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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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The Journal of Korean Institute of Communications and Information Sciences
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The Korean Institute of Communications and Information Sciences
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Volume & Issues
Volume 27, Issue 12C - Dec 2002
Volume 27, Issue 11C - Nov 2002
Volume 27, Issue 10C - Oct 2002
Volume 27, Issue 10A - Oct 2002
Volume 27, Issue 9C - Sep 2002
Volume 27, Issue 8C - Aug 2002
Volume 27, Issue 8B - Aug 2002
Volume 27, Issue 8A - Aug 2002
Volume 27, Issue 7C - Jul 2002
Volume 27, Issue 7B - Jul 2002
Volume 27, Issue 7A - Jul 2002
Volume 27, Issue 6C - Jun 2002
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Volume 27, Issue 6A - Jun 2002
Volume 27, Issue 5C - May 2002
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Volume 27, Issue 4A - Apr 2002
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Volume 27, Issue 2A - Feb 2002
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Volume 27, Issue 1C - Jan 2002
Volume 27, Issue 1B - Jan 2002
Volume 27, Issue 1A - Jan 2002
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Performance Analysis of Space-Time Transmit Diversity with Adaptive Equalizer in Multipath Channels
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 629~634
The next generation of wireless communication system is expected to provide users with wireless multimedia services such as high speed internet access and wireless mobile computing. Depending on the Quality of Service(QoS) requirements and different applications per user, many wireless communication systems have been proposed. The simple decoding algorithm of space-time block coding is based on the assumption of flat fading channel, where no intersymbol interference exists. In this paper, we propose to combine space-time transmit diversity with adaptive equalizer. The proposed system effectively eliminates interference caused by multipath environment. Simulation results show that the proposed system provides improved bit error rate performance.
Parallel Code Acquisition Techniques in Chip-Asynchronous DS/SS System
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 635~640
We investigate optimal and suboptimal decision rules for parallel code acquisition in chip asynchronous direct-sequence spread-spectrum systems. The conventional decision rule for parallel acquisition is to choose the largest correlator output of a receiver. However, such a scheme is optimum only for chip synchronous models. In this paper, an optimal decision rule is derived based on the maximum-likehood criterion for chip asynchronous models. A simpler suboptimal decision rule is also discussed. The performance of the optimum and suboptimum decision rules is compared to that of the conventional decision rule. Numerical results show that, for chip asynchronous models, both the optimal and suboptimal decision rules outperform the conventional decision rule.
Performance Evaluation of Channel Estimation and Interference Cancellation Techniques for Multiuser with Transmitter Diversity System
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 641~650
Space-Time Block Code(STBC) provides full diversity gains with simple linear processing at the receiver. Interference Cancellation(IC) techniques in system using STBC improve the capacity and performance of wireless systems with co-channel users. Various IC techniques, Minimum Mean-Squared Error(MMSE) and Zero-Forcing(ZF) algorithms in system with STBC were proposed in the literatures in multiuser environment. The performance of these IC techniques were simulated by assuming perfect channel state information(CSI) of multiuser at the receiver. However, in practice it is difficult to know perfect CSI of multiuser at the receiver. Thus, channel estimation scheme is essential at the receiver. Also SNR estimation scheme is required to operate the MMSE IC algorithm. In this paper, we present estimation schemes of CSI and SNR using training sequences. Through extensive computer simulation, we compare and evaluate the performance of IC techniques using the proposed CSI and SNR estimation techniques.
A Study On ECLMS Using Estimated Correlation
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 651~658
Although least mean square(LMS) algorithm is known to one of the most popular algorithm in adaptive signal processing because of the simplicity and the small computation, the choice of the step size reflects a tradeoff between the misadjustment and the speed of adaptation. In this paper, we present a new variable step size LMS algorithm, so-called ECLMS(Estimated correlation LMS), using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multiple-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.
Improvement of Normalized CMA Channel Equalization and Turbo Code for DS-CDMA System
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 659~667
In this dissertation, in the Turbo Code used for error correction coding of the recent digital communication systems, we propose a new S-R interleaver that has the better performance than the existing block interleaver, and the Turbo Decoder that has the parallel concatenated New structure using the MAP algorithm. For real-time voice and video services over the third generation mobile communications, the performance of two proposed methods is analyzed by the reduced decoding delay using the variable decoding method by computer simulation over multipath channels of DS-CDMA system. Also, a Modified NCMA based on conventional NCMA is proposed to improve the channel efficiency in the mobile communication system, and is investigated over the multi-user environment of DS-CDMA system through computer simulation.
Pre-distorter Method Using LUT with 2ι Partition Interpolation in the OFDM System
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 668~675
This paper proposes pre-distorter combined LUT with 2ιpartition interpolation method to reduce nonlinear distortion which was caused by high PAPR and to update LUT quickly. Pre-distorted gain and phase can be found by using LUT which consisted of AM/AM and AM/PM value, and OFDM signal amplitude. The proposed 2ιpartition interpolation can accurately find predistorted gain and phase using bit shift and add component instead of increasing size of LUT which requires increasing the amount of computation. The performance of the proposed method was measured by the difference between HPA input and output characteristics by the LUT size, constellation, SER performance by the HPA, and LUT update error by the HPA characteristic changes. As a result, it is shown that when the size of the LUT is 32 and 64 for 16-QAM and 64-QAM, nonlinear distortion nearly didn't occurred.
Adaptively Compensated-Disparity Prediction Scheme for Stereo Image Compression and Reconstruction
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 676~682
In this paper, an effective stereo image compression and reconstruction technique using a new adaptively compensated-disparity prediction scheme is proposed. That is, by adaptively predicting the mutual correlation between the stereo image using the proposed method, the bandwidth of the stereo input image can be compressed to the level of the conventional 2D image and the predicted image also can be effectively reconstructed using this transmitted reference image and disparity data in the receiver. Especially, in the proposed method, once the feature values are extracted from the input stereo image, then the matching window size for the predicted image reconstruction is adaptively selected in accordance with the magnitude of this feature values. From this adaptive disparity estimation method, reduction of the mismatching probability of the disparity vectors is expected and as a result, the image quality in the reconstructed image can be improved. In addition, from some experiments using the CCETT's stereo images of 'Fichier', 'Manege' and 'Tunnel', it is shown that the proposed method improves the PSNR of the reconstructed image to about 9.08 dB on average by comparing with that of the conventional methods. And also, it is found that there is almost no difference between the original image and the predicted image reconstructed through the proposed method by comparison to that of the conventional methods.
FPGA Implementation of Real-time 2-D Wavelet Image Compressor
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 683~694
In this paper, a digital image compression codec using 2D DWT(Discrete Wavelet Transform) is designed using the FPGA technology for real time operation The implemented image compression codec using wavelet decomposition consists of a wavelet kernel part for wavelet filtering process, a quantizer/huffman coder for quantization and huffman encoding of wavelet coefficients, a memory controller for interface with external memories, a input interface to process image pixels from A/D converter, a output interface for reconstructing huffman codes, which has irregular bit size, into 32-bit data having regular size data, a memory-kernel buffer to arrage data for real time process, a PCI interface part, and some modules for setting timing between each modules. Since the memory mapping method which converts read process of column-direction into read process of the row-direction is used, the read process in the vertical-direction wavelet decomposition is very efficiently processed. Global operation of wavelet codec is synchronized with the field signal of A/D converter. The global hardware process pipeline operation as the unit of field and each field and each field operation is classified as decomposition levels of wavelet transform. The implemented hardware used FPGA hardware resource of 11119(45%) LAB and 28352(9％) ESB in FPGA device of APEX20KC EP20k600CB652-7 and mapped into one FPGA without additional external logic. Also it can process 33 frames(66 fields) per second, so real-time image compression is possible.
Multimedia Watermark Detection Algorithm Based on Bayes Decision Theory
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 695~704
Watermark detection plays a crucial role in multimedia copyright protection and has traditionally been tackled using correlation-based algorithms. However, correlation-based detection is not actually the best choice, as it does not utilize the distributional characteristics of the image being marked. Accordingly, an efficient watermark detection scheme for DWT coefficients is proposed as optimal for non-additive schemes. Based on the statistical decision theory, the proposed method is derived according to Bayes decision theory, the Neyman-Pearson criterion, and the distribution of the DWT coefficients, thereby minimizing the missed detection probability subject to a given false alarm probability. The proposed method was tested in the context of robustness, and the results confirmed the superiority of the proposed technique over conventional correlation-based detection method.
Motion Estimation by Fermat Number Transform
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 705~710
In this paper, we present a method to find motion vectors in frequency domain for video data compression. The proposed algorithm is based on the Format Number Transform (FNT), and it declares the most correlated-block as the best matching block, as opposed to declaring the block with least sum of differences between blocks. We show that the proposed method is equivalent to declaring the block with the minimum L2-norm as the best matching block. Unlike other previous fast algorithms, the time requirement for the proposed algorithm does not defend on the image type for finding the optimum solution.
Performance Evaluation of Differentiated Services to MPEG-4 FGS Video Streaming
The Journal of Korean Institute of Communications and Information Sciences, volume 27, issue 7A, 2002, Pages 711~723
A finer granular scalable (FGS) version of ISO/IEC MPEG-4 video streaming is investigated in this work with the prioritized stream delivery over loss-rate differentiated networks. Our proposed system is focused on the seamless integration of rate adaptation, prioritized packetization, and simplified differentiation for the MPEG-4 FGS video streaming. The proposed system consists of three key components: 1) rate adaptation with scalable source encoding, 2) content-aware prioritized packetization, and 3) loss-based differential forwarding. More specifically, a constant-quality rate adaptation is first achieved by optimally truncating the over-coded FGS stream based on the embedding rate-distortion (R-D) information (obtained from a piecewise linear R-D model). The rate-controlled video stream is then packetized and prioritized according to the loss impact of each packet. Prioritized packets are transmitted over the underlying network, where packets are subject to differentiated dropping and forwarding. By focusing on the end-to-end quality, we establish an effective working conditions for the proposed video streaming and the superior performance is verified by simulated MPEG-4 FGS video streaming.