• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06d
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• pp.80-83
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• 1998

In this paper, we propose a new filtered-x least mean fouth (LMF) algorithm where the error raised to the power of four is minimized and analyze its convergence behavior or a multiple sinusoidal acoustic noise and Gaussian measurement noise. Application of the filtered-x LMF adaptive filter to active noise cancellation (ANC) requires estimating of the transfer characteristic of the acoustic path between the ouput and error signal of the adaptive canceller. The results of the convergence analysis of the filtered-x LMF algorithm indicates that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct component . Phase estimation error and estimated again. In particular , the convergence is shown to be strongly affected by the accuracy of the phase response estimate. Also, we newly show that convergence behavior can differ depending on the relative sizes of the Gaussian measurement noise and convergence constant.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.423-426
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• 2003

Adaptive algorithm used in Acoustic Echo Canceller (AEC) needs fast convergence algorithm when reference signal is colored speech signal. Set-Membership Affine Projection (SMAP) algorithm is derived from the constraint, which is the minimum value adaptive filter coefficient error. In this paper, we test the characteristic about noise of the SMAP algorithm and proposed modified version of SMAP algorithm fur using at AEC. As the projection order increase, the convergence characteristic of the SMAP algorithm is improved where no noise space. But if the noise uncorrelated with input signal exists, the AEC shows bad performance. In this paper, we propose normalized version of adaptive constants using estimated error signal for robust to noise and show the good performance through AEC simulation.

• 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.

• Speech Sciences
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• v.9 no.4
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• pp.275-287
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• 2002

In man-machine communication by speech under the noisy environment, the quality of speech may be degraded severely for the machine to recognize correctly. Especially when the corrupting noise occupies the same band as the speech, the conventional fixed filters cannot filter out the noise effectively. In recent, to resolve such a problem adaptive noise canceller (ANC) is frequently used, which is based upon adaptive filters. The Adaptive recursive filters perform better than adaptive nonrecursive filters due to the added poles, but the stability may be severely threatened. In this paper an ANC system employing the adaptive recursive filter is proposed to enhance the speech corrupted by noise. And the stability of the adaptive recursive filter is guaranteed by employing the adaptive compensator.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.3
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• pp.138-141
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• 2006

Blood flow is one of vital signals related to human physiological information. Photoplethysmograph (PPG) has been used to measure indirectly heart rate, blood oxygen saturation ($SpO_2$), and so on. Because PPG signal is weak and sensitive to motion artifacts, it is very important to continuously obtain stable PPG signal during free movement. In this study, we applied the scaled Fourier linear combiner (SFLC) using both the adaptive filter and FLC to remove effectively the motion artifacts as well as background noise in the real time without additional signal correlated with motion from a accelerometer. The proposed method would be useful to reduce the movement and background noise which are not synchronized with heart rate.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2E
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• pp.45-52
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• 1999

This paper presents the integrated speech enhancement system for hands-free mobile communication. The proposed integrated system incorporates both acoustic echo cancellation and engine noise reduction device to provide signal enhancement of desired speech signal from the echoed plus noisy environments. To implement the system, a delayless subband adaptive structure is used for acoustic echo cancellation operation. The NLMS based adaptive noise canceller then applied to the residual echo removed noisy signal to achieve the selective engine noise attenuation in dominant frequency component. Two sets of computer simulations are conducted to demonstrate the effectiveness of the system; one for the fixed acoustical environment condition, the other for the robustness of the system in which, more realistic situation, the acoustic transmission environment change. Simulation results confirm the system performance of 20-25dB ERLE in acoustic echo cancellation and 9-19 dB engine noise attenuation in dominant frequency component for both cases.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.10-17
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• 2010

In a passive sonar, the improvement of detection performance by using noise cancellation is usually a important problem. In this paper, we have analyzed the own-ship noise cancellation in the two operation modes which are used in the HMS system. In the operator mode, an adaptive line enhancer(ALE) is applied to improve the tonal detection by using broadband noise cancellation and the normalized least mean square(NLMS) algorithm is applied to the design of an adaptive filter. The reference input that is correlated with a primary input can be used to remove the noise incident on the observation directionin the automatic mode. Computer simulations with real sea that data show that the proposed adaptive noise canceller has good performance in passive detection under HMS operation.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.143-146
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• 1996

An adaptive digital filtering of the electroencophalogram(EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, an artifact which not be acceptable in some cases of automatic EEG processing. The types of electrical artifact seen on EEG records is described. Those are the EOG and the PVC roller pump noise. And we study the method for cancelling these artifacts. The method does not need the reference channel, and are obtained by cascading the linear predictor and the noise canceller. The simulation results illustrate the performances of the presented method in terms of the capability of interferences suppression.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.802-805
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• 1994

There are strong ocean wave interference with big amplitude very low frenqencies are similar to the ship's hydrodynamic signals. To detect ship's hydrodynamic field will be faced various natural hydrodynamic interferences which are radom and the prior knowlege of which are not know. This paper proposes to use the adaptive noise cancelling principle and used the model of adaptive wave canceller to eliminate the ocean wave interfrence and detect the ship's hydrodynamic signals. Computer simulation results shown that signal to noise ratio can be raised from several to ten times. It shows the fact that this mathod can detect the ship's hydrodynamic signals from the strong ocean wave interferences while it is difficult for the old methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.5C
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• pp.478-486
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• 2007

Double-talk detection errors can result in either large residual echo or distorting the near-end talker's input speech. Thus accurate double-talk detection is an important problem in the acoustic echo canceller to improve the speech quality. In the double-talk detection algorithm using a cross-correlation coefficient, double-talk detection errors can occur in the initial convergence period of an adaptive filter or in noisy environment since the cross-correlation coefficient becomes large in such situations. In this paper, we propose a new double-talk detection algorithm based on the cross-correlation method using a normalized error signal power to reduce the double-talk detection errors. The experimental results have shown the performance improvement of an acoustic echo canceller as well as the noise-robustness of the proposed double-talk detector.