• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.8B
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• pp.1003-1011
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• 2011

In this paper, a real-time multi-source sensor data management system based on wireless sensor network is proposed and implemented. The proposed management system is designed to transmit the wireless data to the server in order to monitor and control the multi-source target's status efficiently by analyzing them. The proposed system is implemented to make it possible to control and transmit the wireless sensor data by classifying them, of which data are issued from the clustered sources composed of a number of the remote multiple sensors. In order to evaluate the performance of the proposed system, we measure and analyze both the transmission delay time according to the distance and the data loss rate issued from multiple data sources. From these results, it is verified that the proposed system has a good performance.

• International Journal of Control, Automation, and Systems
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• v.6 no.2
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• pp.186-193
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• 2008

The Internet's low cost and ubiquity present an attractive option for real-time distributed control of processes on the factory floor. When integrated with the Internet, the LonWorks open control network can give ubiquitous accessibility with the distributed control nature of information on the factory floor. One of the most important points in real-time distributed control of processes is timely response. There are many processes on the factory floor that require timely response. However, the uncertain time delay inherent in the network makes it difficult to guarantee timely response in many cases. Especially, the transmission characteristics of the LonWorks/IP network show a highly stochastic nature. Therefore, the time delay problem has to be resolved to achieve high performance and quality of the real-time distributed control of the process in the LonWorks/IP Virtual Device Network (VDN). It should be properly predicted and compensated. In this paper, a new distributed control scheme that can compensate for the effects of the time delay in the network is proposed. It is based on the PID controller augmented with the Smith predictor and disturbance observer. Designing methods for output feedback filter and disturbance observer are also proposed. Tracking position control experiment of a geared DC Servo motor is performed using the proposed control method. The performance of the proposed controller is compared with that of the Internal Model Controller (IMC) with the Smith predictor. The result shows that the performance is improved and guaranteed by augmenting a PID controller with both the Smith predictor and disturbance observer under the stochastic time delay in the LonWorks/IP VDN.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• ETRI Journal
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• v.31 no.2
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• pp.111-120
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• 2009

Network-on-chip (NoC) architecture provides a highper-formance communication infrastructure for system-on-chip designs. Circuit-switched networks guarantee transmission latency and throughput; hence, they are suitable for NoC architecture with real-time traffic. In this paper, we propose an efficient integrated scheme which automatically maps application tasks onto NoC tiles, establishes communication circuits, and allocates a proper bandwidth for each circuit. Simulation results show that the average waiting times of packets in a switch in $6{\times}6$6, $8{\times}8$, and $10{\times}10$ mesh NoC networks are 0.59, 0.62, and 0.61, respectively. The latency of circuits is significantly decreased. Furthermore, the buffer of a switch in NoC only needs to accommodate the data of one time slot. The cost of the switch in the circuit-switched network can be reduced using our scheme. Our design provides an effective solution for a critical step in NoC design.

• Journal of the Korean Institute of Intelligent Systems
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• v.22 no.2
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• pp.148-153
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• 2012

In this paper, we propose a fault-tolerant QoS routing scheme based on interference awareness for providing both high throughput and minimum end-to-end delay for wireless sensor networks. With the proposed algorithm, it is feasible to find out the optimal transmission path between sensor nodes to the sink node by using cumulative path metric where real-time delivery, high energy efficiency and less interference are considered as in path selection. Finally, simulation results show that network throughput and delay can be improved by using the proposed routing scheme.

• ETRI Journal
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• v.32 no.4
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• pp.493-502
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• 2010

The operation time of an encoder is one of the critical implementation issues for satisfying the timing requirements of Long Term Evolution (LTE) systems because the encoder is based on binary operations. In this paper, we propose a design and implementation of a latency efficient encoder for LTE systems. By virtue of 8-bit parallel processing of the cyclic redundancy checking attachment, code block (CB) segmentation, and a parallel processor, we are able to construct engines for turbo codings and rate matchings of each CB in a parallel fashion. Experimental results illustrate that although the total area and clock period of the proposed scheme are 19% and 6% larger than those of a conventional method based on a serial scheme, respectively, our parallel structure decreases the latency by about 32% to 65% compared with a serial structure. In particular, our approach is more latency efficient when the encoder processes a number of CBs. In addition, we apply the proposed scheme to a real system based on LTE, so that the timing requirement for ACK/NACK transmission is met by employing the encoder based on the parallel structure.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.177-178
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• 2007

WiBro system provides reliable broadband communication services for mobile and portable subcribers. It allows interference-free reception under the conditions of multipath propagation and transmission errors. Thus, powerful channel-error correction ability Is required. CC/CTC Decoder which Is mandatory for WiBro system needs lots of computations for real-time operation. So, it is desired to design a CC/CTC Decoder having highly optimized hardware scheme for low latency operation under high data rates. This paper proposes an efficient CC/CTC Decoder structure for high data rate WiBro system. Particularly, the proposed CTC Decoder architecture reduces decoding delay by applying pipelining and multiple decoding blocks. Simulation results show that reduction of about 80% of processing time is enabled with the proposed CC/CTC Decoder despite of increase in are.

• KIPS Transactions on Computer and Communication Systems
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• v.6 no.6
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• pp.255-260
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• 2017

The limitation of real-time video conferencing system is that the delay of network and buffering and the transmission of user information are not efficiently performed between systems, so real - time performance is not guaranteed completely. In order to overcome this problem, the study on the extension of the network infrastructure and the jitter delay is actively carried out, but the study on the buffering delay is insufficient. In this paper, we propose a frame-rate control buffer management (FRCB) scheme to solve the problem caused by buffering delay. The FRCB is used to prevent overflow and underflow of the buffer by adopting the two-stage buffer threshold of Fast-play THreshold (FTH) and Slow-play THreshold (STH). Therefore, it showed better performance than jitter buffer even under high CPU load, and showed that it is suitable for high quality real time video conferencing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1C
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• pp.55-64
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• 2006

In this paper, a new web-based remote 3D imaging communication system employing an adaptive matching algorithm is suggested. In the proposed method, feature values are extracted from the stereo image pair through estimation of the disparity and similarities between each pixel of the stereo image. And then, the matching window size for disparity estimation is adaptively selected depending on the magnitude of this feature value. Finally, the detected disparity map and the left image is transmitted into the client region through the network channel. And then, in the client region, right image is reconstructed and intermediate views be synthesized by a linear combination of the left and right images using interpolation in real-time. From some experiments on web based-transmission in real-time and synthesis of the intermediate views by using two kinds of stereo images of 'Joo' & 'Hoon' captured by real camera, it is analyzed that PSNRs of the intermediate views reconstructed by using the proposed transmission scheme are highly measured by 30dB for 'Joo', 27dB for 'Hoon' and the delay time required to obtain the intermediate image of 4 view is also kept to be very fast value of 67.2ms on average, respectively.

• Journal of KIISE:Information Networking
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• v.28 no.3
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• pp.389-405
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• 2001

In these days, we have increasing demands on the real-time services, especially for the multimedia data transmission in both of wired and wireless environments and thus efficient and stable ways of transmitting realtime data are needs. Although RTP is widely used for internet-based realtime applications, it cannot avoid packet losses, due to the use of UDP stack and its underlying layers. In the case of mobile computing applications, the packet losses are more frequent and consecutive because of the limited bandwidth. In this paper, we first statistically analyze the characteristics of packet losses in the wired and wireless communications, based on Gilbert model, and a new packet recovery scheme for realtime data transmission is presented. To reflect the transmission characteristics of the present network environment, our scheme makes the sender to dynamically adjust the amount of redundant information, using the current packet loss characteristic parameters reported by the receiver. Additionally, we use relatively large and discontinuous offset values, which enables us to recover from both of the random and consecutive packet losses. Due to these characteristics, our scheme is suitable for the mobile computing environment where packet loss rates are relatively high and varies rapidly in a wide range. Since our scheme is based on the analytic model form statistics, it can also be used for other network environments. We have implemented the scheme with Mobile IP and RTP/RTCP protocols to experimentally verify its efficiency.