• 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.

• Progress in Superconductivity
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• v.4 no.2
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• pp.114-120
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• 2003

We employed a method eliminating a temporally partial principal component (PC) of multichannel-recorded neuromagnetic fields for excluding spatially correlated noises from event-evoked signals. The noises in magnetoencephalography (MEG) are considered to be mainly spontaneous neuromagnetic fields which are spatially correlated. In conventional MEG experiments, the amplitude of the spontaneous neuromagnetic field is much lager than that of the evoked signal and the synchronized characteristics of the correlated rhythmic noise makes it possible for us to extract the correlation noises from the evoked signal by means of the general PC analysis. However, the whole-time PC of the fields still contains a little projection component of the evoked signal and the elimination of the PC results in the distortion of the evoked signal. Especially, the distortion will not be negligible when the amplitude of the evoked signal is relatively large or when the evoked signals have a spatially-asymmetrical distribution which does not cancel out the corresponding elements of the covariance matrix. In the period of prestimulus, there are only the spontaneous fields and we can find the pure noise PC that is not including the evoked signal. Besides that, we propose a method, called the extended temporal decorrelation method (ETDM), to suppress the distortion of the noise PC from remanent evoked signal components. In this study, we applied the Partial Principal component elimination method (PPCE) and ETDM to simulated signals and the auditory evoked signals that had been obtained with our homemade 37-channel magnetometer-based SQUID system. We demonstrate here that PPCE and ETDM reduce the number of epochs required in averaging to about half of that required in conventional averaging.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.17 no.1 s.104
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• pp.45-51
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• 2006

In this paper, a new predistortion linearizer for controlling the amplitude of low frequency intermodulation distortion signals is proposed. The low frequency intermodulation distortion(IMD) components are generated by harmonic generator. A vector modulator, modulate fundamental signal with low frequency IMD signals, generates predistortion IMD signals and controls amplitude and phase of them with modulation factors. As a result, this predistorter is suppressed IMD signals of power amplifier effectively. The predistortion linearizer has been manufactured to operate in cellular base-station transmitting band($869{\sim}894\;MHz$). The experimental results show that IMD3 of power amplifier are improved more than 20 dB for CW two-tone signals. Also, it's improved the adjacent channel power ratio(ACPR) more than 10 dB for IS-95 CDMA IFA signals.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.3
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• pp.55-61
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• 2013

This paper compare the performance of S-MMA(Sliced-MultiModulus Algorithm) adaptive equalization algorithm by effect of slice weighting value for the minimization of the distortion and noise in the communication channel.. In the traditional MMA algorithm, the output signal of equalizer and the dispersion constant of transmitting signal is used for calculating the equalizer coefficient, but in S-MMA, the output of equalizer and dispersion constant and the considering the output of decision device by the power of slice constant are used in order to simultaneously compensate the distortion of amplitude and phase distortion. It is confirmed by computer simulation that the slice weighting value affects the performance of adaptive equalization algorithm. The performance index includes the output signal constellation, the residual isi and maximum distortion and MSE that is for the convergence characteristics, the SER according to the signal and noise power ratio at the channel is used. As a result of simulation, the residual isi, maximum distortion and MSE performances are better in the small weighting values. But in SER performance is better in the large weighting values.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.5
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• pp.486-492
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• 2004

In this paper, we newly propose the binary CA-CDMA(constant amplitude CDMA) system using pre-coding method to solve the high PAPR problem caused by multi-user signal transmission in the CDMA system. 4-user CA-CDMA, the basis of proposed binary CA-CDMA system, makes binary output signal for 4 input users. It produces the output of binary(${\pm}$2) amplitude by using a parity signal resulting from the XOR operation of 4 users data. Another sub-channel or more bandwidth is not necessary because it is transmitted together with user data and can be easily recovered in the receiver. The extension of the number of users can be possible by the simple repetition of the basic binary 4-user CA-CDMA. For example, binary 16-user CA-CDMA is made easily by allocating the four 4-user CA-CDMA systems in parallel and leading the four outputs to the fifth 4-user CA-CDMA system as input, because the output signal of each 4-user CA-CDMA is also binary. By the same extension procedure, binary 64 and 256-user CA-CDMA systems can be made with the constant amplitude. As a result, the code rate of this proposed CA-CDMA system is just 1 and binary CA-CDMA does not change the transmission rate with the constant output signal(PAPR = 0 ㏈). Therefore, the power efficiency of the HPA can be maximized without the nonlinear distortion. From the simulation results, it is verified that the conventional CDMA system has multi-level output signal, but the proposed binary CA-CDMA system always produces binary output. And it is also found that the BER of conventional CDMA system is increased by nonlinear HPA, but the BER of proposed binary CA-CDMA system is not changed.

• Journal of Internet Computing and Services
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• v.19 no.6
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• pp.1-7
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• 2018

This paper present an 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) coding method integrated with ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding) and LMS-MPC(Least Mean Square - Multi Pulse Coding) used V/UV/S(Voiced / Unvoiced / Silence) switching, compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. In integrating several methods, it is important to adjust the bit rate of voiced and unvoiced sound source to 8kbps while reducing the distortion of the speech waveform. In adjusting the bit rate of voiced and unvoiced sound source to 8 kbps, the speech waveform can be synthesized efficiently by restoring the individual pitch intervals using multi pulse in the representative interval. I was implemented that the ACLMS-MPC method and evaluate the SNR of APC-LMS in coding condition in 8kbps. As a result, SNR of ACLMS-MPC was 15.0dB for female voice and 14.3dB for male voice respectively. Therefore, I found that ACLMS-MPC was improved by 0.3dB~1.8dB for male voice and 0.3dB~1.6dB for female voice compared to existing MPC, ACFBD-MPC and LMS-MPC. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or internet phone. In the future, I will study the evaluation of the sound quality of 6.9kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.570-573
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• 2004

This paper present a distributed RC lines (URCs) oscillator with sinusoidal output. The frequency of oscillator can be controlled and adjustable by varying an one pole amplifier. The circuit incorporated an gain controller loop for amplitude stabilization with low distortion. The realization of simulation and experimental results are in reasonably good agreement with the theoretical , and very low harmonic distortion. In this circuit can be suitable for LSI process fabrication and the circuit application in electronic communications system.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.122-125
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• 2011

In real environment, background noise exists everywhere and degrades the performance of system. To reduce this distortion, a speech enhancement algorithm can be very useful and variety methods have been proposed. In this paper, we propose a postfilter to improve the performance of optimally modified log-spectral amplitude (OM-LSA) estimator. Proposed algorithm uses the formant postfilter to minimize perceptual distortion caused by background noise. We adjust an emphasizing parameter which is varied by spectral flatness and first reflection coefficient. The performance of the proposed algorithm is evaluated by measuring the log-spectral distance (LSD) and the perceptual evaluation of speech quality (PESQ) score. The test results show the improvement of proposed algorithm compared to conventional OM-LSA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8A
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• pp.760-767
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• 2010

For complex channel blind equalization, this study presents the performance and characteristics of two complex blind information theoretic learning algorithms (ITL) which are based on minimization of Euclidian distance (ED) between probability density functions compared to constant modulus algorithm which is based on mean squared error (MSE) criterion. The complex-valued ED algorithm employing constant modulus error and the complex-valued ED algorithm using a self-generated symbol set are analyzed to have the fact that the cost function of the latter forces the output signal to have correct symbol values and compensate amplitude and phase distortion simultaneously without any phase compensation process. Simulation results through MSE convergence and constellation comparison for severely distorted complex channels show significantly enhanced performance of symbol-point concentration with no phase rotation.

• Proceedings of the KIEE Conference
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• 2008.07a
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• pp.1391-1392
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• 2008

The modulation quality of the I/Q modulator in a wireless transmitter usually affects system performance and it mostly depends on both a nonlinearity and a distortion, from the third order intermodulation(IM3) signal and the imperfection features such as an input amplitude error and a local phase error, respectively. This paper focused on how much the Single Sideband Ratio(SSR), which indicates the signal distortion, changes according to the variation of the imperfection features. Since a desired signal, side band and IM3 signals at the I/Q modulator output are also represented with those power series coefficients and the imperfection features, the effects of the imperfection features on SSR can be clearly analyzed.