• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6A
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• pp.920-930
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• 1999

Many researches on the multi-level QAM(Quadrature Amplitude Modulation) which is known to be a promising digital data transmission method for efficient use of channel bandwidth have been carried on, and their applications to various real fields are now being in progress. However, in the band-limited communication systems, each transmitted symbols is distorted by the ISI(Intersymbol Interference) and the phase error. Therefore, an equalizer and a carrier recovery must be taken into considerations to attenuate the effects of these distortions in the receiver. This paper presents an effective receiver structures that is applicable to the multi-level QAM. The proposed receiver system is consisted of an equalizer with Godard’s blind algorithm and a carrier recovery circuit. The phase error is estimated with a $\theta$-matching algorithm and is used in the carrier recovery to recover the correct phase. The simulation results are included to evaluate performance of the proposed receiver system for the various channel models.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.1
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• pp.85-91
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• 2014

This paper relates with the VSDA (Variable stepsize Square contour Decision directed Algorithm) adaptive equalization algorithm that is used for the minimization of the intersymbol interference due to the distortion which occurs in the time dispersive channel for the transmission of 16-QAM signal.. In the conventional SCA, it is possible to compensates the amplitude and phase in the received signal that are mixed with the intersymbol interference by the constellatin dependent constant by using the 2nd order statistics of the transmitted signal. But in the VSDA, it is possible to the increasing the equalization performance by adding the concept of distance adjusted approach for constellation matching and the cost function of decision directed. We compare the performance of VSDA and SCA algorithm by the computer simulation. For this, the equalizer output signal constellation, residual isi, maximum distortion and MSE were used in the performace index. As a result of computer simulation, the VSDA algorithm has better than the SCA in convergence speed, but it gives nearly same equalization performance in other index.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.2
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• pp.53-57
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• 2002

Recently, digital communication system becomes high speed, as communication demand dose not only increases sharply, but an image, voice various kinds data also comes multimedia. In transmitting data at a high speed, the main problem is fading by multipath. A linear or nonlinear distortion arise In multipath channel fading from ISI(Intersymbol Interference). For restoring this distorted signal, A lot of equalizer and adaptive algorithm is introduced. This paper compares and analysises, for improving communication quality in channel which is long delay spread, performance of decision feedback equalizer by RLS algorithm, a channel estimation by RLS-MLSE and viterbi equalizer Particularly, there Is exactly channel estimation of impluse response and excellent property of equalization about channel, which delay spread is long impluse response comparatively and is property of non-minimun phase.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.3
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• pp.63-69
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• 2013

This paper related with the performance of SE-MMA (Signed-Error MMA) that is the reduction of computational operation number in algorithm than MMA blind eualization algorithm which are possible to elimination of intersymbol interferance in the band limited and time dispersive nonlinear communication channel. In MMA algorithm which are possible to reduction of amplitude and phase rotation by intersymbol interference that is occurred in channel without using the training sequence, it uses the error signal that is the difference of the equalizer output and constant modulus, the statisticlly characteristic of transmitted signal. But in SE-MMA, it uses the polarity of the error signal, then it is possible to reduce the updating the tap coefficient and to simplify the H/W implementation. The computer simulation were performed in order to compare the performance of SE-MMA and conventional MMA algorithm. For this, the recovered signal constellation that is the output of the equalizer, the convergence performance by MSE, MD (maximum distortion) and residual isi characteristic learning curve, SER were used. As a result of simulation, the SE-MMA has more fast convergence speed than the MMA. But in the other index after reaching the seady state, it gives more worst performance values in the used index.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.69-72
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• 1999

In this paper, an efficient time domain equalization algorithm for discrete wavelet multitone(DWMT) data transmission is developed. In this algorithm, the time domain equalizer(TEQ) consists of two stages, i.e., the channel impulse response shortening equalizer(TEQ-S) in the first stage and the channel frequency flattening equalizer(TEQ-F) in the second stage. TEQ-S reduces the length of transmission channel impulse response to decrease intersymbol interference(ISI) followed by TEQ-F that enhances the channel frequency response characteristics to the level of an ideal channel, hence diminishes the bit error rate. TEQ-S is implemented using the least-squares(LS) method, while TEQ-F is designed by using the least mean-square(LMS) algorithm. Since DWMT system also requires of the frequency domain equalizer in order to further reduce ICI and ISI the hardware complexity is an another concern. However, by adopting an well designed and trained TEQ, the hardware complexity of the whole DWMT system can be greatly reduced.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.9B
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• pp.1226-1235
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• 2001

The convergence analysis of the improved multiplication free adaptive digital filter (IMADF) with a fractionally-spaced equalizer (FSE) to remove the intersymbol interference (ISI) in fast data transmission is presented. The IMADF structure use the one-step predicted filter in the multiplication-free adaptive digital filter (MADF) structure using the DPCM and Sign algorithm. In the experimental results, the IMADF algorithm has reduced the computational complexity by use of only the addition operation without a multiplier. Also, under the condition of identical stationary-state error, it could obtain the stabled convergence characteristic that the IMADF algorithm is almost same as the sign algorithm, but is better than the MADF algorithm. Here, this algorithm has effective characteristics when the correlation of the input signal is highly.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2467-2469
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• 1998

We examine the performance of high-speed non-directed infrared links using pulse-position modulation (PPM) with trellis-coded modulation (TCM) to mitigate the effects of multipath-induced intersymbol interference(ISI) on such links. Using the measured characteristics of multipath indoor infrared channels, we calculate BER curves and ISI power penalties for maximum-likelihood sequence detection of 8-PPM links using rate-2/3 trellis codes operation at 10 and 30Mb/s. Our results indicate that the suggested decoder of PPM with TCM is a very promising decoder on high-speed non-directed infrared channels.

• ETRI Journal
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• v.29 no.5
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• pp.607-613
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• 2007

Most signal-to-noise ratio (SNR) estimation techniques in digital communication channels derive the SNR estimates solely from samples of the received signal after the matched filter. They are based on symbol SNR and assume perfect synchronization and intersymbol interference (ISI)-free symbols. In severe channel distortion where ISI is significant, the performance of these estimators badly deteriorates. We propose an SNR estimator which can operate on data samples collected at the front-end of a receiver or at the input to the decision device. This will relax the restrictions over channel distortions and help extend the application of SNR estimators beyond system monitoring. The proposed estimator uses the characteristics of the second order moments of the additive white Gaussian noise digital communication channel and a linear predictor based on the modified-covariance algorithm in estimating the SNR value. The performance of the proposed technique is investigated and compared with other in-service SNR estimators in digital communication channels. The simulated performance is also compared to the Cram$\acute{e}$r-Rao bound as derived at the input of the decision circuit.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.2
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• pp.137-146
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• 2002

In order to remove the intersymbol interference(ISI) phenomenon in data transmission channel, the structure and convergence analysis of the improved multiplication free adaptive digital filter(IMADF) is presented. Under conditions of zero-mean, wide-sense stationary and white Gaussian noise, it is shown that this paper analyze the convergence characteristics of the IMADF with a fractionally-spaced equalizer(FSE). In the experimental results, the convergence characteristics of the IMADF algorithm is almost same as the sign algorithm, but is better than the MADF algorithm. Here, this algorithm has useful characteristics when the correlation of the input signal is highly.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.1978-1982
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• 2001

In a Multi-carrier modulation system, CP (Cyclic prefix) is inserted in the transmit tame in order to eliminate the ISI (Intersymbol Interference) and ICI (Interchannel Interference) caused by delay spread of a received signal, which in rum degrades the throughput of the system. TEQ (Time-domain equalizer) improves the system throughput by shortening the CIR (Channel Impulse Response) time and maintaining the CP length to the minimum regardless of the channel condition. In this paper, a new MMSE (Minimum Mean Square Error) TEQ algorithm is proposed and its performance is analyzed in order to speed up computing the optimum tap coefficients of the equalizer by employing Newton method.