• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.4
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• pp.123-128
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• 2017

This paper relates with the Hybrid-SMMA adaptive equalization algorithm that is possible to improve the equalization performance based on adaptive modulus and adaptive stepsize which is propotional to the output power of equalizer in the current SMMA (Sliced Multi Modulus Algorithm). The fixed statistic modulus of transmitted signal is used in current SMMA algorithm in order to generate the error signal for updating the tap coefficient of equalizer. The proposed Hybrid-SMMA based on the adaptive modulus which is propotion to the equalizer output signal power and adaptive stepsize which is fuction of the nonlinearties of error signal. The computer simulation was performed in order to confirm the improved equalization performance of proposed algorithm. As a result of computer simulation, the proposed Hybrid-SMMA has fairly good in every performance than the current SMMA.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.4
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• pp.427-432
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• 2010

To reduce the outage probability and to increase the transmission capacity, the importance of repeaters in cellular systems is increasing. But a RF(Radio Frequency) repeater has a problem that the output of the transmit antenna is partially feedback to the receive antenna, which is feedback interference. In this paper, we proposed adaptive Sign-Sign LMS(Least Mean Square) algorithm using correlations for the performance enhancement of RF repeater. The weight vector is updated by using sign of input signal and error signal to the least squared error of the conventional algorithms. When compared with the conventional method, the proposed canceller achieves the maximum 10 dB performance gain in terms of the MSE(Mean Square Error).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1C
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• pp.96-102
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• 2010

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.1
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• pp.85-90
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• 2018

This paper compares the equalization algorithm of CR-MMA (Constellation Reduction-Multi Modulus Algorithm) and FC-MMA (Fast Convergence-Multi Modulus Algorithm) for the compensation of channel's distortion in transmitting the 2 dimensional 16-QAM signals. The CR-MMA adaptive equalizer use the error signal by reduce the nonconstant modulus signal to constant modulus signal in order to updates the tap coefficient, and the FC-MMA adptive equalizer use the error signal applying the modified dispersion constant considering the number of symbol level instead of signal point reduction. These two algorithm are known to its superior to the convergence characteristic among the MMA series equalizer. In this paper, the other equalization performance including the convergence characteristic was compared by computer simulation. As a result of computer simulation, FC-MMA has more good performance in the residual isi, maximum distortion and SER performance than CR-MMA, but not in convergence speed.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.91-97
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• 2016

This paper evaluates the equalization algorithm of FC-MMA (Fast Convergence-Multi Modulus Algorithm) and RMMA (Region based Multi Modulus Algorithm) for the compensation of intersymbol interference which is due to the distortion of communication channel. In order to obtain the error signal for adaptive equalization, the FC-MMA use the modified dispersion constant considering the number of signal symbol, the RMMA separates the 4 region which every symbol points are located, then reduce the symbol point based on this region into constant modulus symbol point. By applying the different principle in order to get the error signal for the updating the tap coefficient of adaptive equalizer, it has the different equalization performance by these error signal. The computer simulation was performed in order to compare the different equalization performance in this paper. The performance index includes the output signal constellation, the residual isi and maximum distortion that is for the convergence characteristics, the SER. As a result of computer simulation, RMMA has more good performance in the residual isi, maximum distortion after in steady state and SER performance than FC-MMA, but not in convergence speed to reach the steady state.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.77-82
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• 2017

This paper related with the VSS_SE-MMA (Varying Step Size_Signed Error-MMA) which possible to improving the equalization performance that employing the varying adaptive step size based on the nonlinearities of error signal of SE-MMA (Signed Error-MMA), compensates the intersymbol interference by distortion occurs at the communication channel, in the transmitting the spectral efficient nonconstant modulus signal such as 16-QAM. The SE-MMA appeared to the reducing the computational arithematic operation using the polarity of error signal in the updating the tap coefficient of present MMA adaptive equalizer, but have a problem of equalization performance degradation. The VSS_SE-MMA improves the problem of such SE-MMA, using the varying step size consider the error signal in the update the adaptive equalizer tap coefficient, and its improved performance were confirmed by simulation. For this, the output signal constellation of equalizer, the residual isi and maximum distortion, MSE and SER were applied. As a result of computer simulation, it was confirmed that the VSS_SE-MMA algorithm has nearly same in convergence speed and has more good performance in every performance index at the steady state.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.3
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• pp.419-426
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• 2017

This paper describes an acoustic echo canceller with double-talk using BSS(: Blind Signal Separation) based on the geometric concept. The acoustic echo canceller may be deteriorated or diverged during the double-talk period. So we use the blind signal separation to detect the double talking by separating the near-end speech signal from the mixed microphone signal. In the closed reverberation environment, the blind signal separation extracts the near-end signal from unknown signals with the transformation and rotation based on the geometric concept. By this method, the acoustic echo canceller operates irrespective of double-talking. We verified performances of the proposed acoustic echo canceller by computer simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods thoroughly, and operates stably in the normal state without diverging of coefficients after ending the double-talking.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.979-986
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• 2018

This paper deals with acoustic echo cancellation using blind signal separation method. This method does not degrade the echo cancellation performance even during double-talk. In the closed echo environment, the mixing model of acoustic signals is multi-channel, so the convolutive blind signal separation method is applied and the mixing coefficients are calculated by using the feedback model without directly calculating the separation coefficients for signal separation. The coefficient update is performed by iterative calculations based on the second-order statistical properties, thus estimates the near-end speech. A number of simulations have been performed to verify the performance of the proposed blind signal separation method. The simulation results show that the acoustic echo canceller using this method operates safely regardless of the presence of double-talk, and the PESQ is improved by 0.6 point compared with the general adaptive FIR filter structure.

• The Journal of the Korea institute of electronic communication sciences
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• v.17 no.3
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• pp.409-416
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• 2022

This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

• Journal of the Institute of Convergence Signal Processing
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• v.8 no.4
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• pp.297-306
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• 2007

Model reference control is control method such that overall response of a plant plus controller approaches that of a given reference model. The reference model provides desired trajectory the plant should follow. There are many kinds of control methods in MRC. However, this paper focuses on Model Reference Sliding Mode Control. The plant of these controls is an uncertain and linear system varying in time, of second order, and with SISO. In this paper, a design scheme of reference model is proposed for MRSMC. The scheme determines reference model based on the information on bounded control inputs, reference inputs and system parameters. It is used to choose a Fixed Reference Model in the process of controller design, to update Variable Reference Model when stepwise reference inputs change and to update Instant Reference Model at every sampling time. The simulation results show that the proposed method yields better control performance than the conventional MRC subject to the stepwise reference input when applied to the position control system for motor system.