• Journal of information and communication convergence engineering
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• v.22 no.2
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• pp.89-97
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• 2024

This research conducted a comparative analysis of two communication systems. The first system utilizes a conventional series configuration consisting of a symbol-rate least mean square (LMS) equalizer followed by a timing recovery loop. The second system introduces an innovative approach that integrates a symbol-rate LMS equalizer and a timing recovery component within a single loop, allowing mutual feedback between the two blocks. In this integrated system, the equalizer also provides timing error information, thereby eliminating the requirement for a separate threshold error detector. This study examines the performance curves of both system configurations. The simulation results revealed that the integrated system may offer improved stability in terms of multiple transmission challenges, including phase and frequency offsets and intersymbol interference. Further analysis and discussion highlight the significant insights and implications of the proposed architecture. Overall, the present findings provide an alternative perspective on the joint implementation of equalization and timing recovery in communication systems.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5B
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• pp.859-870
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• 2000

In this paper, an efficient H/W implementation technique for guard interval in OFDM(Orthogonal Frequency Division Multiplexing) systems is proposed and applied to ASIC chip design of an OFDM-based 25 Mbps wireless ATM modem. In OFDM systems, a cyclic prefix, longer than the largest multipath delay spread, is usually inserted to maintain the orthogonality of subchannels, by making the linear convolution of the channel ok like circular convolution inherent to the discreate Fourier domain, as well as to prevent the ISI(Intersymbol Interference) within the OFDM block. However, the OFDM system using the cyclic prefix requires an additional H/W in transmitter in order to store the original samples and to append the cyclic prefix to the beginning of each block. In this paper, a new approach using a cyclic prefix, even with a significantly lower H/W complexity. Finally, the performance of the proposed approach is demonstrated by applying it to ASIC chip design of an OFDM-based 25 Mbps wireless ATM modem.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7A
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• pp.724-729
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• 2007

In this paper, we consider a joint optimization of transmitter and receiver in additive cyclostationary noise with zero forcing criterion. We assume that the period of the cyclostationary noise is the same as the inverse of the symbol transmission rate and that the noise has a positive-definite autocorrelation function. The data sequence is modeled as a wide-sense stationary colored random process and the channel is modeled as a linear time-invariant system with a frequency selective impulse response. Under these assumptions and a constraint on the average power of the transmitted signal, we derive the optimum transmitter and receiver waveforms that jointly minimizes the mean square error with no intersymbol interference. The simulation results show that the proposed system has a better BER performance than the systems with receiver only optimization and the systems with no transmitter and receiver optimization.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.7
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• pp.3-8
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• 2016

We propose a modified form of the decision-directed least mean square (DD LMS) algorithm that is widely used in the optimization of self-adaptive equalizers, and show the modified version greatly improves the initial convergence properties of the conventional algorithm. Existing DD LMS regards the difference between a equalizer output and a quantization value for it as an error, and achieves an optimization of the equalizer based on minimizing the mean squared error cost function for the equalizer coefficients. This error generating method is useful for binary signal or a single-level signals, however, in the case of multi-level signals, it is not effective in the initialization of the equalizer. The modified DD LMS solves this problem by modifying the error generation. We verified the usefulness and performance of the modified DD LMS through experiments with multi-level signals under distortions due to intersymbol interference and additive noise.

• Journal of IKEEE
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• v.3 no.1 s.4
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• pp.141-149
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• 1999

If signal time duration of MC-CDMA method which has studied for next-generation high-speed data transmission is not sufficiently large compared to delay spread of channel, the performance is degraded by generation of intersymbol interference. In this paper, this problem will solve through serial to parallel convertor and make large sufficiently time duration of signal compared to delay spread of channel and rise variable spectral efficiency through the number of serial to parallel convertor subchannel we will add to parallel frequency diversity block for improve the performance in mobile Communication. Spectral efficiency of the proposed system is counted and compared to spectral efficiency of MC-CDMA and investigated through computer simulations by using multipath Rayleigh fading channel.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.20 no.1
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• pp.101-106
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• 2020

This paper relates with the VS-QE-MMA (Varying Stepsize-Quantized Error-MMA) based on the varying stepsize for improving the equalization performance in the QE-MMA adaptive equalization algorithm that is possible to reducing the intersymbol interference occurred at channel. The SE-MMA use the high-order statistics of transmitted signal and sign of error signal. The QE-MMA was appeared for the H/W implementation easiness substitutes the multiplication and substraction into the shift and substraction in the updating the tap coefficient based on the power-of-two operation of error signal magnitude. The QE-MMA gives degradation of equalization performance due to the such simplification of arithmetic operation. For improving this problem, the propose algorithm, namely VS-QE-MMA, applies the varying stepsize of the nonlinear transformation of error signal. It was confirmed by simulation that the VS-QE-MMA gives better performance than current QE-MMA in the same channel and signal to noise ratio. As a result of simulation, the VS-QE-MMA has more better performance in the every performance index, and it was also confirmed that the varying stepsize effect can be obtained in the greater than 10dB of signal to noise ratio.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.1
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• pp.131-137
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• 2015

This paper relates with the VSS-MMA (Variable Step Size-Multiple Modulus Algorithm) adaptive equalization algorithm which is possible to improving the equalization performance by use the nonlinear fuction of error signal in the MMA adaptive equalization algorithm that are used for the minimization of the intersymbol interference due to the distortion which occurs in the time dispersive channel for the transmission of QAM signal in the system.. In the conventional MMA, we obtains the tap coefficient of adaptive equalizer using the fixed step size, but in the VSS-MMA, we obtains the tap coefficient of adaptive equalizer using the variable step size based on a nonlinear function of error signal. By adapting the variable step size, it was confirmed that the improved equalization performance were obtained by computer simulation. For this, the equalizer output signal constellation, residual isi, maximum distortion, MSE and SER were used in the performace index.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.4
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• pp.173-179
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• 2002

This paper describes an equalizer using the DFE (Decision Feedback Equalizer) structure, CMA (Constant Modulus Algorithm) and LMS (Least Mean Square) algorithms. The DFE structure has better channel adaptive performance and lower BER than the transversal structure. The proposed equalizer can be used for 16/64 QAM modems. We employ high speed multipliers, square logics and many CSAs (Carry Save Adder) for high speed operations. We have developed floating-point models and fixed-point models using the COSSAP$\^$TM/ CAD tool and developed VHDL filter. The proposed equalizer shows low BER in multipath fading channel. We have performed models. From the simulation results, we employ a 12 tap feedback filter and a 8 tap feedforward logic synthesis using the SYNOPSYS$\^$TM/ CAD tool and the SAMSUNG 0.5$\mu\textrm{m}$ standard cell library (STD80) and verified function and timing simulations. The total number of gates is about 130,000.

• Journal of Navigation and Port Research
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• v.37 no.3
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• pp.269-273
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• 2013

Underwater acoustic channels are generally recognized as one of the most difficult communication media because of the multipath propagation, dispersion, and so on. MIMO (Multiple-input multiple-output) techniques have been actively pursued in underwater acoustic communications recently to increase the data rate over the bandwidth-limited channels. The transmit diversity techniques can be applied in this case, and one of them is Alamouti's scheme. In this paper the performances of the transmit diversity technique are evaluated via experiment. Two transmitters and two receivers were used in experiment, and the experiment was performed in indoor water tank. The error rate 5~8% was confirmed in experimental results, and these are the improved values than the error rate 14.8% for SISO(Single-input single-output) channel under same data rate condition.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.11
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• pp.2295-2303
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• 2009

Underwater acoustic(UWA) communication has multipath error because of reflection by sea-level and sea-bottom. The multipath of UWA channel causes signal distortion and error floor. In this paper, we proposed the compensation method of multipath effect using the impulse response of the UWA channel and then analysis the performance of channel coding such as LDPC code, concatenate code. Also we analysed the time-delay errors and estimated amplitude errors of estimated channel information and its affection on the performance. As shown in simulation results, the performance of proposed compensation method is better than the performance of conventional method.