• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7C
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• pp.562-568
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• 2008

This paper proposes the scheme which obtain the coefficients of TDL filter and two normalization algorithms among methods which get solution of equivalent Wiener-Hopf Equation in Gram-Schmidt algorithm. Compared to the conventional NLMS algorithm, normalizes with sum of power of inputs, the presented algorithms normalize using sums of eigenvalues. Using computer simulation, we perform an system identification in an unstable environment where two poles are located in near position outside unit circle. Consequently, the proposed algorithms get the coefficients of TDL filter in Gram-Schmidt algorithm recursively and show better convergence performance than conventional NLMS algorithm.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.77-80
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• 2005

In this paper, we introduce the efficient carrier phase estimating algorithm collaborate with the channel decoder of turbo coded QPSK modulation for mobile DVB-RCS systems. At low SNR, the phase estimation using soft information of turbo decoder is able to improve power efficiency because of achieving the good synchronization. We investigate performance of external single estimator and internal multiple estimator in the PSP (Per Survivor Processing) manner over AWGN channel. For phase estimation, the LMS (Least Mean Square) scheme is considered. Three different APP-based methods are also proposed.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.36 no.5
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• pp.367-376
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• 1987

In this paper we have disigned and implemented a real time ALE (adaptive line enhancer) using a high speed digital processor,NEC 7720. For the ALE system, we have employed an adaptive LMS(least mean square) algorithm proposed by Widrow and Hoff and a 32-order FIR(finite impulse response) filter. Extensive computer simulations have been performed to investigate the peformance of the ALE and to determine necessary parameters for hardware design. The developed software for an NEC 7720 was tested in real time operation using an NEC7720 hardware emulator. The ALE has been tested by sinusoidal waves and real CW (continuous wave) signals. It was found that the experimental results were well agreed with the computer simulation results. Thus it can be concluded that the ALE is useful for detection and enhancement of a sinusoidal signal which is corrupted by an additive Gaussian noise.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2200-2203
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• 2003

In an active noise control(ANC) system using the Filtered-X least mean square(LMS) algorithm, the online secondary path modeling method by exploiting a random noise generator is applied. This method is suitable for secondary path modeling. However, it is increased the residual error of the ANC system. In this paper, we presents an ANC system improved online secondary path modeling method which is modified Kuo and Zhang model that is the secondary path estimation by the additive noise. In addition, our proposed model is used that additive noise is transformed into the signal multiplied reference signal by gain control parameter and delayed.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.30 no.2
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• pp.125-131
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• 1994

This paper describes a method to measure the distortion level of loudspeaker using a LMS(Least Mean Square) adaptive filter. Conventional technique to measure the distortion level uses a band-pass filter with a sharp cut-off frequency characteristics. However. such the band-pass filter has a bed time response characteristics. On the other hand, the proposed method offers us an easy way to measure the specified harmonic distortion level with a small hardware. Moreover, our method is not affected by noise which has no correlation with the test signal, and the measurement can be carried out in a noisy environment. The effectiveness of the proposed method is confirmed by experiment using a loudspeaker in a noisy room.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2002.11a
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• pp.9-14
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• 2002

Direct-conversion receiver(DCR) architecture has superior advantages in size, cost, and power over superheterodyne receiver architectures. However, the use of direct-conversion receiver architecture has been limited due to the direct-current offset noise. The ac coupling, which is used to overcome the direct-current offset noise, causes an inter-symbol interference(ISI), whose effects can be effectively mitigated using an equalizer. In this paper, the performance of a direct-conversion receiver with ac coupling in the presence of direct-current offset is analyzed via computer simulation. The simulation result shows that by using decision feedback equalizer with LMS(Least Mean Square) algorithm, signal-to-noise ratio loss of the direct-conversion receiver compared to the idea receiver can be reduced to less than 1㏈ for corner frequencies as large as 10% of the symbol rate.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.67-72
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• 2006

적응신호처리 분야에서 LMS(Least Mean Square) 알고리즘은 수식이 간단하고, 적은 계산 량으로 인해 널리 사용되고 있지만, 시간영역의 적응알고리즘은 입력신호의 고유치 분포 폭이 넓게 분포할 때는 수렴속도가 느려지는 단점이 있다. 본 논문에서는 적응 신호처리의 수렴속도를 향상 시키고 복잡한 계산 량을 줄이는 새로운 fast running FIR 필터 구조를 제안한다. 그리고 제안한 알고리즘을 가변스텝 사이즈 웨이블렛 기반 적응 알고리즘에 적용한다. 실제로 합성 음성을 사용하여 적응 잡음 제거기에 적용하여 컴퓨터 시뮬레이션을 통해 제안한 알고리즘과 기존 알고리즘과의 성능을 비교한다.

• Speech Sciences
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• v.8 no.1
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• pp.159-165
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• 2001

In this paper, we make an experiment on speech recognition systems with barging-in and non-barging-in utterances. Barging-in capability, with which we can say voice commands while voice announcement is coming out, is one of the important elements for practical speech recognition systems. Barging-in capability can be realized by echo cancellation techniques based on the LMS (least-mean-square) algorithm. We use three kinds of telephone interface boards with barging-in capability, which are respectively made by Dialogic Company, Natural MicroSystems Company and Korea Telecom. Speech database was made using these three kinds of boards. We make a comparative recognition experiment with this speech database.

• The Journal of the Acoustical Society of Korea
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• v.24 no.3E
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• pp.99-108
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• 2005

To investigate the equalizer performance in underwater acoustic communication m the presence of intersymbol interference (ISI) due to multipath, computer simulations are carried out in discrete multipath shallow water channels for three different horizontal ranges. For the purpose of computation simplicity, least mean square (LMS) algorithm is adopted both in linear equalizer and nonlinear equalizer, decision feedback equalizer (DFE) to cancel out ISI effects. Binary phase shift keying (BPSK) signals have been transmitted with high data rate of 2000bps through the use of equalization technique. The results demonstrate that equalization is an efficient way to achieve high transmission data rate in the shallow water channel.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.357-358
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• 2006

We propose the robust PBFLMS in noises, which is the enhanced acoustic echo canceller using ACPBF-LMS(Alternative Constrained Partitioned Block Frequency domain Least Mean Square) algorithm. The defect of the block structure filtering is the deterioration of convergence efficiency from noise and interference. To improve the performance of convergence efficiency, noise effect should be reduced. The new method of reducing noise effect is proposed, which apply the estimated background noise to adaptive filter step size. By experiments, the proposed acoustic echo canceller has TCL of 50dB, and always provides faster convergence speed and lower complexity than the full-tap NLMS. We also carried out an implementation of PBFLMS using ARM9EJ-S.