• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.7
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• pp.1028-1035
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• 1993

The signed regressor (SR) algorithm employs one bit quantization on the input regressor (or tap input) in such a way that the quantized input sequences become +1 or -1. The algorithm is computationally more efficient by nature than the popular least mean square (LMS) algorithm. The behavior of the SR algorithm unfortunately is heavily dependent on the characteristics of the input signal, and there are some Inputs for which the SR algorithm becomes unstable. It is known, however, that such a stability problem does not take place with the SR algorithm when the input signal is Gaussian, such as in the case of speech processing. In this paper, we explore a statistical analysis of the SR algorithm. Under the assumption that signals involved are zero-mean and Gaussian, and further employing the commonly used independence assumption, we derive a set of nonlinear evolution equations that characterizes the mean and mean-squared behavior of the SR algorithm. Experimental results that show very good agreement with our theoretical derivations are also presented.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.2
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• pp.69-77
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• 2002

Speaker verification can be classified in two categories, text-dependent speaker verification and text-independent speaker verification. In this paper, we discuss text-dependent speaker verification. Text-dependent speaker verification system determines whether the sound characteristics of the speaker are equal to those of the specific person or not. In this paper we obtain the speaker data using a sound card in various noisy conditions, apply a new Hidden LMS (Least Mean Square) adaptive algorithm to it, and extract LPC (Linear Predictive Coding)-cepstrum coefficients as feature vectors. Finally, we use a competitive learning neural network for speaker verification. The proposed hidden LMS adaptive filter using a neural network reduces noise and enhances features in various noisy conditions. We construct a separate neural network for each speaker, which makes it unnecessary to train the whole network for a new added speaker and makes the system expansion easy. We experimentally prove that the proposed method improves the speaker verification performance.

• Journal of Satellite, Information and Communications
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• v.7 no.1
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• pp.82-85
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• 2012

In this paper, we present the forward link ACM method to improve the link availability and system throughput. Also, we compare the prediction algorithm between slope based prediction and LMS algorithm. The simulation results show that the 99% of predicted values in LMS algorithm is within 3dB and that of predicted values in the slope based prediction method is within 4.5dB.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.45 no.10
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• pp.863-871
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• 2017

LMS (least mean square) algorithm widely used in the AVCS (active vibration control system) of helicopters calculates control input using the forward path transfer function and error signal. If the error signal is sinusoidal, it can be represented as the combination of cosine and sine functions with frequency and phase synchronized with the reference signal. The control input also has the same frequency, therefore control algorithm can be simply implemented if the cosine and the sine amplitudes of the control input are calculated and the frequency and phase of the reference signal are used. Calculation of the control input is implemented as simple matrix operation and the change of the control command is slower than the frequency of the error signal, consequently control algorithm can be operated at lower frequency. The signal processing algorithm extracting cosine and sine components of the error signals are modeled using Simulink and PIL (processor-in-the-loop) mode simulation was executed for real-time performance evaluation.

• Communications for Statistical Applications and Methods
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• v.25 no.6
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• pp.673-685
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• 2018

Procrustes analysis is a useful technique useful to measure, compare shape differences and estimate a mean shape for objects; however it is based on a least squares criterion and is affected by some outliers. Therefore, we propose two generalized Procrustes analysis methods based on M-estimation and least median of squares estimation that are resistant to object outliers. In addition, two algorithms are given for practical implementation. A simulation study and some examples are used to examine and compared the performances of the algorithms with the least square method. Moreover since these resistant GPA methods are available for higher dimensions, we need some methods to visualize the objects and mean shape effectively. Also since we have concentrated on resistant fitting methods without considering shape distributions, we wish to shape analysis not be sensitive to particular model.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.245-250
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• 2009

In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.

• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.377-384
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• 2015

An active noise control (ANC) algorithm for noise canceling headphone is proposed. In this study, the feedback ANC operated with the filtered-x least mean square algorithm (FxLMS) algorithm is used to attenuate the undesired noise. Also an adaptive residual music canceller (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Simulation results show that a high quality of music sound can be consistently achieved in a time-varying secondary path situation.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.331-337
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• 2018

In this paper, we apply the NR (Noise Rating) to the evaluation of the residual noise of the window type ANC (Active Noise Control) and study the active noise controller for minimizing the NR value in rooms. We proposed a shape of a noise shaping filter of the Filtered-E LMS (Least Mean Square) algorithm that reduces the NR value within the effective operating frequency band of the ANC. The usefulness of the proposed scheme is verified by simulation and showed that the NR value of the residual noise is lower than the filters used in the conventional psychoacoustic ANC.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.3
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• pp.473-479
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• 2005

A key issue toward mobile multimedia communications is to create technologies for broadband signal transmission that ran support high quality services. Such a broadband mobile communications system should be able to overcome severe distortion caused by time-varying multi-path fading channel, while providing high spectral efficiency and low power consumption. For these reasons, an adaptive suboptimum decision feedback equalize. (DFE) for the single-carrier short-burst transmissions system is considered as one of the feasible solutions. For the performance improvement of the system with the short-burst format including the short training sequence, in this paper, the multiple-training least mean square (MTLMS) based DFE scheme with soft decision feedback is proposed and its performance is investigated in mobile wireless channels throughout computer simulation.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.8 no.6
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• pp.589-599
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• 1997

In this paper a decision feedback multiuser receiver which rejects multiuser interference is designed in synchronous CDMA systems. Multiuser interference rejection is achieved by decision feedback from the output of a matched filter bank in case of AWGN channels and PAKE receiver bank in case of fading channels. The coefficients of the decision feedback structure are adapted by the Least Mean Square(LMS) algorithm. Simulation results show that the decision feedback multiuser receiver has better performance than the conventional receiver.