• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.751-759
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• 2009

Due to the advance of WLAN technology, the use of the multimedia service such as the video streaming service has been increased in the home network. However, we need to study the method which decreases the transmission delay and the frame loss rate to provide QoS of the video streaming service. Therefore, this paper proposes an active buffer management mechanism to guarantee QoS of the streaming service in IEEE 802.11e EDCA. The proposed protocol discards the frame in the HoL of the buffer based on the importance of each frame and the virtual transmission delay of frame newly arriving at the buffer. In the simulation results, the proposed algorithm not only decreases the frame loss probability of important I and P frames but also stabilizes the transmission delay. It may increase the QoS of video streaming services.

• Transactions on Electrical and Electronic Materials
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• v.10 no.6
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• pp.222-226
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• 2009

The objective of this study is to share multimedia contents included in existing digital devices and to solve the problems of an increase in installation fees and non-environmentally friendly interiors. This study designed a new digital signal transmitter and receiver using power line transmission and HDMI in order to solve the problems in the existing systems. The transmitter and receiver designed in this study used an AD9867BCPZ PLC chip in which the transmission came from digital signals originating in a PC, and the system architecture was configured so that the outputs signals were connected to a TV from the receiver. The experiment was implemented by adding a Video Test Generator, a USBPre external sound card, and Smaart Live 6 for analyzing the characteristics of the configured system. In the video test results, it was verified that communication was actively implemented, and the image quality showed a constant level from the measurement of the captured video. In the case of the sound, it was verified that more than 90% of the sound signals were normally transmitted and received from the examination of their phases and magnitudes. Thus, the performance of the system designed in this study was verified, which leads to the resolution of some of the problems found in current digital devices.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.3
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• pp.345-353
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• 2011

An 1.485 Gbit/s video signal transmission system using the carrier frequency of H-band(220~325 GHz) was implemented and demonstrated for the first in domestic. The RF front-end was composed of Schottky barrier diode sub-harmonic mixers(SHM) and frequency triplers, and diagonal horn antennas for transmitter and receiver, respectively. The transmitted carrier frequency of 246 GHz was implemented in the H-band, and LO frequencies of H-band SHM is 120 GHz and 126 GHz for transmit and receive chains, respectively. The modulation scheme is ASK(Amplitude Shift Keying) where IF frequency is 5.94 GHz and the envelop detection was used in heterodyne receiver architecture, and direct detection receiver using ZBD(Zero Bias Detector) was implemented as well. The 1.485 Gbit/s video signal with HD-SDI format was successfully transmitted over wireless link distance of 5 m and displayed on HDTV at the transmitted average output power of 20 ${\mu}W$.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.4092-4104
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• 2017

In this paper, an efficient 4K and 8K UHD(Ultra High Definition) transmission scheme is proposed on the convergence networks with broadcasting and LTE(Long Term Evolution) by using CoMP(Coordinated Multi-Point). A video data is compressed and divided into BL(Base Layer), E(Enhanced layer)1, E2 and E3 by scalable HEVC(High Efficiency Video Coding). The divided layers can be combined by the scalable HEVC such as mobile HD, full HD, 4K and 8K UHD(Ultra High Definition). The divided layers are transmitted through the convergence networks with DVB-T2(Digital Video Broadcasting-$2^{nd}$ Generation Terrestrial) broadcasting system and LTE CoMP. This scheme transmits mobile HD and full HD layers through DVB-T2 broadcasting system by using M-PLP(Multiple-physical Layer Pipes), and adaptively transmits 4K or 8K UHD layer through LTE CoMP with MMT(MPEG Media Transport) server. An adaptive transmitting and receiving scheme in the LTE CoMP system provides 4K or 8K UHD layer to a user according to the user status. The proposed scheme is verified by showing the system-level simulation results which is better BER(bit-error-rate) performance than the conventional scheme. The results show that the proposed scheme provides the stable video contents to the user especially at the cell edge.

• Journal of KIISE
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• v.41 no.10
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• pp.824-831
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• 2014

HTTP (Hypertext Transfer Protocol) Adaptive Streaming is gaining attention because it changes bitrates to adapt changing network conditions. Since HAS (HTTP Adaptive Streaming) client downloads the video data based on TCP (Transmission Control Protocol), it estimates incorrectly the available bandwidth and leads to an unnecessary video quality change in long-delay networks. In this paper, we propose a buffer-based quality control scheme in order to improve the service quality and smooth playback in the HAS. The proposed scheme estimates accurately the available bandwidth based on a modified streaming model that considers network delay. It also calculates the sustainability of the video quality to prevent an unnecessary quality change and determines the inter-request time on the basis of the buffer status. Through the simulation, we prove that our scheme improves the QoS (Quality of Service) of the HAS service and controls the video quality smoothly in long-delay networks.

• Journal of Broadcast Engineering
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• v.11 no.4 s.33
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• pp.541-553
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• 2006

Since DMB on-air was started, not high-qualified audio, video services but various service models have been required. This paper is about systems for one of these services, video associated data service. A terrestrial DMB system to make contents of video associated data services and transmit them on DMB channel is proposed in this paper. This system satisfies standard of the video associated data services for terrestrial DMB; MPEG-4 BIFS (BInary Format for Scene) Core2D scene description profile and graphics profile. This system is designed to support two major features of broadcasting, real-time authoring non automatic transmission and non real-time authoring automatic transmission, and focuses on the abilities to make high-qualified contents efficiently and transmit them to video encoder reliably. This system proved its performance through conformance tests with various receivers, so can be used in future on-air.

• Journal of Korea Multimedia Society
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• v.13 no.3
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• pp.349-358
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• 2010

Adaptability is an important issue in video streaming over mobile environments, since the clients may request videos with great differences in their workload. In this paper, we propose the issues in limited bandwidth scheduling for efficient MPEG-4 video stream transmission over a mobile or wireless network. In the phase of admission control, the amount of bandwidth allocated to serve a video request is the mean bandwidth requirement of its requested video. The dynamic allocation of bandwidth in the phase of scheduling depends on the playback buffer levels of the clients with an objective to make it more adaptive to the playback situation of individual clients. In the proposed RTA scheduling protocol, more bandwidth may be allocated temporarily to the client whose buffer level is low. By employing the buffer level based scheduling policy, this protocol attempts to maximize the real-time performance of individual playback while minimizing the impact of transient overloading. Extensive simulation experiments have been performed to investigate the performance characteristics of the RTA protocol as comparing with BSBA protocol. This RTA protocol shows the better performance by transferring more frames than BSBA protocol.Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• Journal of Broadcast Engineering
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• v.19 no.1
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• pp.14-23
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• 2014

This paper proposed controlling methods for SDN(Software Defined Network) based multiple radio access technology as the solutions of following two issues which were mainly occurred by explosive increasing of video traffic. The first one is a requirement for traffic off-loading caused by 3rd-party video service providers from the mobile network operator's viewpoint. The other one is a provision of high-speed video contents transmission services with low price. Furthermore, the performance evaluation was also conducted on the real test-bed which is composed of OpenStack cloud and SDN technology such as OpenFlow and Open vSwitch. A virtual machine running on the OpenStack provide a video service and the terminal which is able to use multiple radio access technology supports two 2.4GHz WLANs(Wireless Local Area Network) and three 5GHz WLANs, concurrently. Finally, we can get 820Mbps of the maximum transmission speed by using that five WLAN links for the single service at the same time.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.31-40
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• 2005

In this paper, we develop an approach toward JSC(Joint Source-Channel Coding) method for MPEG-4 based FGS(Fine Granular Scalability) video coding and transmission in fixed and mobile receiving environment(Digital Audio Broadcasting, DAB). The source coder used MPEG-4 FGS video codec, the channel coder used RCPC(Rate Compatible Punctured Convolution) code and the modulation method used QPSK modulation. We have considered channel environment of AWGN and mobile receiving environment. This study determined optimum Trade-off point between source bit rate and channel coding rate in variable channel states. We compared FGS-JSC method and general single layer CBR(Constant Bit Rate) transmission. In this results, FGS-JSC was appeared better performance than CBR transmission.

• Journal of Internet of Things and Convergence
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• v.8 no.1
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• pp.53-58
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• 2022

Smoothing is a transmission plan that converts video data stored at a variable bit rate into a constant bit rate. In the study of [6-7], when a data rate increase is required, the frame with the smallest increase is set as the start frame of the next transmission rate section, when a data tate decrease is required. the frame with the largest decrease is set as the start frame of the next transmission rate section, And the smoothing algorithm was proposed and performance was evaluated in an environment where network traffic is not considered. In this paper, the smoothing algorithm of [6-7] evaluates the adaptive CBA algorithm and performance with minimum frame rate, average frame rate, and frame rate variation from 512KB to 32MB with E.T 90 video data in an environment that considers network traffic. As a result of comparison, the smoothing algorithm of [6-7] showed superiority in the comparison of the minimum refresh rate.