• ETRI Journal
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• v.40 no.3
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• pp.309-317
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• 2018

Single carrier-frequency-division multiple access (SC-FDMA) has been adopted as the uplink transmission standard in fourth-generation cellular networks to facilitate power efficiency transmission in mobile stations. Because multiuser multiple-input multiple-output (MU-MIMO) is a promising technology employed to fully exploit the channel capacity in mobile radio networks, this study investigates the uplink transmission of MU-MIMO SC-FDMA systems with orthogonal space-frequency block codes (SFBCs). It is preferable to minimize the length of the cyclic prefix (CP). In this study, the chained turbo equalization technique with chained turbo estimation is employed in the designed receiver. Chained turbo estimation employs a short training sequence to improve the spectrum efficiency without compromising the estimation accuracy. In this paper, we propose a novel and spectrally efficient iterative joint-channel estimation, multiuser detection, and turbo equalization for an MU-MIMO SC-FDMA system without CP-insertion and with short TR. Some simulation examples are presented for the uplink scenario to demonstrate the effectiveness of the proposed scheme.

• Journal of Electrical Engineering and Technology
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• v.12 no.3
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• pp.1280-1285
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• 2017

In this paper, we consider iterative soft-decision feedback equalizers (sDFE), a.k.a. turbo equalizers for single-carrier transmission. Turbo equalizer takes log-likelihood ratio (LLR) feedback from channel decoder and convert the LLR into symbol estimates and variances to be used for the LLR update at the sDFE. Specifically, we consider both time domain and frequency-domain sDFE and compare their performances. The results shows that frequency-domain sDFE performs better than time-domain one and also that considerable gain can be obtained especially when the channel has deep nulls.

• International Journal of Naval Architecture and Ocean Engineering
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• v.13 no.1
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• pp.617-627
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• 2021

Underwater Acoustic Channels (UAC) have inherent sparse characteristics. The traditional adaptive equalization techniques do not utilize this feature to improve the performance. In this paper we consider the Variable Adaptive Subgradient Projection (V-ASPM) method to derive a new sparse equalization algorithm based on the Minimum Symbol Error Rate (MSER) criterion. Compared with the original MSER algorithm, our proposed scheme adds sparse matrix to the iterative formula, which can assign independent step-sizes to the equalizer taps. How to obtain such proper sparse matrix is also analyzed. On this basis, the selection scheme of the sparse matrix is obtained by combining the variable step-sizes and equalizer sparsity measure. We call the new algorithm Sparse-Control Proportional-MSER (SC-PMSER) equalizer. Finally, the proposed SC-PMSER equalizer is embedded into a turbo receiver, which perform turbo decoding, Digital Phase-Locked Loop (DPLL), time-reversal receiving and multi-reception diversity. Simulation and real-field experimental results show that the proposed algorithm has better performance in convergence speed and Bit Error Rate (BER).

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.7
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• pp.1-5
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• 2012

At the present time when the limits of the magnetic storage systems appear, the holographic data storage (HDS) devices with high data transfer rate and high recording density are emerging as attractive candidates for next-generation optical storage devices. In this paper, to effectively improve the detection performance that is degraded by the two-dimensional inter-symbol interference under the HDS channel environment and the pixel misalignment, an iterative two-dimensional equalization scheme is proposed based on the contraction mapping theorem. In order to evaluate the performance of the proposed scheme, for various holographic channel environments we measure the BER performance using computer simulation and compare the proposed one with the conventional threshold detection scheme, which verifies the superiority of the proposed scheme.

• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.124-130
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• 2013

The performance of underwater acoustic communication system is sensitive to the ISI(Inter-Symbol Interference) due to delay spread develop of multipath signal propagation. The equalizer is used to combat the ISI. In this paper, the performances of underwater acoustic communication with turbo equalizer were evaluated by real data collected in Korean littoral sea. As a result, when one iterative decoding using turbo equalizer is applied, the performance was improved 1.5 dB than the case of the non-iterative equalizer at BER $10^{-4}$. In the case of two or three iterations the performance was enhanced about 3.5 dB, but the performance wasn't improved any more in the case of more than three times.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.28 no.11
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• pp.837-842
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• 2017

In this paper, as a basic study for combination of turbo equalizer and 4D-8PSK-TCM(4 Dimensional 8 Phase Shift Keying Trellis Coded Modulation) system, which is recommended for X-band band satellite communication in CCSDS, a system combining 8PSK-TCM system and turbo equalizer has designed and performance evaluation of the system has performed. BER performance of 8PSK and 8PSK-TCM systems has evaluated and analyzed in AWGN, Proakis B and Proakis C channels in detail. As results of simulation, BER performance of 8PSK-TCM system is better than 8PSK system in AWGN environment. Furthermore, it can be confirmed that the system combining the 8PSK-TCM system with the turbo equalizer can further improve the BER performance through iterative equalization in the ISI channel environment. The 8PSK-TCM system combined with the turbo equalizer requires SNR of 5.9 dB for BER performance of $10^{-3}$ with 5 iterative equalization in Proakis B and SNR of 9.1 dB for BER of $10^{-2}$ with 5 iterative equalization in Proakis C.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5C
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• pp.295-301
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• 2011

In [1], it has been shown that the energy spreading transform (EST) based iterative equalizer (IE) could enhance its performance by improving the reliability of the decision feedback symbols without the help of the complicated channel decoder. In the matched filter (MF) based IE proposed in [1], however, its feedforward filter (FFF) has been designed in the frequency domain while its feedback filter (FBF) in the time domain. So its complexity increases proportional to the channel memory length. To solve this problem, in this paper, both FFF and FBF are designed in the frequency domain. This enables the proposed frequency domain IE (FD-IE) to achieve the lower complexity over the conventional method in the highly dispersive channel. In addition, simulation results demonstrate that the BER performance of the proposed method is the same as the conventional.

• The Journal of the Acoustical Society of Korea
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• v.34 no.4
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• pp.295-302
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• 2015

Acoustic channels are characterized by long multipath spreads that cause inter-symbol interference. The way in which this fact influences the design of the receiver structure is considered. To satisfy performance and throughput, we presented consecutive iterative BCJR (Bahl, Cocke, Jelinek, Raviv) equalization to improve the performance and throughput. To achieve low error performance, we resort to powerful BCJR equalization algorithms that iteratively update probabilistic information between inner decoder and outer decoder. Also, to achieve high throughput, we divide long packet into consecutive small packets, and the estimate channel information of previous packets are compensated to next packets. Based on experimental channel response, we confirmed that the performance is improved for long length packet size.

• International Journal of Naval Architecture and Ocean Engineering
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• v.7 no.2
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• pp.426-434
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• 2015

In this paper we proposed a dada-aided Doppler estimation method for underwater acoustic communication. The training sequence is non-dedicate, hence it can be designed for Doppler estimation as well as channel equalization. We assume the channel has been equalized and consider only flat-fading channel. First, based on the training symbols the theoretical received sequence is composed. Next the least square principle is applied to build the objective function, which minimizes the error between the composed and the actual received signal. Then an iterative approach is applied to solve the least square problem. The proposed approach involves an outer loop and inner loop, which resolve the channel gain and Doppler coefficient, respectively. The theoretical performance bound, i.e. the Cramer-Rao Lower Bound (CRLB) of estimation is also derived. Computer simulations results show that the proposed algorithm achieves the CRLB in medium to high SNR cases.

• Journal of Communications and Networks
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• v.5 no.2
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• pp.157-166
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• 2003

This paper deals with the problem of channel identification for Single Input Multiple Output (SIMO) slow fading channels using clustering algorithms. Due to the intrinsic memory of the discrete-time model of the channel, over short observation periods, the received data vectors of the SIMO model are spread in clusters because of the AWGN noise. Each cluster is practically centered around the ideal channel output labels without noise and the noisy received vectors are distributed according to a multivariate Gaussian distribution. Starting from the Markov SIMO channel model, simultaneous maximum ikelihood estimation of the input vector and the channel coefficients reduce to one of obtaining the values of this pair that minimizes the sum of the Euclidean norms between the received and the estimated output vectors. Viterbi algorithm can be used for this purpose provided the trellis diagram of the Markov model can be labeled with the noiseless channel outputs. The problem of identification of the ideal channel outputs, which is the focus of this paper, is then equivalent to designing a Vector Quantizer (VQ) from a training set corresponding to the observed noisy channel outputs. The Linde-Buzo-Gray (LBG)-type clustering algorithms [1] could be used to obtain the noiseless channel output labels from the noisy received vectors. One problem with the use of such algorithms for blind time-varying channel identification is the codebook initialization. This paper looks at two critical issues with regards to the use of VQ for channel identification. The first has to deal with the applicability of this technique in general; we present theoretical results for the conditions under which the technique may be applicable. The second aims at overcoming the codebook initialization problem by proposing a novel approach which attempts to make the first phase of the channel estimation faster than the classical codebook initialization methods. Sample simulation results are provided confirming the effectiveness of the proposed initialization technique.