• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9C
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• pp.895-904
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• 2007

This paper proposes a robust sub-optimal smart antenna in Code Division Multiple Access (CDMA) basestation. It makes use of the property of the Least Mean Square (LMS) algorithm and the Conjugate Gradient Method (CGM) algorithm for beamforming processes. The weight update takes place at symbol level which follows the PN correlators of receiver module under the assumption that the post correlation desired signal power is far larger than the power of each of the interfering signals. The proposed algorithm is simple and has as low computational load as five times of the number of antenna elements(O(5N)) as a whole per each snapshot. The output Signal to Interference plus Noise Ratio (SINR) of the proposed smart antenna system when the weight vector reaches the steady state has been examined. It has been observed in computer simulations that proposed beamforming algorithm improves the SINR significantly compared to the single antenna case. The convergence property of the weight vector has also been investigated to show that the proposed hybrid algorithm performs better than CGM and LMS during the initial stage of the weight update iteration. The Bit Error Rate (BER) characteristics of the proposed array has also been shown as the processor input Signal to Noise Ratio (SNR) varies.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.6
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• pp.39-47
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• 2012

As a scheme to efficiently reduce the effects of multipath fading in next generation wireless communication systems, cooperative communication systems have recently come into the spotlight. Since these cooperative communication systems use cooperative relays with diverse fading coefficients to transmit information, having all relays participate in cooperative communication may result in unnecessary waste of resources, and thus relay selection schemes are required to efficiently use wireless resources. In this paper, we propose an efficient relay selection scheme through self-learning in cooperative wireless networks using Q-learning algorithm. In this scheme, we define states, actions and two rewards to achieve good SER (Symbol Error Rate) performance, while selecting a small number of cooperative relays. When these parameters are well-defined, we can obtain good performance. For demonstrating the superiority of the proposed Q-learning, We compared the proposed scheme with Q-learning and a relay selection scheme with a mathematical analysis. The simulation results show that, compared to a scheme that obtains optimum relays through a mathematical analysis, the proposed scheme uses resources efficiently by using smaller numbers of relays with comparable SER performance. According to these simulation results, the proposed scheme can be considered as a good attempt for future wireless communication.

• Journal of Navigation and Port Research
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• v.38 no.4
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• pp.349-356
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• 2014

In the underwater communication, the performance of system is reduced because of the inter-symbol interference occur by the multi-path. In the recent years, to deal with poor channel environment and improve the throughput, the efficient concatenated structure of equalization, channel codes and Space Time Codes has been studied as MIMO system in the underwater communication. Space Time Codes include Space Time Block Codes and Space Time Trellis Codes in underwater communication. Space Time Trellis Codes are optimum for equalization and channel codes among the Space Time Codes to apply in the MIMO environment. Therefore, in this paper, turbo pi codes are used for the outer code to efficiently transmit in the multi-path channel environment. The inner codes consist of Space Time Trellis Codes with transmission diversity and coding gain in the MIMO system. And Zero Forcing method is used to remove inter-symbol interference. Finally, the performance of this model is simulated in the underwater channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.7C
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• pp.706-714
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• 2009

In this paper, we proposed a burst mode AGC loop and preamble detector applicable for VDL(VHF Digital Link) mode-2 using D8PSK modulation scheme and the performance analysis of proposed schemes is described. Generally the AGC scheme can be divided into two types, continuos and burst mode AGC. The continuos mode is performed well only with an analog feedback AGC loop. But the analog feedback AGC loop is not suitable for burst mode since its gain lock time is more than preamble duration, which causes the preamble detector misses preamble. Hence a fast digital AGC loop is required for burst mode. Also the AGC loop has to be designed with full gain during idle time to detect bursts although the signal level is very low. If the time to acquire gain lock is slow, the preamble detector fail to detect burst due to saturation of a lot of preamble samples. The receiver performance might be down even if the burst was detected because the preamble is used to estimate several parameters need to demodulation at receiver. In this paper we analysed relationships between the AGC loop and preamble detector. we present an AGC loop and preamble detector in burst mode.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.1
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• pp.1-7
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• 2010

PAPR of the pilot symbols can be reduced down by the CAZAC sequence in the SC-FDMA communication system. However, it is very complicated and takes quite a long time to compute the interpolation between the OFDM information symbols for the channel estimation because the pilot data are trasmitted in the block type. Furthermore, situation will be much more serious in the severe fading channel. Actually the pilot insertion of the comb type is much efficient and convenient for the channel estimation since the calculation of the interpolation can be made in the frequency domain symbol by symbol. But, the PAPR will be regrown when the pilot data are inserted with the information data in the comb type. So, in this paper, we like to study the PAPR reduction and comb type pilot assignment for the efficient channel estimation. Unlike the conventional SLM(selected mapping) method requiring the side information, our improved SLM method is to use the phase rotation sequence into information data without rotating phase of pilot. We use different pilot data according to the different phase rotation sequence. From the simulation result, it can be confirmed that when SLM method of 4 phase rotation sequence is used, PAPR is almost same to the block type method without pilot.

• Cartoon and Animation Studies
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• s.17
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• pp.53-67
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• 2009

According to the changes of current regulation of media, the broadcasting environment has been loosened by market power, and came into competitive era for broadcasting industry with new technology of combined broadcasting and communication. To resolve these issues, the broadcasting provides are trying process of new channel brand identity to introduce their channels. The purpose of the research is to analyze Spot animation that has fast access to viewers among channel brand identity factors, review expandability and usage of Spot animation character to plan, manage character that can improve awareness and loyalty. Therefore it will study creation background and production, activities of each character representing KBS and MBC, compare the structure, color, material, expression of Spot animation character and summarize properties. To conclusion, three characters of KBS and MBC failed to apply structured and professional plan and management, only expressed image that each channel pursue on Station ID Spot animation and had issues of not deeply sighting expandability and usefulness as a character. The important factor to improve channel brand value of broadcasting provider is to have differentiated identity, and the core of it is to make image-responsible Station ID to be effective. Therefore, Spot animation and character of Station ID is a very important factor so there must be professional research and structural management from the early planning and production for character that represents the providers.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.9
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• pp.16-23
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• 2009

In this paper, we propose a new distributed Alamouti space-time block coding scheme using cooperative relay system composed of one source node, three relay nodes and one destination node. The source node is assumed to be equipped with two antennas which respectively use a 2-beam array to communicate with two nodes selected from the three relay nodes. During the first time slot, the two signals which respectively were transmitted by one antenna at the source, are selected by one relay node, added, amplified, and forwarded to the destination. During the second time slot, the other two relay nodes implement the conjugate and minusconjugate operations to the two received signals, respectively, each in turn is amplified and forwarded to the destination node. This transmission scheme represents a new distributed Alamouti space-time block code that can be constructed at the relay-destination channel. Through an equivalent matrix expression of symbols, we analyze the performance of this proposed space-time block code in terms of the chernoff upper bound pairwise error probability (PEP). In addition, we evaluate the effect of the coefficient $\alpha$ ($0{\leq}{\alpha}{\leq}1$) determined by power allocation between the two antennas at the source on the received signal performance. Through computer simulation, we show that the received signals at the three relays have same variance only when the value of $\alpha$ is equal to $\frac{2}{3}$, as a consequence, a better performance is obtained at the destination. These analysis results show that the proposed scheme outperforms conventional proposed schemes in terms of diversity gain, PEP and the complexity of relay nodes.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.45 no.7
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• pp.61-68
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• 2008

In this paper, we propose an efficient hardware architecture and algorithm to increase an encoding process rate and implement a hardware for CABAC (Context Adaptive Binary Arithmetic Coding) which is used with one of the entropy coding ways for the latest video compression technique, H.264/AVC (Advanced Video Coding). CABAC typically provides a better high compression performance maximum 15% compared with CAVLC. However, the complexity of operation of CABAC is significantly higher than the CAVLC. Because of complicated data dependency during the encoding process, the complexity of operation is higher. Therefore, various architectures were proposed to reduce an amount of operation. However, they have still latency on account of complicated data dependency. The proposed architecture has two techniques to implement efficient pipeline architecture. The one is quick calculation of 7, 8th bits used to calculate a probability is the first step in Binary arithmetic coding. The other is one step reduced pipeline arcbitecture when the type of the encoded symbols is MPS. By adopting these two techniques, the required processing time was reduced about 27-29% compared with previous architectures. It is designed in a hardware description language and total logic gate count is 19K using 0.18um standard cell library.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.49-58
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• 2006

In this paper, a curve-fitting channel estimation method is proposed for orthogonal frequency division multiplexing (OFDM) system in a time-varying frequency-selective fading channel. The method can greatly improve channel state information (CSI) estimation accuracy by performing smoothing and interpolation through consecutive curve-fitting processes in both time domain and frequency domain. It first evaluates least-squares (LS) estimates using pilot symbols and then the estimates are approximated to a polynomial with proper degree in the LS error sense, starting from one preferred domain in which pilots we densely distributed. Smoothing, interpolation, and prediction are performed subsequently to obtain CSI estimates for data transmission. The channel estimation processes are completed by smoothing and interpolating CSI estimates in the other domain once again using the channel estimates obtained in one domain. The performance of proposed method is influenced heavily on the time variation and frequency selectivity of channel and pilot arrangement. Hence, a proper degree of polynomial and an optimum approximation interval according to various system and channel conditions are required for curve-fitting. From extensive simulation results in various channel environments, we see that the proposed method performs better than the conventional methods including the optimal Wiener filtering method, in terms of the mean square error (MSE) and bit error rate (BER).

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.8
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• pp.96-105
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• 1998

The aim of this paper is to propose the method of WDHMM(Weighted DHMM), using the MLP-VQ for the improvement of speaker-independent connect-digit recognition system. MLP neural-network output distribution shows a probability distribution that presents the degree of similarity between each pattern by the non-linear mapping among the input patterns and learning patterns. MLP-VQ is proposed in this paper. It generates codewords by using the output node index which can reach the highest level within MLP neural-network output distribution. Different from the old VQ, the true characteristics of this new MLP-VQ lie in that the degree of similarity between present input patterns and each learned class pattern could be reflected for the recognition model. WDHMM is also proposed. It can use the MLP neural-network output distribution as the way of weighing the symbol generation probability of DHMMs. This newly-suggested method could shorten the time of HMM parameter estimation and recognition. The reason is that it is not necessary to regard symbol generation probability as multi-dimensional normal distribution, as opposed to the old SCHMM. This could also improve the recognition ability by 14.7% higher than DHMM, owing to the increase of small caculation amount. Because it can reflect phone class relations to the recognition model. The result of my research shows that speaker-independent connected-digit recognition, using MLP-VQ and WDHMM, is 84.22%.