• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.11
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• pp.2594-2603
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• 1997

A novel method of adaptive beam forming is presented in this paper. The proposed technique provides for a suboptimal beam pattern that increases the Signal to Noise/Interference Ratio (SNR/SIR), thus, eventually increases the capacity of the communication channel, under an assumption that the desired signal is dominant compared to each component of interferences at the receiver, which is precoditionally achieved in Code Division Multiple Access (CDMA) mobile communications by the chip correlator. The main advantages of the new technique are:(1)The procedure requires neither reference signals nor training period, (2)The signal interchoerency does not affect the performance or complexity of the entire procedure, (3)The number of antennas does not have to be greater than that of the signals of distinct arrival angles, (4)The entire procedure is iterative such that a new suboptimal beam pattern be generated upon the arrival of each new data of which the arrival angle keeps changing due tot he mobility of the signal source, (5)The total amount of computation is tremendously reduced compared to that of most conventional beam forming techniques such that the suboptimal beam pattern be produced at vevery snapshot on a real-time basis. The total computational load for generating a new set of weitht including the update of an N-by-N(N is the number of antenna elements) autocovariance matrix is $0(3N^2 + 12N)$. It can further be reduced down to O(11N) by approximating the matrix with the instantaneous signal vector.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1742-1754
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• 1999

In this paper, we propose a rate control scheme, using histogram based rate-distortion (R-D) estimation, which produces a consistent picture quality between consecutive frames. The histogram based R-D estimation used in our rate control scheme offers a closed-form mathematical model that enable us to predict the bits and the distortion generated from an encoded frame at a given quantization parameter (QP) and vice versa. The most attractive feature of the R-D estimation is low complexity of computing the R-D data because its major operation is just to obtain a histogram or weighted histogram of DCT coefficients from an input picture. Furthermore, it is accurate enough to be applied to the practical video coding. Therefore, the proposed rate control scheme using this R-D estimation model is appropriate for the applications requiring low delay and low complexity, and controls the output bit-rate ad quality accurately. Our rate control scheme ensures that the video buffer do not underflow and overflow by satisfying the buffer constraint and, additionally, prevents quality difference between consecutive frames from exceeding certain level by adopting the distortion constraint. In addition, a consistent considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K=10, considering CLP=$10^{-3}$ as a criterion, it is observed that the performance improment of about 3.5 dB and 1.5 dB is obtained, respectively, in terms of $E_b$/$N_o$ by employing the concatenated FEC code with pilot symbols.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.4
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• pp.506-519
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• 1999

An adaptive array antennal and a CCI canceller have been considered as techniques for cancelling Multi-User Interference(MUI) in Direct Sequence Code Division Multiple Access(DS-CDMA) system. These techniques have different problems respectively in the process of cancelling MUI as the number of users increases. For that reason, the scheme which can cancel MUI effectively by compensating for the problems of each of the techniques has been required. For the scheme, the technique to connect an adaptive array antenna and a CoChannel Interference(CCI) canceller in cascade form has been studied. In the existing study about the cascade connection method, the effect of cancelling MUI about two interference signals is analyzed, but the analysis for the quantitative BER(Bit Error Rate) improvement according to the number of users is not considered. Therefore, in this paper, we have analyzed the degree of BER performance improvement quantitatively according to the number of users by introducing the receiving system, which connects an adaptive array antenna and a CCI canceller to a DS-CDMA system in cascade form. For the method of analyzing the performance, we have performed the theoretical analysis and the simulation, considering the case of adopting only an adaptive array antenna and of cascade connection respectively, and having compared and analyzed the results. From the results, it is confirmed that in the case of adopting only an adaptive array antenna, the problems occur in the process of cancelling MUI according to the number of users and the receiving direction of interference signals, and can be compensated by the cascade connection method. In conclusion, we have known that MUI is cancelled effectively by using the cascade connection method, and the much better BER performance improvement is obtained.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.5
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• pp.486-492
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• 2004

In this paper, we newly propose the binary CA-CDMA(constant amplitude CDMA) system using pre-coding method to solve the high PAPR problem caused by multi-user signal transmission in the CDMA system. 4-user CA-CDMA, the basis of proposed binary CA-CDMA system, makes binary output signal for 4 input users. It produces the output of binary(${\pm}$2) amplitude by using a parity signal resulting from the XOR operation of 4 users data. Another sub-channel or more bandwidth is not necessary because it is transmitted together with user data and can be easily recovered in the receiver. The extension of the number of users can be possible by the simple repetition of the basic binary 4-user CA-CDMA. For example, binary 16-user CA-CDMA is made easily by allocating the four 4-user CA-CDMA systems in parallel and leading the four outputs to the fifth 4-user CA-CDMA system as input, because the output signal of each 4-user CA-CDMA is also binary. By the same extension procedure, binary 64 and 256-user CA-CDMA systems can be made with the constant amplitude. As a result, the code rate of this proposed CA-CDMA system is just 1 and binary CA-CDMA does not change the transmission rate with the constant output signal(PAPR = 0 ㏈). Therefore, the power efficiency of the HPA can be maximized without the nonlinear distortion. From the simulation results, it is verified that the conventional CDMA system has multi-level output signal, but the proposed binary CA-CDMA system always produces binary output. And it is also found that the BER of conventional CDMA system is increased by nonlinear HPA, but the BER of proposed binary CA-CDMA system is not changed.

• Journal of Advanced Navigation Technology
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• v.14 no.5
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• pp.684-690
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• 2010

Recently, the NG(Next Generation) system is studied for supporting convergence of various services and multi mode of single terminal. And a demand of user for taking the various services is getting increased, for supporting these services, many systems being able to transmit a large message have been appeared. In the NG system, it has to be supporting the CDMA and WCDMA besides the tele communication systems using OFDM method with single terminal An intergrated system can be improved with adopting of SoC technique. For adopting SoC technique on the intergrated terminal, we have to solve the non linear problem of HPA(High Power Amplifier). Nonlinear characteristic of HPA distorts both amplitude and phase of transmit signal, this distortion cause deep adjacent channel interference. We adopt a polynomial pre-distortion technique for this problem. In this paper, a noble modem design for NG mobile communication service and a method using polynomial pre-distorter with PAPR technique for counterbalancing nonlinear characteristic of the HPA are proposed.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Magazine of the Korean Society of Agricultural Engineers
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• v.22 no.4
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• pp.108-114
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• 1980

A deterministic conceptual erosion model which simulates detachment, entrainment, transport and deposition of eroded soil particles by rainfall impact and flowing water is presented. Both upland and channel phases of sediment yield are incorporated into the erosion model. The algorithms for the soil erosion and sedimentation processes including land and crop management effects are taken from the literature and then solved using a digital computer. The erosion model is used in conjunction with the modified Kentucky Watershed Model which simulates the hydrologic characteristics from watershed data. The two models are linked together by using the appropriate computer code. Calibrations for both the watershed and erosion model parameters are made by comparing the simulated results with actual field measurements in the Four Mile Creek watershed near Traer, Iowa using 1976 and 1977 water year data. Two water years, 1970 and 1978 are used as test years for model verification. There is good agreement between the mean daily simulated and recorded streamflow and between the simulated and recorded suspended sediment load except few partial differences. The following conclusions were drawn from the results after testing the watershed and erosion model. 1. The watershed and erosion model is a deterministic lumped parameter model, and is capable of simulating the daily mean streamflow and suspended sediment load within a 20 percent error, when the correct watershed and erosion parameters are supplied. 2. It is found that soil erosion is sensitive to errors in simulation of occurrence and intensity of precipitation and of overland flow. Therefore, representative precipitation data and a watershed model which provides an accurate simulation of soil moisture and resulting overland flow are essential for the accurate simulation of soil erosion and subsequent sediment transport prediction. 3. Erroneous prediction of snowmelt in terms of time and magnitute in conjunction with The frozen ground could be the reason for the poor simulation of streamflow as well as sediment yield in the snowmelt period. More elaborate and accurate snowmelt submodels will greatly improve accuracy. 4. Poor simulation results can be attributed to deficiencies in erosion model and to errors in the observed data such as the recorded daily streamflow and the sediment concentration. 5. Crop management and tillage operations are two major factors that have a great effect on soil erosion simulation. The erosion model attempts to evaluate the impact of crop management and tillage effects on sediment production. These effects on sediment yield appear to be somewhat equivalent to the effect of overland flow. 6. Application and testing of the watershed and erosion model on watersheds in a variety of regions with different soils and meteorological characteristics may be recommended to verify its general applicability and to detact the deficiencies of the model. Futhermore, by further modification and expansion with additional data, the watershed and erosion model developed through this study can be used as a planning tool for watershed management and for solving agricultural non-point pollution problems.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.4
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• pp.31-37
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• 2008

Conventional cooperative communications can attain gain of spatial diversity and path loss reduction because destination node independently received same signal from source node and relay node located between source node and destination node. However, these techniques bring about decreased spectral efficiency with relay node and increased complexity of receiver by using maximal ratio combining (MRC). This paper has proposed cooperative ARQ protocol that can improve the above problems and can get the better performance. This method can increase the spectral efficiency than conventional cooperative communication because if the received signal from source node is satisfied by the destination preferentially, the destination transmits ACK message to both relay node and source node and then recovers the received signal. In addition, if ARQ message indicates NACK relay node operates selective retransmission and we can increase reliability of system compared with that of general ARQ protocol in which source node retransmits data. In the proposed protocol, the selective retransmission and ARQ message are to be determined by comparing log-likelihood ratio (LLR) computation of received signal from source node with predetermined threshold values. Therefore, this protocol don't waste redundant bandwidth with CRC code and can reduce complexity of receiver without MRC. We verified spectral efficiency and BER performance for the proposed protocol through Monte-Carlo simulation over Rayleigh fading plus AWGN.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.6
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• pp.106-112
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• 2001

Unfortunately, the area without economical efficiency, especially the far distance sea, is much lower than that of a urban area-built-up area. It should be promoted the equivalent level to a urban area in the light of future-oriented universal service. Actually, Because propagation environment of mobile communication in the sea is greatly different from that for inland focused on built-up area, a propagation prediction model in the sea should be distinguished from inland-based one. Accordingly, the purpose of this study is to suggest the propagation prediction model for the sea service as a method to minimize unnecessary facilities investment and maintenance caused by additional or new building of a base station. If mobile phone service for far distance sea is provided by expanding limited communication zone of narrow band CDMA mobile communication whose spread band FA is 1.2288MHz. Suggested propagation prediction model includes five parameters to minimize facilities investment of a base station and maximize channel capacity: equivalent line of sight, chip delay by PN code, antenna altitude, power of base station and gain of antennas. Finally, suggested propagation prediction model is simulated and, the results are examined for its utility by comparing with loss of free space.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.1
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• pp.11-21
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• 2008

In conventional video coding, the complexity of encoder is much higher than that of decoder. However, as more needs arises for extremely simple encoder in environments having constrained energy such as sensor network, much investigation has been carried out for eliminating motion prediction/compensation claiming most complexity and energy in encoder. The Wyner-Ziv coding, one of the representative schemes for the problem, reconstructs video at decoder by correcting noise on side information using channel coding technique such as turbo code. Since the encoder generates only parity bits without performing any type of processes extracting correlation information between frames, it has an extremely simple structure. However, turbo decoding errors occur in noisy side information. When there are high-motion or occlusion between frames, more turbo decoding errors appear in reconstructed frame and look like Salt & Pepper noise. This severely deteriorates subjective video quality even though such noise rarely occurs. In this paper, we propose a computationally extremely light encoder based on symbol-level Wyner-Ziv coding technique and a new corresponding decoder which, based on a decision whether a pixel has error or not, applies median filter selectively in order to minimize loss of texture detail from filtering. The proposed method claims extremely low encoder complexity and shows improvements both in subjective quality and PSNR. Our experiments have verified average PSNR gain of up to 0.8dB.