• Nuclear Engineering and Technology
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• v.52 no.10
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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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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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.

• Journal of Internet Computing and Services
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• v.12 no.4
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• pp.45-50
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• 2011

In this paper, the encoding time delay of FH/TDD(Frequency Hopping/Time Division Duplex) based Motion JPEG image compression CODEC is analyzed for radio video transmissions of multi-camera systems in a vehicle. And, Synchronized connection of minimum channel collision is designed with synchronized shift and access according to channel status for Motion JPEG based FH/TDD multiple access.

• ETRI Journal
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• v.39 no.4
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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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• v.13 no.1
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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%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5A
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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.

• The KIPS Transactions:PartC
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• v.19C no.4
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• pp.247-252
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• 2012

Network coding has been shown to improve various performance metrics in network systems. However, if network coding is implemented as software a huge time delay may be incurred at encoding/decoding stage so it is imperative for network coding to be parallelized to reduce time delay when encoding/decoding. In this paper, we compare the performance of parallelized decoders for random linear network coding (RLC) and pipeline network coding (PNC), a recent development in order to alleviate problems of RLC. We also compare multi-threaded algorithms on multi-core CPUs and massively parallelized algorithms on GPGPU for PNC/RLC.

• Journal of information and communication convergence engineering
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• v.5 no.1
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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.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1039-1051
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• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.

• Proceedings of the KIEE Conference
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• 2001.07d
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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.