• The Journal of the Acoustical Society of Korea
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• v.38 no.4
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• pp.371-377
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• 2019

In this paper, we present the sea trial results of long distance underwater acoustic communication in the East Sea, October 2018. One transmitter and sixteen vertical array receivers were used to collect underwater acoustic communication signals, and the maximum distance between the transmitter and the receiver was 90 km. Information was transmitted by BFSK (Binary Frequency Shift Keying) and BCSK (Binary Chirp Shift Keying) method, which are typical digital frequency modulation techniques. Experimental results show that there is no error in all cases at the transmission distance of 60 km, and BFSK and BCSK have average uncoded bit error rate of 0.0197 and 0.0007 respectively without channel coding at 90 km transmission distance.

• Journal of Korea Society of Industrial Information Systems
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• v.27 no.5
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• pp.25-36
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• 2022

This paper proposes a random sign reversal technique in space frequency block code for single carrier modulation. The traditional space time and frequency block coding technique may be confronted with radio environments openly, severe radio hijacking problems are to be overcome. In order to avoid such an open radio issue, random coded data protection technique for space-time block code was proposed, but this algorithm can change channel combination per an Orthogonal Frequency Division Multiplexing block. This kind of slow switching increases the probability that nearby receivers will detect the transmitted data. This paper proposes a fast switching algorithm per data symbols' basis which is a random sign reversal technique in space frequency block code for Single Carrier Modulation. It is shown in simulation that the proposed one has a superior performance in comparison with the performance of the receiver which do not know the random timing sequence of sign reversal.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.1
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• pp.45-50
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• 2023

In order to improve a total system throughput when a base station (BS) transmits data to user equipments (UEs), we propose a scheme to apply multiuser multiple-input multiple-output (MU-MIMO), space-time block coding (STBC), and non-orthogonal multiple access (NOMA) together. An MU-MIMO is applied to two UEs near the BS and STBC is applied to a UE far from the BS because of the difficulty of obtaining the channel information. Also NOMA is applied to differentiate the data from the near UE and the far UE. Two orthognal precoding vectors are used for the MU-MIMO UEs and it causes no interference between them. The STBC technique with the two procoding vectors are also used for the far UEs. Through performance analysis and simulation, we show that the proposed scheme has higher total system throughput than the conventional ones.

• Journal of Broadcast Engineering
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• v.17 no.4
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• pp.640-658
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• 2012

One of the existing terrestrial multi-channel DTV service frameworks, called KoreaView, provides four programs, composed of MPEG-2 based one HD video and H.264/AVC based three SD videos within one single 6MHz frequency bandwidth. However the additional 3 SD videos can not provide enough quality due to its reduced spatial resolution and low target bitrates. In this paper, we propose a framework, which is called a terrestrial multi-channel high quality hybrid DTV service, to overcome such a weakness of KoreaView services. In the proposed framework, the three additional SD videos are encoded based on an H.264/SVC Spatial Base layer, which is compliant with H.264/AVC, and are delivered via broadcasting networks. On the other hand, and the corresponding three additional HD videos are encoded based on an H.264/SVC Spatial Enhancement layer, which are transmitted over broadband networks such as Internet, thus allowing the three additional videos for users with better quality of experience. In order to verify the effectiveness of the proposed framework, various experimental results are provided for real video contents being used for DTV services. First, the experimental results show that, when the SD sequences are encoded by the H.264/SVC Spatial Base layer at a target bitrate of 1.5Mbps, the resulting PSNR values are ranged from 34.5dB to 42.9dB, which is a sufficient level of service quality. Also it is noted that 690kbps-8,200kbps are needed for the HD test sequences when they are encoded in the H.264/SVC Spatial Enhancement layer at similar PSNR values for the same HD sequences encoded by MPEG-2 at a target bitrate of 12 Mbps.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4A
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• pp.332-340
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• 2010

In this paper, we deal with multi-user diversity scheduling methods that transmit simultaneously signals from multiple users using superposition coding multiplexing. These methods can make various scheduling methods be obtained, according to strategies for user selection priority from the first user to the first-following users, strategies for per-user power allocation, and resulting combining strategies. For the first user selection, we consider three strategies such as 1) higher priority for a user with a better channel state, 2) following the proportional fair scheduling (PFS) priority, 3) higher priority for a user with a lower average serving rate. For selection of the first-following users, we consider the identical strategies for the first user selection. However, in the second strategy, we can decide user priorities according to the original PFS ordering, or only once an additional user for power allocation according to the PFS criterion by considering a residual power and inter-user interference. In the strategies for power allocation, we consider two strategies as follows. In the first strategy, it allocates a power to provide a permissible per-user maximum rate. In the second strategy, it allocates a power to provide a required per-user minimum rate, and then it reallocates the residual power to respective users with a rate greater than the required minimum and less than the permissible maximum. We consider three directions for scheduling such as maximizing the sum rate, maximizing the fairness, and maximizing the sum rate while maintaining the PFS fairness. We select the max CIR, max-min fair, and PF scheduling methods as their corresponding reference methods [1 and references therein], and then we choose candidate scheduling methods which performances are similar to or better than those of the corresponding reference methods in terms of the sum rate or the fairness while being better than their corresponding performances in terms of the alternative metric (fairness or sum rate). Through computer simulations, we evaluate the sum rate and Jain’s fairness index (JFI) performances of various scheduling methods according to the number of users.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9C
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• pp.849-860
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• 2007

Because the H.264/AVC standard adopts the variable length coding algorithm, the rate of encoded video bitstream fluctuates a lot as time flows, though its compression efficiency is superior to that of existing standards. When a video is transmitted in real-time over networks with fixed low-bandwidth, it is necessary to control the bit rate which is generated from encoder. Many existing rate control algorithms have been adopting the quadratic rate-distortion model which determines the target bits for each frame. We propose a new rate control algorithm for H.264/AVC video transmission over networks with fixed bandwidth. The proposed algorithm predicts quantization parameter adaptively to reduce video distortion using the quadratic rate-distortion model, which uses the target bit rate and the mean absolute difference for current frame considering pixel difference between macroblocks in the previous and the current frame. On video samples with high motion and scene change cases, experimental results show that (1) the proposed algorithm adapts the encoded bitstream to limited channel capacity, while existing algorithms abruptly excess the limit bit rate; (2) the proposed algorithm improves picture quality with $0.4{\sim}0.9dB$ in average.

• 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 the Institute of Internet, Broadcasting and Communication
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• v.17 no.6
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• pp.69-78
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• 2017

Shannon of the 5G smartphone and Fourier of the signal processing meet in the sampling theorem (2 times the highest frequency 1). In this paper, the initial Shannon Theorem finds the Shannon capacity at the point-to-point, but the 5G shows on the Relay channel that the technology has evolved into Multi Point MIMO. Fourier transforms are signal processing with fixed parameters. We analyzed the performance by proposing a 2N-1 multivariate Fourier-Jacket transform in the multimedia age. In this study, the authors tackle this signal processing complexity issue by proposing a Jacket-based fast method for reducing the precoding/decoding complexity in terms of time computation. Jacket transforms have shown to find applications in signal processing and coding theory. Jacket transforms are defined to be $n{\times}n$ matrices $A=(a_{jk})$ over a field F with the property $AA^{\dot{+}}=nl_n$, where $A^{\dot{+}}$ is the transpose matrix of the element-wise inverse of A, that is, $A^{\dot{+}}=(a^{-1}_{kj})$, which generalise Hadamard transforms and centre weighted Hadamard transforms. In particular, exploiting the Jacket transform properties, the authors propose a new eigenvalue decomposition (EVD) method with application in precoding and decoding of distributive multi-input multi-output channels in relay-based DF cooperative wireless networks in which the transmission is based on using single-symbol decodable space-time block codes. The authors show that the proposed Jacket-based method of EVD has significant reduction in its computational time as compared to the conventional-based EVD method. Performance in terms of computational time reduction is evaluated quantitatively through mathematical analysis and numerical results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.6A
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• pp.533-539
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• 2002

It is well known the fact that turbo lodes has better performance as the number of iteration and the interleaver size increases in the AWGN channel environment. However, as the number of iteration and the interleaver size are increased, it is required much delay and computation for iterative decoding, and caused the flattening effect phenomenon which is very litter BER performance improvement at the arbitrary SNR. In this paper, We proposed the new asymmetric turbo codes, which consist of parallel concatenated turbo codes that use mixed types of component codes with different not only constraint length but also generate polynomial and analyzed its BER performance for log-MAP decoding algorithm with frame size of 128, 256, 512 and 1024 bits, and coding rate of 1/3. As a results of simulation, proposed asymmetric turbo codes verify that its BER performance is superior to conventional symmetric turbo codes. It can be also observed that the flattening effect phenomenon is very reduced by applying the proposed asymmetric turbo codes. It gains respectively 1.7dB ~2.5dB and 2.0dB~2.5dB SNR improvements in the case of short frame(128, 256) and large frame(512, 1024) size for the BER $10_{-4}$>/TEX> region.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.5
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• pp.397-409
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• 2003

This paper describes an implementation of radix-4 trellis parallel architecture and backward state transition control trace back Viterbi decoder, and presents the application results to high speed wireless LAN. The radix-4 parallelized architecture Vietrbi decoder can not only improve the throughput with simple structure, but also have small processing delay time and overhead circuit compared to M-step trellis architecture one. Based on these features, this paper addresses a novel Viterbi decoder which is composed of branch metric computation, architecture of ACS and trace back decoding by sequential control of backward state transition for the implementation of radix-4 trellis parallelized structure. With the proposed architecture, the decoding of variable code rate due to puncturing the base code can easily be implemented by the unified Viterbi decoder. Moreover, any additional circuit and/or peripheral control logic are not required in the proposed decoder architecture. The trace back decoding scheme with backward state transition control can carry out the sequential decoding according to ACS cycle clock without additional circuit for survivor memory control. In order to evaluate the usefulness, the proposed method is applied to channel CODEC of the IEEE 802.11a high speed wireless LAN, and HDL coding simulation results are presented.