repeating. After reaching the end node, the audio frame is loopbacked through the uplink. On repeating through the uplink, each node makes a copy of audio slot that node has to receive, then play the audio. When the audio transmission is completed, each node works as a normal L2 switch, thus data frames are switched during the remaining period. For supporting this L1/L2 hybrid switching capability, we insert a glue logic for parsing and multiplexing audio and data frames at MII(Media Independent Interlace) between the physical and data link layers. The proposed scheme can provide a good delay performance and transmission efficiency than legacy Ethernet based audio distribution systems. For verifying the feasibility of the proposed L1/L2 hybrid switching scheme, we use OMNeT++ as a simulation tool with various parameters. From the simulation results, one can find that the proposed scheme can provides outstanding characteristics in terms of both jitter characteristic for audio traffic and transmission efficiency of data traffics.

• Journal of KIISE:Software and Applications
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• v.29 no.6
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• pp.355-366
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• 2002

This paper introduces the MPEG-4 video frame-based bitrate control methodology using two dimensional history pool and a sliding window. Proposed method preferentially clusters the encoded results according to the image characteristics and stores those results into the 2 dimensional history pool. Among the stored results in the pool, the sliding window collects the encoded results whose characteristics are very similar to the image frame to be encoded. Feedback regression is finally carried out bated on the collected results. Therefore proposed method can actively adapt to the rapid varying image characteristics by reducing the occurrences of the extrapolations when determining the quantization steps. Proposed method has better performances than the MPEG-4 frame-based bitrate control algorithm by evaluating with the actually encoded bits per frame, encoded PSNR's, and frame skips.

• The KIPS Transactions:PartC
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• v.14C no.3 s.113
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• pp.293-304
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• 2007

In this paper, a client-side framework for processing of the profile that is necessary for providing adapted content to user's context in the client is designed and implemented. The profile must be constituted context information and various user's information for providing the adapted content to user's context. The client device also provides functionalities such as the creation, the management, and the transmission of the profile. The profile which is used in the proposed profile framework consists of various related information of a user for content adaptation. The technology such as creation, transmission and manage of the profile for effective processing is proposed and apply this technologies to client profile framework during the design are applied. As the result of evaluation, techniques of the proposed framework for processing profiles is more effective than previous techniques.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.2A
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• pp.139-147
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• 2008

DVB-S2 (Digital Video Broadcasting-Satellite, Version 2) system transmits frames which adapt their structures based on the channel conditions, thus requiring simultaneous detection of the start of the Same (SoF) and the frame structure at the initial acquisition stage of the receiver. Also, a very low value of the minimum operating signal-to-noise ratio (SNR) for the acquisition necessitates a method utilizing multiple received frames to meet the required performance. In this paper, performance of joint time synchronization and frame structure detection methods using multiple DVB-S2 frames is evaluated by deriving the detection error probability. In particular, we evaluate the performance and complexity variations when the soft- and hard-decision values of the signal correlation output are used, present the synchronization parameters to optimize the performance, and verify the analysis results via computer simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3A
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• pp.291-303
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• 2006

In this paper, we analyze functional requirements of the IEEE 802.22 WRAN, and propose a downlink 프레임 structure satisfying the requirements. The proposed downlink 프레임 structure maximizes e transmission efficiency by adopting the cognative radio to assign the sub-channel by reflecting the channel environment of WRAN. We also calculate the signalling overhead for both downlink and uplink, and analyze the performances of time synchronization, frequency synchronization and cell identification based on the 프리앰블 in downlink and suggest the channel estimation method tough 프리앰블 or pilot. As a final result, e stationary beamforming (SBF) algorithm with dynamic channel allocation(DCA) is proposed. The proposed OFDMA downlink 프레임 structure with channel adaptive sub-channel allocation for cognitive radio applications is verified to meet the requirements of IEEE 802.22 WRAN, by computer simulations.