• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.8
• /
• pp.1774-1781
• /
• 1997

In this paper, we propose a full digital frequency and phase locked loop for CATV and HDTV receivers adopting VSB modulation. The CATV and HDTV receivers proposed by the Grand-Alliance in USA are ultilizing analog signal processing technology for carrier recovery. By the way, it is not a good architecture for the development of single chip ASIC operating in digital domain. To solve this problem while improving the performance, we first down convert the received r.f. signal to a near baseband signal for a low-rate AD converter and then we use digital signal processing techniques. The proposed system has the frequency pull-in range of -200 KHz +2.50 KHz. Moreover, it has the ability of adaptive loop bandwidth control according to the amount of frequency offset to improve the acquisition time while reducing the phase noise.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.11 no.9
• /
• pp.3502-3509
• /
• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.17 no.7
• /
• pp.1618-1626
• /
• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• The KIPS Transactions:PartC
• /
• v.13C no.2 s.105
• /
• pp.241-248
• /
• 2006

Mobile computing has such characteristics as portability, wireless network, mobility, etc. These characteristics cause various problems to mobile terminals like frequent disconnection, high error rate, and varying network status. These problems motivate us to develop an adaptive error control mechanism for supporting multimedia service in mobile computing environment. In this paper, we propose the Adaptive Error Control(AEC) scheme using client's buffer size and current error rate. After categorizing the status into four groups according to client's buffer size and current error rate, this scheme applies an appropriate error control scheme to each status. In this scheme, thresholds of buffer size and error rate are determined by the data transmission time, play rate and average VOP size, and by the probability of error for a sequence of packets. The performance of proposed scheme is evaluated by flaying MPEG-4 files on an experimental client/server environment, respectively. The results show that error correcting rate is similar to other schemes while the time for correcting error reduce a little. In addition, the size of data for correcting error is decreased by 23% compared with FEC and Hybrid FEC, respectively. Theses results demonstrate that the proposed scheme is more suitable in mobile computing environment with small bandwidth and varying environment than existing schemes.

• Journal of Electrical Engineering and Technology
• /
• v.10 no.3
• /
• pp.1319-1327
• /
• 2015

As wind power farm (WPF) installations continue to grow, monitoring and controlling large-scale WPFs presents new challenges. In this paper, a hierarchical network architecture is proposed in order to provide remote monitoring and control of large-scale WPFs. The network architecture consists of three levels, including the WPF comprised of wind turbines and meteorological towers, local control center (LCC) responsible for remote monitoring and control of wind turbines, and a central control center (CCC) that offers data collection and aggregation of many WPFs. Different scenarios are considered in order to evaluate the performance of the WPF communications network with its hierarchical architecture. The communications network within the WPF is regarded as the local area network (LAN) while the communication among the LCCs and the CCC happens through a wide area network (WAN). We develop a communications network model based on an OPNET modeler, and the network performance is evaluated with respect to the link bandwidth and the end-to-end delay measured for various applications. As a result, this work contributes to the design of communications networks for large-scale WPFs.

• 전자공학회논문지 IE
• /
• v.43 no.2
• /
• pp.80-86
• /
• 2006

The wireless communication based on LDPC and adaptive spatial-subcarrier coded modulation using MQAM for orthogonal frequency division multiplexing (OFDM) wireless transmission by using instantaneous channel state information and employing multiple antennas at both the transmitter and the receiver. Adaptive coded modulation is a promising idea for bandwidth-efficient transmission on time-varying, narrowband wireless channels. On power limited Additive White Gaussian Noise (AWGN) channels, low density parity check (LDPC) codes are a class of error control codes which have demonstrated impressive error correcting qualities, under some conditions performing even better than turbo codes. The paper demonstrates OFDM with LDPC and adaptive modulation applied to Multiple-Input Multiple-Output (MIMO) system. An optimization algorithm to obtain a bit and power allocation for each subcarrier assuming instantaneous channel knowledge is used. The experimental results are shown the potential of our proposed system.

• Journal of IKEEE
• /
• v.14 no.1
• /
• pp.33-39
• /
• 2010

In this paper, a 5Gb/s receiver with an adaptive equalizer for serial link interfaces is proposed. For effective gain control, a least-mean-square (LMS) algorithm was implemented with two internal signals of slicers instead of output node of an equalizing filter. The scheme does not affect on a bandwidth of the equalizing filter. It also can be implemented without passive filter and it saves chip area and power consumption since two internal signals of slicers have a similar DC magnitude. The proposed adaptive equalizer can compensate up to 25dB and operate in various environments, which are 15m shield-twisted pair (STP) cable for DisplayPort and FR-4 traces for backplane. This work is implemented in $0.18-{\mu}m$ 1-poly 4-metal CMOS technology and occupies $200{\times}300{\mu}m^2$. Measurement results show only 6mW small power consumption and 2Gbps operating range with fabricated chip. The equalizer is expected to satisfy up to 5Gbps operating range if stable varactor(RF) is supported by foundry process.

• The Transactions of the Korea Information Processing Society
• /
• v.4 no.7
• /
• pp.1770-1780
• /
• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• The KIPS Transactions:PartC
• /
• v.10C no.7
• /
• pp.891-898
• /
• 2003

This paper analyzes and proposes a scheme that improves the performance of the RTP that is developed to support the end-to-end transmission function and QoS monitor function for real-time multimedia data transmission. Although the existing RTP module supports real-time transmission, it has some problems in guaranteeing QoS parameters. To solve this problem, we propose a new Selective Repeat Adaptive Rate Control (SRARC). The SRARC can support QoS by referring to the data transmission status from the client and then classifying the network status into three levels. It selectively transmits multimedia data and dynamically controls transmission rates based on such information as bandwidth, packet loss rate, and latency that can be calculated in data transfer phase. To verify the SRARC, we implement it in real local area networks and compare the QoS parameters of the SRARC with those of the SR and RTP By the experimental results, the SRARC shows better performance in the aspects of bandwidth usage rate, packet loss rates, and transmission delays than the existing RTP schemes.

• Journal of IKEEE
• /
• v.2 no.2 s.3
• /
• pp.225-232
• /
• 1998

Currently NTSC, PAL, and SECOM are widely used for TV broadcasting systems. In Korea, NTSC has been used to reduce transmission bandwidth and broadband flickers using the Interlaced scanning method. Image data in the Interlaced scanning method require De-interlacing compensation for PC-based multimedia applications. The existing compensation algorithms such as ZOI, FOI, and ELA provieds simple computations and effective image compensation while the PSNR is low and horizontal and vertical edges are hardly detected. In this paper, the ADI(Adaptive De-Interlacing) algorithm that can increase PSNR and detect horizontal and vertical edges is proposed and a hardware system is implemented using three ACTEL 1020B FPGA chips. The system consists of the algorithm part implemented using two FPGAs and the memory control part implemented using rest one. Also the system operation is investigated for real time processing.