• Journal of Biomedical Engineering Research
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• v.22 no.3
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• pp.249-257
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• 2001

The voluntary contracted EMG signal of spinal cord injured patients is very small because the information from central nervous system is not sufficiently transmitted to $\alpha$ motor neuron or muscle fiber. Therefore the acquisited EMG signal from needle or surface electrodes can not be identified obvious voluntary contraction pattern by muscle movement. In this paper we propose the extraction technique of voluntary muscle contraction and relaxation pattern from EMG signal of spinal cord injured patient whose EMG signal is composed of the linear sum of mo색 unit action potentials with two noise sources, additive noise assumed to be white Gaussian noise and high frequency discharge assumed to be not motor unit action potential but impulsive noise. In order to eliminate impulsive noise and additive noise from voluntary contracted EMG signal, we use the FatBear filter which is a nonarithmetic piecewise constant filter, and multiscale nonlinear wavelet denoising processing, respectively. The proposed technique is applied to the EMG signal acquisited from transverse myelitis patients to extract voluntary muscle contraction pattern.

• Journal of Korea Multimedia Society
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• v.23 no.3
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• pp.402-411
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• 2020

Power line noise in electrocardiogram signals makes it difficult to diagnose cardiovascular disease. ECG signals without power line noise are needed to increase the accuracy of diagnosis. In this paper, it is proposed DNN(Deep Neural Network) model to remove the power line noise in ECG. The proposed model is learned with noisy ECG, and clean ECG. Performance of the proposed model were performed in various environments(varying amplitude, frequency change, real-time amplitude change). The evaluation used signal-to-noise ratio and root mean square error (RMSE). The difference in evaluation metrics between the noisy ECG signals and the de-noising ECG signals can demonstrate effectiveness as the de-noising model. The proposed DNN model learning result was a decrease in RMSE 0.0224dB and a increase in signal-to-noise ratio 1.048dB. The results performed in various environments showed a decrease in RMSE 1.7672dB and a increase in signal-to-noise ratio 15.1879dB in amplitude changes, a decrease in RMSE 0.0823dB and a increase in signal-to-noise ratio 4.9287dB in frequency changes. Finally, in real-time amplitude changes, RMSE was decreased 0.3886dB and signal-to-noise ratio was increased 11.4536dB. Thus, it was shown that the proposed DNN model can de-noise power line noise in ECG.

• The Journal of the Acoustical Society of Korea
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• v.40 no.1
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• pp.46-54
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• 2021

In this paper, active noise control was performed to reduce radiated noise in the low frequency band of dishwashers. First, through an analysis of the noise environment of the dishwasher, it was confirmed that the pump noise contributed the most to the radiated noise in the low frequency band, From the result of the noise environment analysis, the reference signal was selected to be the vibration signal of the pump body. The reference signal was obtained by using the accelerometer on the pump body, which can prevent acoustic feedback. The error signal sensor was selected as a microphone located at 1 m in front of the dishwasher and 0.5 m in height. And to design the controller, the error signal and the reference signal were measured at the operational rpms of the dishwasher at 2,500 rpm, 2,600 rpm and 2,800 rpm, and the secondary path transfer function was measured. The designed controller was mounted on Digital Signal Processor (DSP) equipment, and the control performance was verified experimentally. As a result of the measurement at the 3 operational rpms, the 7th multiple component of pump operating frequency decreased by 1.93 dB, 4.43 dB, 5.15 dB per rpm, and the 12th multiple component decreased by 6.67 dB, 2.34 dB, 4.28 dB per rpm. And overall Sound Pressure Level (SPL) decreased by 0.84 dB, 2.58 dB, 1.48 dB by rpm.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.1
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• pp.129-139
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• 1989

During a signal is being transmitted, an interference signal can be introduced through an unknown channel. In these cases, an adaptive system, so called adaptive noise canceller, can restore the original signal from the corrupted signal by first identifying the unknown interference channel on the minimum mean square error criteron, and then by cancelling the interference signal using the identified interference channel. Whereas this method is quite effective when the a priori knowledges about the characteristics of the interference signal and of the intrference channel are unknown or time-varyng, but has a drawback that the presence of the original signal has a severe effect on the optimum value of the interference channel to be identified on the miniumum mean square eror criterion In this paper an adaptive noise canceller and its algorithms are introduced that can restore the original signal more accurately especially when the correlatedness between the original signal and the interference signal is small.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3E
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• pp.87-96
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• 2007

In this paper we propose a new noise reduction algorithm for stationary and nonstationary noisy environments. Our algorithm classifies the speech and noise signal contributions in time-frequency bins, and is not based on a spectral algorithm or a minimum statistics approach. It relies on calculating the ratio of the standard deviation of the noisy power spectrum in time-frequency bins to its normalized time-frequency average. We show that good quality can be achieved for enhancement speech signal by choosing appropriate values for ${\delta}_t\;and\;{\delta}_f$. The proposed method greatly reduces the noise while providing enhanced speech with lower residual noise and somewhat higher mean opinion score (MOS), background intrusiveness (BAK) and signal distortion (SIG) scores than conventional methods.

• ETRI Journal
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• v.38 no.2
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• pp.366-375
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• 2016

This paper describes an algorithm to suppress composite noise in a two-microphone speech enhancement system for robust hands-free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal-dominant time-frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech-dominant TFBs are identified among the previously detected nonstationary signal-dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin-wise output signal-to-noise ratio is obtained with these power estimates and a Wiener post-filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post-filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.11a
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• pp.346.1-346
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• 2002

There have been many studies to visualize the vibration and noise of rotary compressor. Most of these studies assumed that the signal is stationary and the time-averaged signal is used for visualization. However, the noise and vibration signals generated during one cycle of crank shaft vary continuously. In this paper, the noise and vibration of rotary compressor which vary continuously are visualized by short time fourier transform method. (omitted)

• 한국가시화정보학회:학술대회논문집
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• 2001.12a
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• pp.123-137
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• 2001

PIV (particle image velocimetry) systems use a camera to take snapshots of particles carried by a fluid at some precise instants. Signal processing methods are then used to compute the flow velocity field. In this paper, the design of the camera objective (optics) is addressed. The optimization is done in order to maximize the signal-to-noise ratio of in-focus particles. Four different kinds of noise are considered: photon shot noise, thermal and read noise, background glow shot noise, and noise made by the other particles. A semi-empirical model for the lens aberrations of a two-doublet objective is first addressed, since further, it is shown that lens aberrations (low f-value $f_{\#}$) should be used instead of the Fraunhofer diffraction (high f-value) for the fitting of the particle image size with the pixel size. Other important conclusions of the paper include the expression of optimum values for the magnification M, for the exposure period $\tau$ and for the pixel size $\xi$.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.14 no.9 s.90
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• pp.877-883
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• 2004

This paper addresses the way in which we can find where impulsive noise sources are. Specifically, we have an interest in the case that the signal is embedded in noise. We propose a signal processing method that can identify impulsive sources' location. The method is robust with respect to spatially distributed noise. This has been achieved by the modified beamforming method with regard to cepstrum domain is used. It is noteworthy that the cepstrum has the ability to detect periodic pulse signal in noise. Numerical simulation and experiments are performed to verify the method. Results show that the proposed technique is quite powerful for localizing the faults in noisy environments. The method also required less microphones than conventional beamforming method.

• The Journal of the Acoustical Society of Korea
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• v.17 no.1E
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• pp.30-35
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• 1998

Given a noisy sampled at equispaced points with white noise, we consider problems where the signal to be recovered is known to be positive; for example, images, chemical spectra or other measurements of intensities. Shrinking noisy wavelet coefficients via thresholding offers very attractive alternatives to existing methods of recovering signals from noisy data. In this paper, we propose a method of recovering the original signal from a corrupted noisy signal, guaranteeing the recovered signal positive. We first obtain wavelet coefficients by thresholding, and use a nonlinear optimization to find the denoised signal which must be positive. Numerical examples are used to illustrate the performance of the proposed algorithm.