• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.10
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• pp.6296-6302
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• 2014

A new technique, which can increase the channel bandwidth in an optical wireless orthogonal frequency division multiplexing (OFDM) link based on a light emitting diode (LED), is proposed. The technique uses adaptive sampling to convert an OFDM signal to a sparse waveform. In compressive sensing (CS), a sparse signal that is sampled below the Nyquist/Shannon limit can be reconstructed successively with sufficient measurements. The data rate of the proposed CS-based visible light communication (VLC)-OFDM link increases from 30.72 Mb/s to 51.2 Mb/s showing an error vector magnitude (EVM) of 31 % at the quadrature phase shift keying (QPSK) symbol.

• Journal of KIISE:Information Networking
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• v.30 no.4
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• pp.494-503
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• 2003

The IEEE 1394 High Speed Serial Bus is a versatile, high-performance, and low-cost method of promoting interoperability between all types of A/V and computing devices. IEEE 1394 provides two transfer modes: asynchronous mode for best effort service and isochronous mode for best effort service with bandwidth reservation. This paper shows the bus performance and compared the transfer odes first at the link level and then at the application level. For the application level performance, we analyze the bus systems with fixed and adaptive interfaces, applied between the upper layer and the 1394 layer, using polling systems. Also we verifies the analysis models with simulation studies. Based on our analysis, we conclude that the adaptive interface reduces the bus access time and so increases the bus utilization.

• 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.

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.493-504
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• 2007

With the advance and proliferation of computer and wireless network technology, it is common to access to network through the wireless network using mobile device. The ratio of using the streaming media out of many applications through the network is increasing not only in the wired network but also in the wireless network. The streaming media is much bigger than other contents and requires more network bandwidth to communicate and more computing resources to process. However the mobile devices have relatively poor computing resource and low network bandwidth. If the streaming media service is provided for mobile devices without any consideration about the network bandwidth and computing power, it is difficult for the client to get the service of high quality. Since especially mobile devices are supported with very limited energy capacity from the battery, the streaming media service should be adjusted to the varying energy state of mobile devices to ensure the complete playback of streaming media. In this paper, we propose a new method to guarantee the complete playback time of the streaming media for the mobile clients by dynamically controlling transmitted frame rate to the client according to the estimated available time of mobile device using battery model reflecting the characteristic of the battery. Since the proposed method controls the number of frames transmitting to the client according to the energy state of the mobile device, the complete playback time is guaranteed to mobile clients.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.11
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• pp.1034-1042
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• 2004

In this paper, as one of countermeasure techniques for a frequency selective fading, an adaptive analog slope equalizer(ASE) applicable to 64-QAM digital radio relay system is presented in terms of principle, implementation, and its performance. Also interference of cross-talk between I- and Q-channel caused by a fiequency selective lading has been analyzed by doing channel model in the baseband, which make it possible to derive the solution for implementing ASE in If-band. The effects of signal for the faded channel are investigated in the time and the frequency domains, respectively, with/without ASE. As system performance, it is shown that the signature is improved up to 6.2 dB at the edge of signal bandwidth for a given BER 10$\^$-3/.

• Journal of electromagnetic engineering and science
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• v.10 no.2
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• pp.73-77
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• 2010

This paper proposes a novel memory effect measurement technique in RF power amplifiers(PAs) using a two-tone intermodulation distortion(IMD) signal with a very simple and intuitive algorithm. Based on the proposed predistortive tone cancellation technique, the proposed measurement method is capable of measuring the relative phase and magnitude of the third-order and fifth-order IMDs, as well as the fundamental signal. The measured relative phase between the higher and lower IMD signal for specific tone spacing can be interpreted as the group delay(GD) information of the IMD signal concerned. From the group delay analysis, we can conclude that an adaptive control of GD as well as the magnitude and phase is a key function in increasing the linearization bandwidth and the dynamic range in a predistortion(PD) technique.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.477-488
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• 2001

In wireless networking environments, supporting guaranteed quality of service to mobile hosts is difficult due to the facts that wireless networks have limited bandwidth and mobile hosts frequently move in and out of cells. In spite of the characteristics of wireless communications, the quality of some types of services, i.e., real-time services, must be guaranteed at a certain level. When a mobile host moves into another cell, service rates for mobile hosts in wireless networks may be adjusted since wireless networks have limited bandwidths. In this paper, we propose two resource allocation algorithms in wireless mobile networks, using quality of service (QoS) specifications. For efficient use of resources of wireless networks, the proposed algorithms dynamically allocate rates of flows in proportion to QoS with limited resources.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.1
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• pp.202-211
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• 2001

This paper deals with an analysis of multipath which affect a transmission performance in underwater acoustic channel. Underwater acoustic channel with multipath structure is introduced to mathematical modelling for a basin environment. In this paper, SMR(Signal to Multipath Ratio) which is defined as a parameter of multipath's effect is presented as a mathematical equation, and the equation of SMR is simulated by MATLAB program. Furthermore, this paper is also dealt with an implementation of modulation and demodulation system for acoustic transmission. Acoustic Transmission is limited by frequency bandwidth, so $\pi/4 QPSK$(Quadrature Phase Shift Keying) methods which is very useful at frequency ]imitation and FM(Frequency Modulation) are used at acoustic communication system. This implemented hybrid modulation/demodulation system is used as an analog board of image transmission system. In this system, adaptive equalization for reducing the multipath effect and baseline JPEG used for an image compressing are also stated.

• ETRI Journal
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• v.25 no.5
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• pp.356-368
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• 2003

This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

• IEIE Transactions on Smart Processing and Computing
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• v.3 no.1
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• pp.1-9
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• 2014

This paper proposes a low-latency Sample Adaptive Offset filter (SAO) architecture and its Single Instruction Multiple Data (SIMD) optimization scheme to achieve fast High Efficiency Video Coding (HEVC) decoding in a multi-core environment. According to the HEVC standard and its Test Model (HM), SAO operation is performed only at the picture level. Most realtime decoders, however, execute their sub-modules on a Coding Tree Unit (CTU) basis to reduce the latency and memory bandwidth. The proposed low-latency SAO architecture has the following advantages over picture-based SAO: 1) significantly less memory requirements, and 2) low-latency property enabling efficient pipelined multi-core decoding. In addition, SIMD optimization of SAO filtering can reduce the SAO filtering time significantly. The simulation results showed that the proposed low-latency SAO architecture with significantly less memory usage, produces a similar decoding time as a picture-based SAO in single-core decoding. Furthermore, the SIMD optimization scheme reduces the SAO filtering time by approximately 509% and increases the total decoding speed by approximately 7% compared to the existing look-up table approach of HM.