• Nuclear Engineering and Technology
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• 제52권10호
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• pp.2339-2345
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• 2020

Radiation imaging systems consisting of a large number of channels greatly benefit from multiplexing methods to reduce the number of channels with minimizing the system complexity and development cost. In conventional pixelated radiation detector modules, such as anger logic, is used to reduce a large number of channels that transmit signals to a data acquisition system. However, these methods have limitations of electrical noise and distortion at the detector edge. To solve these problems, a multiplexing concept using a digital signal encoding technique based on a time delay method for signals from detectors was developed in this study. The digital encoding multiplexing (DEM) method was developed based on the time-over-threshold (ToT) method to provide more information including the activation time, position, and energy in one-bit line. This is the major advantage of the DEM method as compared with the traditional ToT method providing only energy information. The energy was measured and calibrated by the ToT method. The energy resolution and coincidence time resolution were observed as 16% and 2.4 ns, respectively, with DEM. The position was successfully distributed on each channel. This study demonstrated the feasibility that DEM was useful to reduce the number of detector channels.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• 인터넷정보학회논문지
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• 제12권4호
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• pp.45-50
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• 2011

본 논문에서는 차량용 다중 카메라시스템에서 무선영상을 전송하기 위해 FH/TDD(Frequency Hopping/Time Division Duplex) 기반의 Motion JPEG 영상압축 코덱의 부호화 지연시간을 분석한다. 또한, Motion JPEG기반으로 FH/TDD에 의한 다중접속을 위해 시분할 시에 상호 동기화된 채널이동 과 접근방식(Synchronized shift & access)이 가능하도록 채널의 상태를 파악한 후, 채널충돌(Channel Collision)을 최소화하기 위한 동기화 연결(Synchronized connection) 방식을 설계한다.

• ETRI Journal
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• 제39권4호
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• pp.582-591
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• 2017

To address the wire complexity problem in large-scale globally asynchronous, locally synchronous systems, a current-mode ternary encoding scheme was devised for a two-phase asynchronous protocol. However, for data transmission through a very long wire, few studies have been conducted on reducing the long propagation delay in current-mode circuits. Hence, this paper proposes a current steering logic (CSL) that is able to minimize the long delay for the devised current-mode ternary encoding scheme. The CSL creates pulse signals that charge or discharge the output signal in advance for a short period of time, and as a result, helps prevent a slack in the current signals. The encoder and decoder circuits employing the CSL are implemented using $0.25-{\mu}m$ CMOS technology. The results of an HSPICE simulation show that the normal and optimal mode operations of the CSL achieve a delay reduction of 11.8% and 28.1%, respectively, when compared to the original scheme for a 10-mm wire. They also reduce the power-delay product by 9.6% and 22.5%, respectively, at a data rate of 100 Mb/s for the same wire length.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제13권1호
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• pp.291-304
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• 2019

This paper proposes a dynamic-tiling-based bandwidth-efficient (DTBE) virtual reality (VR) streaming scheme. We consider 360-degree VR contents with multiple view points such as the front, back, upper, and bottom sides. At a given time, the focus of a client is always bound to a certain view among multiple view points. By utilizing this perspective, under our proposed scheme, tiles with high encoding rates are selectively assigned to the focused view where multiple view points consist of multiple tiles with different encoding rates. The other tiles with low encoding rates are assigned to the remaining view points. Furthermore, for reducing view adaptation delay, we design a novel rapid view adaptation mechanism that selectively delivers an I-frame during view point updates by using frame indexing. We implement the proposed scheme on a commercial VR test bed where we adopt the MPEG media transport (MMT) standard with high-efficiency video coding (HEVC) tile modes. The measurement-based experiments show that the proposed scheme achieves an average data usage reduction of almost 65.2% as well as average view adaptation delay reduction of almost 57.7%.

• 한국통신학회논문지
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• 제34권5A호
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• pp.355-362
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• 2009

In real-time communication services, delay constraints are among the most important QoS (Quality of Service) factors. In particular, it is difficult to guarantee the delay requirement over wireless channels, since they exhibit dynamic time-varying behavior and even severe burst-errors during periods of deep fading. Channel throughput may be increased, but at the cost of the additional delays when ARQ (Automatic Repeat Request) schemes are used. For real-time communication services, it is very essential to predict data deliverability. This paper derives the delay distribution and the successful delivery probability within a given delay budget using a priori channel model and a posteriori information from the perspective of queueing theory. The Gilbert-Elliot burst-noise channel is employed as an a Priori channel model, where a two-state Markov-modulated Bernoulli process $(MMBP_2)$ is used. for a posteriori information, the channel parameters, the queue-length and the initial channel state are assumed to be given. The numerical derivation is verified and analyzed via Monte Carlo simulations. This numerical derivation is then applied to a rate control scheme for real-time video transmission, where an optimal encoding rate is determined based on the future channel capacity and the distortion of the reconstructed pictures.

• 정보처리학회논문지C
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• 제19C권4호
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• pp.247-252
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• 2012

네트워크 코딩(Network Coding)은 통신망의 성능 향상에 도움을 줄 수 있으나 이의 소프트웨어적 구현은 부호화/복호화 단계에서 매우 큰 지연시간을 유발할 수 있어 이를 줄일 수 있는 병렬화된 구현이 필수적이라 할 수 있다. 본 논문에서는 랜덤 리니어 네트워크 코딩(Random Linear Network Coding)과 랜덤 리니어 네트워크 코딩의 단점을 보완하고자 최근 제안된 파이프라인 네트워크 코딩(Pipeline Network Coding)의 병렬처리 성능을 비교한다. 또한, 네트워크 코딩의 CPU에서의 병렬처리 기법과 GPGPU(General Purpose Graphics Processing Units)에서의 병렬처리 기법을 비교하여 네트워크 코딩의 사용 시 그 파라미터에 따라 적절한 병렬처리 기법을 선택할 필요성이 있음을 보여준다.

• Journal of information and communication convergence engineering
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• 제5권1호
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• pp.54-58
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• 2007

A real-time frame-layer rate control algorithm with a token bucket traffic shaper is proposed for distortion variation minimization. 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. Experimental results indicate that the proposed control method provides better visual and PSNR performances than the existing rate control method.

• 한국멀티미디어학회논문지
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• 제10권8호
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• pp.1039-1051
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• 2007

본 논문은 비디오통신 시스템에서 초저지연을 달성하면서 립싱크 제어하는 방법을 제안한다. 초저지연 비디오 통신에서 핵심적인 기술은 종단간 지연시간을 줄이는 기술과 립싱크 제어 기술이다. 특히 서비스관점에서 립싱크 제어 기술이 중요한 요인으로 작용하고 있다. 오디오와 비디오의 데이터를 RTP/RTCP 기반으로 패킷을 구성하여 전송하고, 이 패킷을 이용하여 오디오와 비디오의 재생시간을 계산한 후 립싱크 제어를 한다. 본 논문에서는 오디오 데이터가 일정한 간격으로 재생되도록 하고, 오디오가 재생되는 시점에서 가장 근접한 재생시간을 가진 비디오 데이터를 찾아서 재생하는 방법으로 오디오와 비디오간의 립싱크 제어하는 방법을 제안한다. 그리고 종단간 지연시간이 100 ms이하인 초저지연 비디오 통신을 하기 위해서는 송신단의 인코딩 버퍼 제거하여 지연시간을 줄이고, 수신단의 재정렬버퍼 (Reordering Buffer)와 립싱크 버퍼의 크기를 3 프레임으로 처리하여 종단간 지연시간을 최소로 하였다. 실험결과에서 종단간 지연시간이 100 ms이하를 유지하고 오디오와 비디오의 립싱크 제어를 하였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2001년도 하계학술대회 논문집 D
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• pp.2369-2373
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• 2001

Usually the static scheduling algorithms such as DMS (Deadline Monotonic Scheduling) or RMS(Rate Monotonic Scheduling) are used for CAN scheduling due to its ease with implementation. However, due to their inherently low utilization of network media, some dynamic scheduling approaches have been studied to enhance the utilization. In case of dynamic scheduling algorithms, two considerations are needed. The one is a priority inversion due to rough deadline encoding into stricted arbitration fields of CAN. The other is an arbitration delay due to the non-preemptive feature of CAN. In this paper, an extended algorithm is proposed from an existing EDS(Earliest Deadline Scheduling) approach of CAN scheduling algorithm haying a solution to the priority inversion. In the proposed algorithm, the available bandwidth of network media can be checked dynamically by all nodes. Through the algorithm, arbitration delay causing the miss of their deadline can be avoided in advance. Also non real-time messages can be processed with their bandwidth allocation. The proposed algorithm can achieve full network utilization and enhance aperiodic responsiveness, still guaranteeing the transmission of periodic messages.