• Journal of Korea Society of Industrial Information Systems
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• v.24 no.6
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• pp.1-10
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• 2019

This paper proposes a method to improve the detectable distance by reducing noise to perform a signal processing technique on the received signals. To increase the radar detection range, the noise component of the received signal has to be reduced. The proposed method reduces the noise component by employing two methods. First, the radar signals received with multiple pulses are accumulated. As the number of additions increases, the noise component gradually decreases due to noise randomness. On the other hand, the signal term gradually increases and thus signal to noise ratio increases. Secondly, after converting the accumulated signal into the frequency spectrum, a Least Mean Square (LMS) filter is applied. In the case of the radar received signal, desired signal exists in a specific part and most of the rest is a noise. Therefore, if the LMS filter is applied in the time domain, the noise increases. To prevent this, the LMS filter is applied after converting the received signal into the entire frequency spectrum. The LMS filter output is then transformed into the time domain and then range estimation algorithm is performed. Simulation results show that the proposed scheme reduces the noise component by about 25 dB. The experiment was conducted by comparing the proposed results with the conventional results of the radars held by the Korea Aerospace Research Institute for the international space station.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.3
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• pp.501-507
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• 1999

At the mobile unit for adaptation the propagation environment, it is necessity to adapt very fast the weight coefficient vector of adaptive array antenna In this paper, for the BPSK and BFSK signals with S/I=2, S/N=10 subband adaptive array signal processing method to the linear array antenna using the LMS & the Kalman filter algorithm is proposed. For the 4 elements equidistance linear array antenna systems LMS and Kalman algorithms with subband adaptive instruction principles using the subband signal processing method are adopted and the computer simulation results to the constant amplitude envelope signals such as BPSK or BFSK can be seen that the convergence characteristics of directional patterns and the signal following characteristics are more fast and stable.

• Journal of Digital Convergence
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• v.19 no.10
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• pp.29-38
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• 2021

Purpose: This is a study to design an optimal Edutech teaching-learning platform model that can be linked with various types of LMS to activate e-learning. Methods: For this purpose, the contents of e-learning systems that can be used in the 4th industrial technology of cyber universities and general universities were cross-sectionally analyzed. Results: Cyber universities relied entirely on LMS, and general universities supplemented and utilized different Edutech methods for each professor such as Google Classroom, Zoom video communication, and YouTube in addition to LMS. It was considered that it would be meaningful to provide a minimal algorithm mapping to LMS to share metadata such as Google and YouTube for the Edutech teaching and learning platform model. Conclusion: Therefore, this study is expected to contribute to the improvement of teaching methods and academic achievement through the LMS-based Edutech teaching and learning platform model.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.651-658
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• 2002

Although least mean square(LMS) algorithm is known to one of the most popular algorithm in adaptive signal processing because of the simplicity and the small computation, the choice of the step size reflects a tradeoff between the misadjustment and the speed of adaptation. In this paper, we present a new variable step size LMS algorithm, so-called ECLMS(Estimated correlation LMS), using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multiple-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.917-920
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• 2006

In mobile environments, their is more than one propagation path between each transceiver and Receiver have two or more delayed multipath signals. Delayed multipath signals can cause ISI and Receiver needs a adaptive algorithm to estimate a channel periodically. Also adaptive antenna using adaptive algorithm provide a significant increase in capacity, performance and coverage. This paper describes various LMS algorithm and evaluate the performance of array antenna Base station by using LMS algorithm in the presence of multipath signals and multiple users. As a result of simulation, Adaptive array antenna systems are able to adjust their antenna pattern to select desired signals, and reduce interference.

• Journal of the Korea Institute of Military Science and Technology
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• v.24 no.5
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• pp.545-557
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• 2021

Engine is the main source of vibration that generates unwanted noise and vibration of vehicle chassis. Especially, in submarine applications, radiation of noise signatures can be detected at some distance away from the submarine using a sonar array. Thus quiet operation is crucial for submarine's survivability. This study addresses reduction of the force transmissibility originating from engines and transmitted to hull through engine mounts. An inertial damper, as an actuator of hybrid mount system, is addressed to reduce even further the level of vibration. Narrow band FxLMS algorithms are broadly used to cancel the vibration of engine mount because of its excellent performance of canceling narrow band noise. However, in real active dampers, the maximum displacement of damper mass is kinematically restricted. When the control input signal from the FxLMS algorithm exceeds this limitation, the damper mass will collide with the mechanical stops and results in many problems. Originated from these, a modified narrow band FxLMS algorithm based on the equalizer technique with the maximum allowable displacement of active damper mass is proposed in this study. Some simulation results showed that the propose algorithm is effective to suppress vibration of engine mount while ensuring given displacement constraint.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.11C
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• pp.1107-1119
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• 2006

Data transmission experiences multiplicative distortion in frequency nonselective fading channel. This distortion occurs in OFDM communication channel and can be compensated using an equalizer. Usually, in the case of LMS equalizer, eigenvalue distribution of training signal is enlarged. Large eigenvalue distribution causes principally the performance of a communication system to be deteriorated. This paper proposes a new algorithm that shows the same performance as the existing fast wavelet transform algorithm with less computational complexity. The proposed algorithm was applied to an adaptive equalizer of OFDM communication system. Matlab simulation results show a better performance than the existing one. The proposed algorithm was implemented in VHDL and simulated.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.10a
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• pp.231-239
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• 2009

The filtered-x LMS (FX-LMS) algorithm has been applied to the active noise control (ANC) system in an acoustic duct. This algorithm is designed based on the FIR (finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filteredu LMS algorithm (FU-LMS) based on infinite impulse response (IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.3
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• pp.259-269
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• 1991

In this paper, power line harmonic noise which is the most serious problem in the secondary power distribution line is eliminated and analyzed using adaptive noise cancellers with two adaptive algorithms, LMS and individual tap LMS(ITLMS) algorithm. To testify the improvement of data transmission characteristics made by the adaptive filter with two adaptive algorithms, BER was measured in DS spread spectrum communication system including the noise canceller.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1994.04a
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• pp.16-21
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• 1994

For the systems such as ANC(Active Noise Control) systems having auxiliary path after FIR type adaptive filter, Filtered-X LMS algorithm is effective. However behaviors of this algorithm has not been fully understood. The convergence property of this algorithm depends on not only cross correlation matrix between the filtered signals through model and real auxiliary path state solution of weight vector in Filtered-X LMS algorithm is investigated for under-determined case, over-determined case, and nonsingular case. Also, the convergence speed in case of two tone noise is investigated based on the eigenvalue spread of cross correlation matrix.