• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2021.05a
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• pp.351-352
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• 2021

Due to the recent rapid change in the social environment due to Corona 19, the need for non-face-to-face/contact-based information exchange technology is rapidly emerging. Due to these changes, the development of an alternative system using a sense of immersion and a sense of presence is urgently required. In this study, in order to implement a video conferencing system, we implemented a technology for transmitting large-capacity 3D data in real time without delay. For this, the applied algorithm of GAN, the latest deep learning algorithm of the unsupervised learning series, was used.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.21 no.2
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• pp.1-7
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• 2020

In modern society, as technology has advanced and human life area has expanded, there has been an increasing demand for high-quality voice and video communications services without restrictions on time and place. In response to this demand, satellite communications systems that provide a wide range of communications and that offer multiple access are evolving day by day. In satellite communications systems such as Digital Video Broadcasting - Return Channel Via Satellite (DVB-RCS) and Warfighter Information Network-Tactical (WIN-T), the multi-frequency time division multiple access (MF-TDMA) demand assigned multiple access (DAMA) scheme is used for efficient resource allocation. In this scheme, since the satellite terminals periodically request resources from the network controller, and the network controller dynamically allocates resources, it is necessary to arrange resource allocation information from time to time. Shortening of the alignment time is a more important factor in a satellite communications system in which a long transmission delay occurs due to long-distance transmission and reception. In this paper, we propose a sorting algorithm variable-selection scheme that shortens the sorting time by cross-selecting the sorting algorithm based on a threshold value, while setting the number of frames in the MF-TDMA DAMA satellite communications system as the threshold value.

• Journal of Korea Multimedia Society
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• v.7 no.4
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• pp.518-531
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• 2004

This paper discusses a testbed architecture for implementing scalable service differentiation in the Internet. The differentiated services (DiffServ) testbed architecture is based on a model in which a bandwidth broker (BB) can control network resources, and the ALTQ can reserve resources in a router to guarantee a Quality of Service (QoS) for incoming traffic to the testbed. The reservation and releasemessage for the ALTQ is contingent upon a decision message in the BE. The BB has all the information in advance, which is required for a decision message, in the form of PIB. A signaling protocol between the BB and the routers is the COPS protocol proposed at the IETF. In terms of service differentiation, a user should make an SLA in advance, and reserve required bandwidth through an RAR procedure. The SLA and RAR message between a user and the BB has implemented with the COPS extension which was used between a router and the BB. We evaluates the service differentiation for the video streaming in that the EF class traffic shows superb performance than the BE class traffic where is a network congestion. We also present the differentiated service showing a better packet receiving rate, low packet loss, and low delay for the EF class video service.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1742-1754
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• 1999

In this paper, we propose a rate control scheme, using histogram based rate-distortion (R-D) estimation, which produces a consistent picture quality between consecutive frames. The histogram based R-D estimation used in our rate control scheme offers a closed-form mathematical model that enable us to predict the bits and the distortion generated from an encoded frame at a given quantization parameter (QP) and vice versa. The most attractive feature of the R-D estimation is low complexity of computing the R-D data because its major operation is just to obtain a histogram or weighted histogram of DCT coefficients from an input picture. Furthermore, it is accurate enough to be applied to the practical video coding. Therefore, the proposed rate control scheme using this R-D estimation model is appropriate for the applications requiring low delay and low complexity, and controls the output bit-rate ad quality accurately. Our rate control scheme ensures that the video buffer do not underflow and overflow by satisfying the buffer constraint and, additionally, prevents quality difference between consecutive frames from exceeding certain level by adopting the distortion constraint. In addition, a consistent considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K=10, considering CLP=$10^{-3}$ as a criterion, it is observed that the performance improment of about 3.5 dB and 1.5 dB is obtained, respectively, in terms of $E_b$/$N_o$ by employing the concatenated FEC code with pilot symbols.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.1 s.307
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• pp.27-38
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• 2006

The main purpose of rate control is to achieve the highest video quality when bandwidth or storage capacity is limited. For this purpose, we need a rate control algorithm which is adaptively controlled by the motion information of sequences, scene change, buffer capacity and time-varing bandwitdh channels. A rate-control method in the encoder requires the accurate estimation of target bit for each frame and the low end-to-end delay for transmitting video data by intelligent selection of encoding parameters. In this paper, we suggest three kinds of linear relation in the encoder to satisfy the characteristics of rate control. The first relation is that between the percentage of zero quantized transformed coefficients(p) and coded bits. Second relation is that between the PSNR of encoded frame and its Quantization parameter(QP). Finally, we can find out a linear approximation between QP and p. According to the experimental analysis, the proposed method results in an efficient rate control in terms of the bit estimation, the buffer capacity, and PSNR compared with the existing rate control in the H.264 JM 9.3.

• 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.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.210-216
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• 2014

In this paper, it was proposed to develop front and rear image monitoring system for vehicle that help a driver to cope with urgent situation about a dangerous element. When parking a vehicle, the risk factors to be formed by the dead zone can be resolved by using anterior and posterior cameras of the vehicle. In embedded system environment, a SoC(System on Chip) and two high-resolution CMOS (Complementary metal-oxide-semiconductor) image sensors were used to transfer two high-resolution image data through he TCP/ IP-based network. To transfer image data through he TCP/ IP-based network, the images received by two cameras were compressed by using H.264 and they were transmitted with wireless method(Wi-Fi) by using real-time transport protocol (Real-time Transport Protocol). Transmission loss, transmission delay and transmission limit were solved in wireless (Wi-Fi) environment and the bit-rate of two image data compressed by H.264 was adjusted. And the system for the optimal transmission in wireless (Wi-Fi) environment was materialized and experimented.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of the Computational Structural Engineering Institute of Korea
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• v.31 no.5
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• pp.227-234
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• 2018

Researches regarding measurement and control of the dynamic behavior of liquid such as sloshing have been actively on undertaken in various engineering fields. Liquid vibration is being measured in the study of tuned liquid dampers(TLDs), which attenuates wind motion of buildings even in building structures. To overcome the limitations of existing wave height measurement sensors, a method of measuring liquid vibration in a TLD using a laser Doppler vibrometer(LDV) and galvanometer scanner is proposed in this paper: the principle of measuring speed and displacement is discussed; a system of multi-point measurement with a single point of LDV according to the operating principles of the galvanometer scanner is established. 4-point liquid vibration on the TLD is measured, and the time domain data of each point is compared with the conventional video sensing data. It was confirmed that the waveform is transformed into the traveling wave and the standing wave. In addition, the data with measurement delay are cross-correlated to perform singular value decomposition. The natural frequencies and mode shapes are compared using theoretical and video sensing results.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.1
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• pp.178-184
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• 2015

This paper proposes a hardware architecture for high performance Deblocking filter(DBF) in High Efficiency Video Coding for UHD(Ultra High Definition) videos. This proposed hardware architecture which has less processing time has a 4-stage pipelined architecture with two filters and parallel boundary strength module. Also, the proposed filter can be used in low-voltage design by using clock gating architecture in 4-stage pipeline. The segmented memory architecture solves the hazard issue that arises when single port SRAM is accessed. The proposed order of filtering shortens the delay time that arises when storing data into the single port SRAM at the pre-processing stage. The DBF hardware proposed in this paper was designed with Verilog HDL, and was implemented with 22k logic gates as a result of synthesis using TSMC 0.18um CMOS standard cell library. Furthermore, the dynamic frequency can process UHD 8k($7680{\times}4320$) samples@60fps using a frequency of 150MHz with an 8K resolution and maximum dynamic frequency is 285MHz. Result from analysis shows that the proposed DBF hardware architecture operation cycle for one process coding unit has improved by 32% over the previous one.