• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.402-404
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• 2013

The wireless communication channel different speed by depth of water or salt and it is influenced by multi-path according underwater. In the paper, MIMO(Multi-input-Multi-Output) system used turbo Equalizer combining Equalizer with Turbo codes for data rates by multi-path channel. we proposed and simulated that the Decision-Directed method used for phase offset. The simulation of proposed method show that the bit-error rate performance can be severely affected by phase errors.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.101-102
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• 2006

Similar to Orthogonal Frequency Division Multiplexing (OFDM), a Single Carrier with Frequency Domain Equalization (SC-FDE) system is computationally efficient since equalization is performed on a block of data in the frequency domain. In coherent QAM schemes, the mean phase rotation error caused by the residual carrier frequency offset may lead to serious degradation. When the frequency equalizer is combined with the mean phase error tracking algorithm, its performance can be enhanced noticeably.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.121-127
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• 2010

This paper relates with the improved CMA blind adaptive equalization algorithm which uses the constellation matching method that improve the inverse modelling efficiency of a communication channel compared to the present CMA blind adaptive equalizer. The amplitude distortion can be compensated in the present CMA blind adaptive equalizer which is used for the reduction of intersymbol interference by distortion that generate such as a band limited wireless mobile channel, but in the improved adaptive alogorithm operates with the minimize the amplitude phase distortion in the output of equalizer by applying the cost function that is composition of additional signal constellation matching error terms. In order to evaluation of the inverse modeling efficiency of improved algorithm, the residual intersymbol interference and recovered signal constellation were compared by computer simulation. As a result of comparion of computer simulation, the improved algorithm has a good stability in the residual intersymbol interference in the steady state, but it has a slow convergence rate in the adaptation state in initial state.

• The KIPS Transactions:PartC
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• v.13C no.3 s.106
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• pp.369-374
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• 2006

This paper related with the performance improvement of adaptive equalizer that is a based on the tricepstrum eqalization algorithm by using the received signal. Adaptive equalizer used for the improvement of communication performance, like as high speed, maintain of synchronization, BER, at the receive side in the environment of communication channel of the presence of the aditive noise, phase distortion and frequency selective fading, mainly. It's characteristics are nearly same as the inverse characterstics of the communication channel. In this paper, the TEA algorithm using the HOS and the 16-QAM which is 2-dimensional signaling method for being considered signal was used. For the precoding of 16-QAM singnal in the assignment of the signal costellation, Gray code was used, and the improvement of performance was gained by computer simulation in the residual intersymbol interence and mean squared error which is representive measurement of adaptive equalizer.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.229-236
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• 2010

Phase coherent modulation techniques in underwater acoustic channel can improve bandwidth efficiency and data reliability, but they are made difficult by time-varying intersymbol interference. This paper proposes an adaptive equalizer combined with phase estimator which compensates distortions caused by time-varying multipath and phase variation. The experiment in the East sea demonstrates phase coherent signals are distorted by time-varying multipath propagation and the proposed scheme equalizes them. Bit error rate of BPSK and QPSK are 0.0078 and 0.0376 at 300 meter horizontal distance and 0.0146 and 0.0293 at 1000 meter respectively.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.303-313
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• 2010

This paper describes an acoustic communication receiver structure, which is designed for QPSK (Quadrature Phase Shift Keying) signal with 25 kHz carrier frequency and 5 kHz symbol rate, and takes samples from received signal at 100 kHz sampling rate. Based on the described receiver structure, optimum design parameters, such as number of taps of FF (Feed-Forward) and FB (Feed-Back) filters and forgetting factor of RLS (Recursive Least-Square) algorithm, of joint equalizer are determined to minimize the BER (Bit Error Rate) performance of the joint equalizer output symbols when the acquisition data in shallow water using implemented acoustic transducers is decimated at a rate of 2:1 and then enforced to the input of receiver. The transmission distances are 1.4 km, 2.9 km, and 4.7 km. Analysis results show that the optimum number of taps of FF and FB filters are different according to the distance between source and destination, but the optimum or near optimum value of forgetting factor is 0.997. Therefore, we can reach a conclusion that the proper receiver structure could change the number of taps of FF and FB filters with the fixed forgetting factor 0.997 according to the transmission distance. Another analysis result is that there are an acceptable performance degradation when the 16-tap-length simple filter is used as a low-pass filter of receiver instead of 161-tap-length matched filter.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.5 s.335
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• pp.13-22
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• 2005

IEEE 802.11a is a standard for the high-speed wireless local area network (WLAN), supporting from 6 up to 54 Mbps in a 5 GHz band. We propose a channel equalization algerian and a sampling clock recovery algorithm by utilizing the Viterbi decoder output of the IEEE 802.11a WLAN standard. The proposed channel equalizer adaptively compensates channel variations. The proposed system uses re-encoded Viterbi decoder outputs as reference symbols for the adaptation of the channel equalizer. It also extracts sampling phase information with the Viterbi decoder outputs for fine adjustment of the sampling clock. The proposed sampling clock recovery and equalizer are more robust to noise and frequency selective fading environments than conventional systems using only four pilot samples.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.5 no.1
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• pp.13-20
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• 2005

This paper addresses a new blind channel equalization method using fourth-order cumulants of channel inputs and a three-layer neural network equalizer. The proposed algorithm is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum-phase characteristic of the channel. The transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel inputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple recordering and scaling. By using this estimated deconvolution matrix, which is the inverse of the over-sampled unknown channel, a three-layer neural network equalizer is implemented at the receiver. In simulation studies, the stochastic version of the proposed algorithm is tested with three-ray multi-path channels for on-line operation, and its performance is compared with a method based on conventional second-order statistics. Relatively good results, withe fast convergence speed, are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

• Proceedings of the Korea Committee for Ocean Resources and Engineering Conference
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• 2003.05a
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• pp.319-323
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• 2003

In this paper, we propose a hybrid transceiver for underwater acoustic communication, which allows the system to reduce complexity and increase robustness in time variant underwater channel environments. It is designed in the digital domain except for amplifiers and implemented by using a multiple digital signal processors (DSPs) system. The digital modulation technique is quadrature phase shift keying (QPSK) and frame synchronization is an energy (non-coherent) detection scheme based on the quadrature receiver structure. DSP implementation is based on block data parallel architecture (BDPA). We shaw experimental results in th? underwater anechoic basin at KRISO. The results indicate that the frame synchronization is performed without PLL. Also, we shaw that the adaptive equalizer can compensate frame synchronization error and the correction capability is dependent on the length of equalizer.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.11 s.353
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• pp.141-148
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• 2006

This paper deals with the performance improvement by eliminating loop interference signal and inserting adaptive equalizer for phase compensation in OFDMA based Wibro relay station. The Wibro relay station is used for the extension of communication service area and for throughput improvement of base station. The loop interference is important factor of performance determination of relay station when transmitter and receiver is very closely located. In order to design interference canceller, we generated base-band OFDMA signal and then transmitted the signal along with pilot tones alined with two different combinations for training mode. And then, we generated received fading signal due to the loop interference added noise to the received signal. In the receiver, the transmitted signal is recovered by elimination of the interference signal with channel estimate and compensating phase by adaptive equalizer. The performance improvement was verified by computer simulation which show channel estimation, constellation of signal and BER characteristics according to the variation of SNR ratio.