• Transactions of the Korean Society of Automotive Engineers
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• v.2 no.6
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• pp.29-37
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• 1994

Active noise control of random noise which propatate in the vehicle cabin as a form of spherical wave is the target of this study. In the previous study, the adaptive algorithm for adaptive controller is presented for the application in active noise control system. And for the preliminary study of adaptive active noise control in vehicle cabin as a real system, a computer simulation is performed on the effectiveness of the adaptive algorithm in the amplitude of the pressure fluctuation. This work studies the implementation of multi-channel feedforward adaptive algorithm for the reduction of the noise inside a vehicle cabin using a number of secondary sources derived by adaptive filtering of reference noise source. Multi-channel adaptive feedforward algorithm are verified in numerical simulation and semi-experimental justification of developed system is made on a domestic passenger car. In the results of semi-experimental study, the noise of specific region in the interior of automobile are reduced for the appreciabe sound pressure level in the operating engine rpm and finally this study suggests the capabilities of the real time active noise control in 3 dimensional acoustic fields.

• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.2007-2008
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• 2011

In this paper, we implemented active noise control systems for active barriers with attenuation of road noise. And then we constructed by multi-channel audio systems and DAQ part, high performance DSP H/W. Active noise control firmware program was implemented for multi-channel off-line/on-line estimation methods for secondary path transfer functions and FIR/IIR filter structure main noise control algorithms. To evaluate performance of implemented systems, active barriers test bed was implemented and several experiments was performed by various noise cases.

• JSTS:Journal of Semiconductor Technology and Science
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• v.7 no.2
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• pp.88-93
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• 2007

In this paper, a simple and improved noise parameter model of RF MOSFETs is developed and verified. Based on the analytical model of channel thermal noise, closed form expressions for four noise parameters are developed from proposed equivalent small signal circuit. The modeling results show a excellent agreement with the measured data of $0.13{\mu}m$ CMOS devices.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.28A no.1
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• pp.1-7
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• 1991

Degradation of communication quality is due to transmission error rather than quantization noise when DPCM signal is transmitted over a heavily noise channel. The communication quality can be improved by employing an error correcting code to the DPCM signal transmission over such a channel. We considered both quantization noise and transmission error simultaneously in evaluating the signal to noise ratio. To efficiently improve the signal to noise ratio, we analyze the unequal symbol error probability of convolutional code, and encoded to more protect significant symbols from channel errors than least ones, so that the signal to noise ratio is improved. We derived related formulas and also made computer simulations.

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.367-372
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• 2012

This paper introduces noise masking algorithm for 64 channel digital hearing aid. 125 Hz spectral resolution is maintained for 64 channels from 125 Hz to 8000 Hz. The same spectral masking processing effects as the cochlea are considered and applied for the present hearing aid noise reduction processing algorithm. Theoretical algorithm has been ported into assembler language program software and been embedded into a DSP IC chip for the digital hearing aid. Some parts of noise masking software program code are explained, and the results of the real-time noise reduction are verified by electro-acoustic measurements.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.280-287
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• 2016

In this paper, we propose a normalization algorithm that can be applied to adaptive filters for multi-channel active noise control. The FxLMS (Filtered-x Least Mean Square) algorithm for the single-channel active noise control can be normalized in the same way as the NLMS (Normalized Least Mean Square) algorithm, whereas in case of the multi-channel active noise control, the single-channel normalization for the FxLMS algorithm cannot be extended to the normalization for the multi-channel FxLMS algorithm straightforwardly. First, we adopt a generalized normalization algorithm for the multi-channel FxLMS algorithm based on the principle of minimal disturbance and then, proposed a normalized algorithm considering only diagonal elements to avoid computation for matrix inversion. We carried out performance comparisons of the proposed algorithm with other algorithms without normalization. It is shown that the proposed algorithm presents better convergence characteristics under non-stationary environments.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.1
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• pp.40-52
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• 2020

In this paper, we consider a precoder design problem for multiuser interference channel. Most of the conventional schemes for precoder design utilize a signal-to-interference-plus-noise ratio (SINR) as a cost function. However, since the SINR metric of a desired transmitter-receiver pair is a function of precoding vectors of all transmitters in the multiuser interference channel, an analytic closed-form solution is not available for the precoding vector of a desired transmitter that maximizes the SINR metric. To eliminate coupling between the precoding vectors of all transmitters and to find a closed-form solution for the precoding vector of the desired transmitter, we use a signal-to-leakage-plus-noise ratio (SLNR) instead as a cost function because the SLNR at a transmitter is a function of the precoding vector of the desired transmitter only. In addition, channel estimation errors for undesired links are considered when designing the precoding vector because they are inevitable in a multiuser interference channel. In this case, we propose a design scheme for the precoding vector that is robust to the channel estimation error. In the proposed scheme, the precoding vector is designed to maximize the worst-case SLNR. Through computer simulation, we show that the proposed scheme has better performance than the conventional scheme in terms of SLNR, SINR, and sum rate of all users.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.2
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• pp.252-256
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• 2011

We analyze the channel capacity with considering the effect of the loading conditions of indoor PLC networks on channel state information. We consider various numbers of load for two kinds of the networks with regular length branches and a deployed network of indoor PLC. For calculating the channel capacity degradation, two noise scenarios and impedances are considered. From the simulation results, we suggest the robust regression lines for modeling the channel capacity degradation. In the cases of 0 $\Omega$ and $Z_0$ loads, natural log and linear function curve show the best goodness of fit, respectively. For the deployed indoor PLC network with 0 $\Omega$ loads, compared with the networks with regular length branches, the goodness of fit decreases by the amount of 0.12 and 0.15 for low noise and high noise scenarios, respectively. Using the regression lines, we can estimate the channel capacity degradation without measurement.

• Proceedings of the KIEE Conference
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• 1987.07b
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• pp.854-858
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• 1987

This paper considers impulsive noise which produce burst error in high speed(approx.160Kbps) data transmission like ISDN(Integrated Servise Digital Network) using PSTN(Public Switching Telephone Network). To begin with, we obtains the transfer function of subscriber line to calculate the variation of bandwidth when the gain of receiver is fixed and channel capacity of non-gaussian channel in upper-and lower bound, and evaluates the transmission capability. In this paper compares channel capacity bounds which obtains when probability density function of impulsive noise is Laplacian distribution function with impulsive noise generated by waveform synthesier.

• 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.