• Journal of Korea Multimedia Society
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• v.20 no.2
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• pp.101-107
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• 2017

LabVIEW coding for video dehazing was developed. The dark channel prior proposed by K. He was applied to remove fog based on a single image, and K. B. Gibson's median dark channel prior was applied, and implemented in LabVIEW. In other words, we improved the image processing speed by converting the existing fog removal algorithm, dark channel prior, to the LabVIEW system. As a result, we have developed a real-time fog removal system that can be commercialized. Although the existing algorithm has been utilized, since the performance has been verified real - time, it will be highly applicable in academic and industrial fields. In addition, fog removal is performed not only in the entire image but also in the selected area of the partial region. As an application example, we have developed a system that acquires clear video from the long distance by connecting a laptop equipped with LabVIEW SW that was developed in this paper to a 100~300 times zoom telescope.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.1016-1023
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• 1998

DS-CDMA system requeires a power-efficient transmission and reception scheme to increase capacity. In this paper, coherent detection of DS-CDMA system with M-ary orthogonal signaling is considered, and multistage decision-directed channel estimator is proposed. The multistage decision-directed channel estimator is based on the fact that better channel estimation can be obtained by using the final decisions of the detector with a conventional decision-directed channel estimator. In this scheme, symbol detector and channel estimator operate algernaely. by simulation, it is shown that the proposed method achieves performance gain capare to conventional method, specially at low signal to noise ratio over fast time-variant channel with diversity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12A
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• pp.1296-1302
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• 2007

In this paper, we propose joint estimation methods of carrier offset and channel estimation for MB-OFDM system with low complexity. The proposed methods estimate the channel by using low-rank LMMSE channel estimation which reduces the system complexity by applying the optimal number of rank to evaluate the frequency offset and additionally using the simple algorithm using the auto-correlation property of the estimated channel. We simulate the proposed algorithms under the IEEE 802.15 TG3a UWB channel model.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.1-10
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• 2010

Impulse Radio Ultra-wideband (IR-UWB) system employing the short pulse has the tolerable characteristics for the multipath environment. The corresponding transceiver with low power consumption can be simply implemented. On the other hand, in the receiver side, the precise channel estimation is required for the knowledge of essence in channel due to the short period of pulse. The estimated gains and delays in channel are used in the rake receiver. The resulting parameters we search have a strong influence on the performance in the entire system. We introduce that the essential parameters can be obtained more precisely through the preamble in receiver side for the channel estimation and related technologies are presented.

• International Journal of Computer Science & Network Security
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• v.22 no.11
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• pp.241-247
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• 2022

Ad hoc network is self-organized network powered by battery. The reliability of virtual user identification and channel security are reduced when SNR is low due to limited user energy. In order to solve this problem, a joint virtual user identification and channel security en/decoding method is proposed in this paper. Transmitter-receiver-based virtual user identification code is generated by executing XOR operation between orthogonal address code of transmitter and pseudo random address code of receiver and encrypted by channel security code to acquire orthogonal random security sequence so as to improve channel security. In order to spread spectrum as well as improve transmission efficiency, data packet is divided into 6-bit symbols, each symbol is mapped with an orthogonal random security sequence. Subspace-based method is adopted by receiver to process received signal firstly, and then a judgment model is established to identify virtual users according to the previous processing results. Simulation results indicate that the proposed method obtains 1.6dB E_b/N₀ gains compared with reference methods when miss alarm rate reaches 10^-3.

• Journal of Broadcast Engineering
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• v.8 no.3
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• pp.228-237
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• 2003

Eureka-147 DAB(Digital Audio Broadcasting) system on which DMB(Digital Multimedia Broadcasting) system is based, was designed for the requirements of CD qualify audio with ${10}^{-4}$ bit error rate. Audio program may be primary service in DAB system, but multimedia program can be primary service in DMB system. Therefore, the bit error rate required must be below ${10}^{-7}$∼${10}^{-8}$ to transmit multimedia data via DMB channel. In order to meet the requirements and keep backward compatibility of DAB system we propose an outer channel coding scheme using Reed-Solomon coding and convolutional interleaving. This paper shows the simulation results for DMB channel performance based on mobile channel model. Also, it describes the needs and the effects of the outer channel coding.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.1
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• pp.22-32
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• 2011

In this paper, we first analyze the differences of underwater acoustic (UWA) orthogonal frequency division multiplexing (OFDM) systems and conventional terrestrial OFDM system, and give a simple introduction of the backgrounds. By considering the real UWA channel environments, the measured channel data is used to generate the UWA channel model and calculate the relative parameters for underwater OFDM systems. Practical least square (LS) based channel estimation with linear interpolation are adopted to obtain the channel state information (CSI) at receiver side. As multi-input multi-output (MIMO) processing techniques, Alamouti code is implemented and evaluated to perform for space time block coding (STBC) and space frequency block coding (SFBC) for UWA OFDM systems with the MIMO configuration of $2{\times}1$, at the same time, $1{\times}2$ maximum ratio combining (MRC) is performed for the purpose of comparison. The simulation results show that, with perfect channel estimation, SFBC failed to work duo to the serious frequency selectivity of UWA channel environments. When the practical channel estimation is applied, in the case of STBC, the proposed 4-column pilot pattern gives better performance about 7dB than SISO system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.1
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• pp.9-18
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• 2003

In this paper, we analyzed the throughput of DS/CDMA system applying packet combining scheme combined with Type- I Hybrid ARQ scheme over AWGN channel and Nakagami fading channel with RAKE receiver. As the parameter for analysis, we used number of combined packet(L), number of diversity branch$({L_c})$, fading index(m), and length of packet(N), and used CRC-12 error detection code and (2,1,3) convolutional code. As a results, we found that throughput of system over Nakagami fading channel with RAKE receiver was superior to throughput over AWGN channel, and throughput of system decreases rapidly as channel degrades when number of combined packet(L) was increased. However throughput of system with the combining scheme was achieved even at low ${E_b}/{N_o}$. Also, we found that throughput of system was increased when fading index(m) and number of diversity branch$({L_c})$ were increased, but it was decreased when number of user(K) and length of packet(N) were increased.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• Journal of KIISE:Information Networking
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• v.30 no.1
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• pp.9-16
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• 2003

In this paper, we propose a method for supporting VCR operations in a patching based multicast VOD system. Random access, pause and resume operations are supported in our system, and the channel and session scheduling algorithms for the VCR operations are proposed. When it is necessary to join a sharable multicast channel in the admission control for the VCR operations, the patching technique, which shares an on-going regular channel and allocates a new patching channel for the missing data, is used. Therefore, unlike the previous approach that allocates an excessive number of I-channels, service latency is minimized and channel usage is optimized in our system. Moreover buffered data. which is saved in disk during patching, is reused to prevent unnecessary patching channel allocation. For this. the patching management information is extended and a buffed data management scheme is proposed. In our system, the First-Come-First-Served scheduling is used to inform clients the service latency for the VCR operations immediately.