• Journal of Digital Convergence
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• v.10 no.5
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• pp.233-238
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• 2012

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. To solve this problem, this paper present a method of LMS-MPC uses individual pitch and LMS(Least Mean Square). I evaluate the MPC and LMS-MPC using LMS. As a result, SNRseg of LMS-MPC was improved 1.5dB for female voice and 1.3dB for male voice respectively. Compared to the MPC, SNRseg of LMS-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• 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.34 no.5
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• pp.403-409
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• 2015

In this paper, we propose Packet Loss Concealment (PLC) method using adaptive signal scale estimation for performance improvement of G.711 PLC. The conventional method controls a gain using 20 % attenuation factor when continuous loss occurs. However, this method lead to deterioration because that don't consider the change of signal. So, we propose gain control by adaptive signal scale estimation through before and after frame information using Least Mean Square (LMS) predictor. Performance evaluation of proposed algorithm is presented through Perceptual Evaluation of Speech Quality (PESQ) evaulation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.3
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• pp.1-8
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• 2001

In the DS/CDMA mobile communication systems, the parallel interference canceller is used in order to reduce the multiple access interference and the multipath fading. It is needed the accurate interference estimate in the multistage parallel cancellation. In this paper, the adaptive cancellation method and the new tentative decision device arc proposed and the performance is analyzed. The adaptive cancellation method uses the normalized least mean square(NLMS) algorithm to calculate the weight adaptively, and new tentative decision device uses the hyperbolic tangent decision with null zone. Computer simulation shows that the proposed scheme has the improved performance and the number of user is increased 48% compared with the conventional receiver.

• Journal of the Korean Society for Nondestructive Testing
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• v.23 no.5
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• pp.455-463
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• 2003

Adaptive noise cancellation techniques are ideally suitable for reducing spatially varying noise due to the grain structure of material in ultrasonic nondestructive evaluation. Grain noises have an un-correlation property, while flaw echoes are correlated. Thus, adaptive filtering algorithms use the correlation properties of signals to enhance the signal-to-noise ratio (SNR) of the output signal. In this paper, a multi-stage adaptive noise cancellation (MANC) method using adaptive least mean square error (LMSE) filter for enhancing flaw detection in ultrasonic signals is proposed.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.64 no.3
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• pp.169-174
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• 2015

Noise problem that occurs on the road is raising a lot of problems in the economic, social and environmental aspects. The active noise control (ANC) systems based on the filtered-X least mean square(FxLMS) algorithm have a problem with compensating the acoustic feedback of secondary route. However, newly proposed correlation-LMS(CLMS) and expanded CLMS algorithms have slightly much calculation and are minutely behind performance, these have a advantage not in measuring transfer function onerously so that we can easily adapt these in real time. The CLMS and expanded CLMS algorithm was developed to improve the real-time implementation performance under the variable input noise such as road noise environment. In this paper, we compared and analyzed their performance. From the results of the Matlab simulation for an ANC system, it is shown that expanded CLMS algorithms are more convergence speed and keep the desirable performance even in the input of road noise situation.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.27 no.1
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• pp.87-93
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• 2017

This paper investigates active noise control of tower-type air conditioners using the filtered-x least mean square (FXLMS) algorithm to reduce fan blower noise transmission. Firstly, the main components required for the active control system including the error sensor, the control speaker and the reference sensors are selected. Since the noise could significantly reduce if the reference signal includes every frequency response information, a various reference signals from accelerometers and a microphone are used. Secondly, the controller based on the FXLMS algorithm with a single-channel reference signal is implemented. Then, the control performance is examined experimentally for the different reference signals. It is found that the accelerometer signal well possesses the motor vibration related noise and a microphone signal could includes global noise. When using the reference signal with a microphone located near the motor and the fan blower, the active control system reduces the noise globally, except for several peaks.

• Journal of Biomedical Engineering Research
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• v.25 no.4
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• pp.315-321
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• 2004

In this study, we proposed an adaptive feedback cancellation algorithm for multi-band digital healing aids. The adaptive feedback canceller (AFC) is composed of an adaptive notch filter (ANF) for feedback detection and an NLMS (normalized least mean square) adaptive filter for feedback cancellation. The proposed feedback cancellation algorithm is combined with a multi-band hearing aid algorithm which employs the MDCT (modified discrete cosine transform) filter bank for the frequency-dependent compensation of hearing losses. The proposed algorithm together with the MDCT-based multi-channel hearing aid algorithm has been evaluated via computer simulations and it has also been implemented on a commercialized DSP board for real-time verifications.

• Journal of the Korean Society for Aviation and Aeronautics
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• v.18 no.4
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• pp.30-37
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• 2010

This paper deals with attitude determination and parameter estimation problems for a lunar module. For this we first derive equations of motion for the lunar module by considering allocation locations (configurations) of reaction thruster and a reaction wheel assembly. The lunar module is assumed as a rigid body. In order to include the effect of fuel sloshing on the dynamics of the lunar module, we model it as a spherical pendulum for a simple analysis. For estimating angular rates and moment of inertia of the module, we employ an extended Kalman filter and the least mean square algorithms, respectively. Finally we construct a dynamical model for the lunar module by combining all these elements.

• Journal of Korea Multimedia Society
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• v.6 no.3
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• pp.462-468
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• 2003

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This paper present a new method of TSIUVC approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547KHz below and 2.813KHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TSIUVC. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.