• Journal of Information Processing Systems
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• v.8 no.4
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• pp.669-684
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• 2012

Discrete wavelet transforms are extensively preferred in biomedical signal processing for denoising, feature extraction, and compression. This paper presents a new denoising method based on the modeling of discrete wavelet coefficients of ECG in selected sub-bands with Kernel density estimation. The modeling provides a statistical distribution of information and noise. A Gaussian kernel with bounded support is used for modeling sub-band coefficients and thresholds and is estimated by placing a sliding window on a normalized cumulative density function. We evaluated this approach on offline noisy ECG records from the Cardiovascular Research Centre of the University of Glasgow and on records from the MIT-BIH Arrythmia database. Results show that our proposed technique has a more reliable physical basis and provides improvement in the Signal-to-Noise Ratio (SNR) and Percentage RMS Difference (PRD). The morphological information of ECG signals is found to be unaffected after employing denoising. This is quantified by calculating the mean square error between the feature vectors of original and denoised signal. MSE values are less than 0.05 for most of the cases.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.5
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• pp.171-180
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• 2003

CDMA(Code Division Multiple Access) mobile communication systems require accurate channel estimation in the receiver to compensate the fading distortions. Instantaneous channel estimates are obtained by dividing the received symbol by the transmitted symbol and then refined by filtering to reduce the estimation variance. In the channel estimation filter, the determination of the filter size is a very important task which greatly affects the estimation quality. While conventional methods usually use only velocity estimators to determine the channel estimation filter size, this paper proposes a filter size determination method for decision-directed channel estimators considering the symbol error rate and the signal-to-noise ratio in addition to the velocity of the mobile station. This paper shows that the symbol error rate and the signal-to-noise ratio are important factors for the determination of the channel estimation filter size.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.2
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• pp.395-400
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• 2008

This paper relates to the performance analysis of DoA estimation algorithm in 2-dimensional circular array antenna for the receiving of GPS signal which is used for the performance improvement by elimination of jammer signal. By performing the spatial filtering after the DoA estimation in array antenna, the quality of receiving signal can improve by the nulling of jammer signal from the undesired direction and the forming of beam from the desired direction. In this paper, the MUSIC and MinNorm algorithm used for DoA estimation were applied after fixing the angle and power of jammer signal in 4 element and 7 element circular array antenna. In order to performance analysis, the estimation result and estimation error were computed by computer simulation. As a result, the MUSIC and MinNorm were fairly good in azimuth and elevation angle estimation of DoA in case of good signal to noise ratio and the MUSIC has better performance compared to MinNorm in case of poor signal to noise ratio.

• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.148-153
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• 2010

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. A Speech signal consists of Voice and Unvoiced Band in The MBE excitation model. And the energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced. We use the IMBE vocoder for the Voice or Unvoice band of segmented speech signal. Continuously we calculate the segmented SNR using that information and the energy of each band. And we estimate the SNR of continuous speech signal.

• KIEE International Transaction on Electrical Machinery and Energy Conversion Systems
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• v.5B no.4
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• pp.358-365
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• 2005

In induction machine drive without a speed sensor, the estimation of the motor flux and speed often becomes deteriorated at low speeds with low back EMF. Our analysis shows that, in addition to the state resistance variation, the estimated value of field orientation angle is often corrupted by accumulative errors from the integration of voltage variables at motor terminals that have low signal/noise ratio at low frequencies. A repetitive loop path of integration in the feedback can amplify this type of error, thus speeding up the degradation process. The control system runs into information starvation due to the loss of correct field orientation. The machine's spiral vectors are controlled only in a reduced dimension in this situation. A novel control scheme is developed to improve the control performance of motor's current, torque and speed at low frequencies. The scheme gains a full-dimensional vector control and is less sensitive to the combined effect of the error sources at the low frequencies. Experimental tests demonstrate promising performances are achievable even below 0.5 Hz.

• Journal of the Korean Society of Marine Environment & Safety
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• v.21 no.5
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• pp.571-576
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• 2015

This paper presents a comparative study of wave height estimation method that was used for signal to noise ratio and shadowing ratio based on X-band marine radar. If the signal to noise ratio, and is widely used as a method for estimating an wave height, a new method is presented for shadowing ratio. In the case of radar images used in this study it is measuring the data from the coast of Ulsan Jujeon, compared with marine meteorological information from the Meteorological Agency measured a light beacon. We compared the measured data for about 34 days, the typhoon was measured, incluidng a period in the East Sea, and verify the results for various distribution of wave height. For estimate wave height using a shadowing ratio analysis, it does not require calibration and real-time advantages of this part, coming confirmed the possibility of the measurement, the cause detection error for radar image was caused due to determine.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

• Proceedings of the KOSOMBE Conference
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• v.1994 no.12
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• pp.113-116
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• 1994

Estimation of the evoked potential using the iterated bispectrum and cross-correlation (IBC) has been tried for both simulation and real clinical data. Conventional time average (TA) method suffers from synchronization error when the latency time of the evoked potential is random, which results in poor SNR distortion in the estimation of EP waveform. Instead of EP signal average in time domain, bispectrum is used which is insensitive to time delay. The EP signal is recovered by the inverse transform of the Fourier amplitude and phase obtained from the bispectrum. The distribution of the latency time is calculated using cross-correlation between EP signal estimated by the bispectrum and the acquired signal. For the simulation. EEG noise was added to the known EP signal and the EP signal was estimated by both the conventional technique and bispectrum technique. The proposed bispectrum technique estimates EP signal more accurately than the conventional technique with respect to the maximum amplitude of a signal, full width at half maximum(FWHM). signal-to-noise-ratio, and the position of maximum peak. When applied to the real visual evoked potential(VEP) signal. bispectrum technique was able to estimate EP signal more distinctively. The distribution of the latency time may play an important role in medical diagonosis.

• ETRI Journal
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• v.45 no.3
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• pp.492-504
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• 2023

Researchers have recently shown an increased interest in estimating the direction-of-arrival (DOA) of wideband noncircular sources, but existing studies have been restricted to subspace-based methods. An off-grid sparse recovery-based algorithm is proposed in this paper to improve the accuracy of existing algorithms in low signal-to-noise ratio situations. The covariance and pseudo covariance matrices can be jointly represented subject to block sparsity constraints by taking advantage of the joint sparsity between signal components and bias. Furthermore, the estimation problem is transformed into a single measurement vector problem utilizing the focused operation, resulting in a significant reduction in computational complexity. The proposed algorithm's error threshold and the Cramer-Rao bound for wideband noncircular DOA estimation are deduced in detail. The proposed algorithm's effectiveness and feasibility are demonstrated by simulation results.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1571-1576
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• 2013

This paper proposes an improvement algorithm of signal-to-noise ratios (SNRs) for speech signals under noisy environments. The proposed algorithm first estimates the SNRs in a low SNR, mid SNR and high SNR areas, in order to improve the SNRs in the speech signal from background noise, such as white noise and car noise. Thereafter, this algorithm subtracts the noise signal from the noisy speech signal at each bands using a spectrum sharpening method. In the experiment, good signal-to-noise ratios (SNR) are obtained for white noise and car noise compared with a conventional spectral subtraction method. From the experiment results, the maximal improvement in the output SNR results was approximately 4.2 dB and 3.7 dB better for white noise and car noise compared with the results of the spectral subtraction method, in the background noisy environment, respectively.