• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.2
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• pp.407-416
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• 2000

Adaptive digital filtering and noise cancelling technique using a tree structured filter bank are presented to reduce a undesirable aliasing due to the decimation of filtered output and improve the performance in terms of mean-square error and the convergence speed using a aliasing canceller. A signal is split into two subband by analysis filter bank and decimated by decimator and reconstructed by interpolation technique and synthesis filter bank. A variable step-size LMS algorithm is used to improve the convergence speed in case of existing the measurement noise in desired input of filter. It is shown by computer simulation that the proposed subband structure in this paper is superior to conventional subband filter structure in terms of mean-square error and convergence speed.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.17 no.11
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• pp.1077-1085
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• 2007

Typically, three types of levitation technologies are applied to magnetic levitation systems: electromagnetic suspension, electrodynamic suspension, and hybrid electromagnetic suspension. A Halbach array is a special arrangement of permanent magnets which augments the magnetic field on one side of the device while cancelling the field to near zero on the other side. The application of this Halbach array magnet to the electrodynamic suspension has been recently studied in order to increase the levitation capability. This paper is focused on an analytical method of the magnetic levitation system using Halbach array magnet. The suitability of the proposed method is verified with comparing to the finite element method. In addition, dynamic stability of the magnetic levitation system is discussed. From this study, it is confirmed that the proposed method provides a reasonable solution with less computation time compared to the finite element method and the magnetic levitation system using Halbach array magnet is stable dynamically.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.7-10
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• 2004

This paper describes techniques to implement low-cost adapt ive Pipelined LMS filter for ASIC implement ions of high communication receivers. Power consumpiton can be reduced using a careful selection of architectural, algorithmic, and VLSI circuit techlifue A Pipelined architecture for the strength-reduced algorithm is then developed via the relaxed look-ahead transformation. This technique, which is an approximation of the conventional look-ahead compution, maintains the functionality of the algorithm rather than the input-output behavior Convergence maiysis of the Proposed architecture has been presented and support via simulation results. The resulting pipelined adaptive filter achives a higher though put requires lower power as compared to the filter using the serial algorithm.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.272-279
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• 2016

In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 1998.10a
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• pp.282-288
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• 1998

본 논문에서는 뉴로-퍼지 제어기를 이용하여 밀폐공간에서의 능동 소음 제어기를 구현하였다. 능동 소음 제어기는 잡음에 의하여 왜곡된 신호로부터 잡음을 제거하여 원 신호를 복원하는 제어시스템이다. 일반적으로 잡음의 특성이 시간에 따라 변화라고, 전달특성이 비선형적이므로 고정된 제어기에 의해서는 제어할 수 없다. 이 논문에서는 뉴로-퍼지 제어기를 사용하여 파라미터를 오차 역전파 학습을 통하여 변화시킴으로써 잡응의 특성에 효과적을 적응하는 능동 소음 제어기를 구성하였다. 원신호는 음성신호를 사용하였으며 실제 소음과 소음 전달경로인 1차경로를 통과한 왜곡된 소음은 실험에 의해 얻은 데이터를 사용하였다. 제어신호의 전달경로인 2차경로는 100[kHz]에서 1[kHz]까지의 주파수 특성을 고려하여 curve fitting 방법을 사용하여 4차로 모델링한 결과를 사용하였다. 제안한 능동 소음 제어기의 성능을 시뮬레이션을 통하여 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.10A
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• pp.1540-1547
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• 2000

In this paper, the performance of the MMSE multiuser detector is compared with the conventional detector with respect to the signal-to-noise ratio, the number of users and the Nakagami parameter under AWGN as well as Nakagami fading channel. The results show that the MMSE multiuser detector is superior to the conventional detector with respect to cancelling the multiple access interference. However, its drawback is the hardware's complexity. To solve this drawback, the advanced MMSE multiuser detector is presented, and its performance is analyzed. The number of taps in the advanced MMSE multiuser are independent of the processing gain. Thus, the system engineer can choose the appropriate number of taps in the detector to achieve a optimal trade-off between the hardware complexity and the performance of system.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.216-222
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• 2016

Generation of a quiet zone in noisy environment is undoubtedly of considerable realistic significance. This paper describes development and implementation of a multichannel real-time active noise control (ANC) system for 3 dimensional noisy environment to enhance the quiet zone performance in terms of size and noise cancellation gain. The proposed ANC system employes a multichannel delay-compensated filtered-X least mean square (FXLMS) algorithm; its real-time implementation is designed in TMS320C6713 digital signal processor (DSP) board. The system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz, and the performance is then illustrated by measuring the quiet zone in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone of quiet with satisfactory size and maximum 24 dB noise attenuation is successfully generated.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.14 no.11
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• pp.1191-1197
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• 2003

In the multi-carrier OFDM communication system, the inter-carrier-interference(ICI) produced by phase noise in the transceiver local oscillator makes a severe influence on the system performance. In this paper, a new ICI self-cancellation scheme in the data-conversion type is proposed to reduce effectively the ICI. Also, the common phase error(CPE), ICI and carrier to interference power ratio(CIR) are found by the linear approximation of the phase noise. Then, the proposed method is compared with the conventional OFDM to analyze the efficiency of system performance improvement. When the number of subcarriers is 64, there are respectively the SNR gain of 0.6 ㏈ in the phase noise variance of 0.3 with QPSK and 1.5 ㏈ in the phase noise variance of 0.1 with 16 QAM at BER=10$\^$-3/. As a result, the performance degradation by ICI can be effectively lowered in the proposed system with ICI self. cancellation scheme, compared with the conventional OFDM system.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.949-958
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• 2018

A voice recognition control system is a system for controlling a peripheral device by recognizing a voice. Recently, a voice recognition control system have been applied not only to smart devices but also to various environments ranging from IoT(: Internet of Things), robots, and vehicles. In such a voice recognition control system, the recognition rate is lowered due to the ambient noise in addition to the voice of the user. In this paper, we propose a dual channel acoustic beamforming hardware architecture for MEMS(: Microelectromechanical Systems) microphones to eliminate ambient noise in addition to user's voice. And the proposed hardware architecture is designed as ASIC(: Application-Specific Integrated Circuit) using TowerJazz $0.18{\mu}m$ CMOS(: Complementary Metal-Oxide Semiconductor) technology. The designed dual channel acoustic beamforming ASIC has a die size of $48mm^2$, and the directivity index of the user's voice were measured to be 4.233㏈.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.5
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• pp.3132-3138
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• 2014

In many communication systems, intersymbol interference, DC and impulsive noise are hard-to-solve problems. For the purpose of cancelling such interferences, the concept of lagged cross-correlation of probability has been used for blind equalization. However, this algorithm has a large burden of computation. In this paper, a recursive method of the algorithm based on the lagged probability correlation is proposed. The summation operation in the calculation of gradient of the cost is transformed into a recursive gradient calculation. The recursive method shows to reduce the high computational complexity of the algorithm from O(NM) to O(M) for M symbols and N block data having advantages in implementation while keeping the robustness against those interferences. From the results of the simulation, the proposed method yields the same learning performance with reduced computation complexity.