• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1468-1476
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• 1997

In mobile communication systems, the Reduced State Sequence Estimation(RSSE) receiver must be able to track changes in the channel. This is carried out by the adaptive channel estimator. However, when the tentative decisions are used in the channel estimator, incorrect decisions can cause error propagation. This paper presents a new channel estimator using the path history in the Viterbi decoder for preventing error propagation. The selection of the path history in the Viterbi decoder for preventing error propagation. The selection of the path history for the channel estimator depends on the path metric as in the decoding of the Viterbi decoder in RSSE. And a discussion on the channel estimator with different adaptation algorithms such as Least Mean Square(LMS) algorithm and Recursive Least Square(RLS) algorithm is provided. Results from computer simulations show that the RSSE receivers using the proposed channel estimator have better performance than the other conventional RSSE receiver, and that the channel estimator with RLS algorithm is adequate for multipath fading channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.1984-1989
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• 1998

Convergence properties of the stochastic gradient adaptive algorithm based on the least mean absolute third (LMAT) error criterion is presented.In particular, the performnce of the algorithmis examined and compared with least mena square (LMS) algorithm for several different probability densities of the measurement noisein a system identification mode. It is observedthat the LMAT algorithm outperforms the LMS algorithm for most of the noise probability densities, except for the case of the exponentially distributed noise.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.565-567
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• 2004

In active noise control (ANC) systems, the convergence behavior of the conventional Filtered-X Least Mean Square (FXLMS) algorithm may be affected by nonlinear distortions in the secondary path (e.g., in the power amplifiers, loudspeakers, transducers, etc.), which may lead to degradation of the error-reduction performance of the ANC systems. In this paper, a stable FXLMS algorithm with fast convergence is proposed to compensate for undesirable nonlinear distortions in the secondary-path of ANC systems by employing the Volterra filtering approach. In particular, the proposed approach is based on the utilization of the conventional P-th order inverse approach to nonlinearity compensation in the secondary path of ANC systems. Finally, the simulation results showed that the proposed approach yields a better convergence behavior In the nonlinear ANC systems than the conventional FXLMS.

• Proceedings of the Korean Society of Machine Tool Engineers Conference
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• 2002.04a
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• pp.333-338
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• 2002

자기베어링의 회전정밀도에 영향을 미치는 인자로 PWM 전력증폭기, 위치 센서 등과 같은 자기베어링 구성 장치의 동특성 및 정밀도, 시스템의 정확한 모델링, 제어기법, 런아웃 등이 있다. 본 연구에서는 능동 자기베어링을 제어하기 위해 자기베어링의 PWM 전력증폭기와 회전축을 모델링하고 이를 바탕으로 능동 자기베어링 제어를 위한 PID 제어기를 구성하였으며, 변위 센서의 부착위치 및 회전축의 진원도의 영향으로 발생하는 주기적인 런아웃 요소를 첨가하여 런아웃의 영향을 확인하였으며, 런아웃 (Runout)에 의해 발생하는 에러(Error)를 효과적으로 제어하여 자기베어링의 제어 정밀도를 향상시키기 위한 방법으로 기본적인 PID 제어기에 최소평균자승(Least Mean Square, LMS) 알고리즘을 적용한 적응 피드포워드 제어기를 구성하여 자기베어링의 능동 제어에서 발생하는 주기적인 런아웃을 효과적으로 제어할 수 있음을 MATLAB을 통한 시뮬레이션을 통해 확인하였다.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.3
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• pp.158-163
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• 2009

LMS(Least Mean Square) algorithm requires simple equation and is used widely because of the low complexity. If the convergence speed increase, LMS algorithm has a divergence in case of sharp environment changes. And if a stability increase, the convergence speed becomes slow. This algorithm based on a trade off between fast convergence and system stability. To improve this problem, VSSLMS (Variable Step Size LMS) algorithm was developed. The VSSLMS algorithm improved the convergence speed and performance as adjusting step size using error signal. In this paper, I-VSSLMS algorithm is proposed tor improve the performance of adaptive noise canceller in real-time environments. The proposed algorithm is applied to adaptive noise canceller using TMS320C6713 DSP board and we did simulation by real time. Then we compared performance of each algorithm and demonstrated that proposed algorithm has superior performance.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.8
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• pp.28-38
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• 2013

In this paper, we propose a improved turbo equalizer which generates a feedback signal through a simple calculation to improve performance in single carrier system with the LMS(least mean square) algorithm based equalizer and LDPC(low density parity check) codes. LDPC codes can approach the Shannon limit performance closely. However, computational complexity of LDPC codes is greatly increased by increasing the repetition of the LDPC codes and using a long parity check matrix in harsh environments. Turbo equalization based on LDPC code is used for improvement of system performance. In this system, there is a disadvantage of very large amount of computation due to the increase of the repetition number. To less down the amount of this complicated calculation, The proposed improved turbo equalizer adjusts the adoptive equalizer after the soft decision and the LDPC code. Through the simulation results, it's confirmed that performance of improved turbo equalizer is close to the SISO-MMSE(soft input soft output minimum mean square error) turbo equalizer based on LDPC code with the smaller amount of calculation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.6C
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• pp.343-348
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• 2011

This paper introduces a design for real time wireless headset using ANC (active noise control) system based on NFxLMS adaptive filter algorithm. The training time of the proposed system is significantly reduced by using the RMS delay spread of a channel as an error correction parameter, and convergence rate of the FxLMS filter has been improved with updating the coefficients of the NFxLMS filter, which we have got during the training process. Our system has shorter training time and better convergence rate at the same noise reduction level than the conventional system under real noisy environment.

• ETRI Journal
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• v.34 no.1
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• pp.130-133
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• 2012

Recently, several noise-robust adaptive multichannel LMS algorithms have been proposed based on the spectral flatness of the estimated channel coefficients in the presence of additive noise. In this work, we propose a general form for the algorithms that integrates the existing algorithms into a common framework. Computer simulation results are presented and demonstrate that a new proposed algorithm gives better performance compared to existing algorithms in noisy environments.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.899-902
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• 1999

In this paper, we designed a equalizer that moved the multipath of channel in 8-VSB transmission receiver. After doing the initial equalization with "LMS(Least Mean Square)"aigorithm. this equalizer used "Stop-and-Go" algorithm. Because of estimating SER(Symbol to Error Ratio) every a training sequence, this can positively cope with transformation of channel and because of using fast clock than symbol-clock(10.76 MHz), we are able to reduce a multiplier.

• Proceedings of the IEEK Conference
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• 2000.06e
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• pp.81-84
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• 2000

This paper proposes a new sigma delta modulator structure based on a LMS(Least Mean Square) algorithm that minimizes the quantization noise. The proposed architecture provides 40dB SNR improvement and 35dB wider dynamic range over conventional sigma delta modulation. The proposed architecture provides superior performance especially when the input signal is small.