• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.53-56
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• 2006

In this paper, we present the algorithm to enhance the high frequency components of the image by subtracting smoothing version of an image from the original image. In a word, this is the technique to get more a precise and vivid image. The method of Gaussian smoothing is accepted to get the components of the flat image. Moreover we need to consider the size of the gain of the proposed filter in order to preserve the overall brightness of a image. Based on the algorithm verified by MATLAB, we can obtain more vivid and fuller detail of the image than an original image.

• Journal of Korea Society of Digital Industry and Information Management
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• v.5 no.4
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• pp.11-17
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• 2009

Polarography is a subclass of voltammetry where the working electrode is a dropping mercury electrode. More, I developed the experiment for raising up mercury electrode. In Voltammetry, information about an analyte is obtained by measuring the current as the potential is varied at oxidation-reduction reaction. A plot of current vs. potential in a polarography experiment shows the current oscillations correspoding to the drops of Hg falling from the capillary. The drops growth causes capacitive and faradic current. These changing current effects combined with experiments where the potential is continuously changed can result in noisy traces. The raising up type improved upon the method of dropping in hardware. In reduction of the noise, moving average smoothing method have been used. But the other procedure is introduced based on Fourier transformation. So FFT and IFFT engine was implemented and installed in my experiment. However, after experimentation, peak height as the measuring parameter gave a good linear relationship to concentration. The resolution of potential peaks of various kinds, using Zn and Cu as the example, was improved using the smoothing method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.2
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• pp.227-236
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• 2001

This paper proposes a frequency-domain partially adaptive algorithm, called a censoring algorithm, to reduce the computational complexity of the frequency domain adaptive array. The proposed censoring algorithm determines the existence of interferences in the frequency-domain at each frequency bin using a constant false alarm rate (CFAR) processor. The censoring algorithm adapts only those parts of the weights that correspond to the frequency bins expected to contain interferences. The censoring algorithm is also expanded to overcome the signal cancellation phenomenon caused by smart jammers. Accordingly, a censoring spatial smoothing, which combines the censoring algorithm with spatial smoothing, is proposed. Simulation results show that the proposed algorithms are effective in removing interferences with only part of the computational complexity of conventional algorithms yet with the same level of performance.

• Journal of Broadcast Engineering
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• v.20 no.6
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• pp.880-887
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• 2015

In this paper, frame rate up conversion (FRUC) algorithm considering the direction and magnitude of identical motion vectors is proposed. extended bilateral motion estimation (EBME) has higher complexity than bilateral motion estimation (BME). By using average magnitude of motion vector with x and y direction respectively, dynamic frame and static frame are decided. We reduce complexity to decide EBME. also, After we compare the direction and magnitude of identical motion vectors, We reduce complexity to decide motion vector smoothing(MVS). Experimental results show that this proposed algorithm has fast computation and better peak singnal to noise ratio(PSNR) results compared with EBME.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.1
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• pp.30-37
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• 1989

The MUSIC(MUltiple SIgnal Characterization)algorithm has been extenced to the UCERSS(Unit Circle Eigendecomposition Rational Signal Subspace) by taking eigendecimposition on the unit circle in order to estimate incident angles of multiple wide band signals. The purpose of this paper is to presetn SSB-UCERSS(Signal Subspace Based UCERSS) and SS-UCERSS(Spatially Smoothed UCERSS) estimating the incident angles of multiple side band signals in multipath(coherent signals) environments. Computer simulation results indicate that SSB-UCERSS yields the best result, while the SS-UCERSS performs better than the UCERTSS in a multipath environment.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.3
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• pp.500-506
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• 2019

In this study, we present a decision function of optimal smoothing parameter for baseline correction using Asymmetrically reweighted penalized least squares (arPLS). Baseline correction is very important due to influence on performance of spectral analysis in application of spectroscopy. Baseline is often estimated by parameter selection using visual inspection on analyte spectrum. It is a highly subjective procedure and can be tedious work especially with a large number of data. For these reasons, an objective procedure is necessary to determine optimal parameter value for baseline correction. The proposed function is defined by modeling the median value of possible parameter range as the length and order of the background signal. The median value increases as the length of the signal increases and decreases as the degree of the signal increases. The simulated data produced a total of 112 signals combined for the 7 lengths of the signal, adding analytic signals and linear and quadratic, cubic and 4th order curve baseline respectively. According to the experimental results using simulated data with linear, quadratic, cubic and 4th order curved baseline, and real Raman spectra, we confirmed that the proposed function can be effectively applied to optimal parameter selection for baseline correction using arPLS.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.7 s.98
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• pp.740-745
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• 2005

This paper concerns estimating a desired signal in a multipath environment using linearly constrained array processor with master-slave type array processing and spatial smoothing method in GSM system. In computer simulation, it is shown that the spatial smoothing approach performs better than the master-slave type array processing while both methods perform better than linearly constrained array processing with respect to SINR and BER performances.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.11b
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• pp.699-704
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• 2002

A method using beamforming algorithm has been developed to measure oblique incidence reflection coefficients of sound absorption materials. MUSIC(Multiple Signal Classification) method detects the angles of incidence and reflection. By separating the incident and reflected waves using beamforming method, the reflection coefficient is calculated. Spatial smoothing technique is also used to reduce the coherence between the incident and reflected waves. The test materials were modeled as a locally reacting surface. Numerical and experiment results are performed to verify the acuracy of proposed method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.7
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• pp.709-718
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• 1992

A new edge detection operator superimposing two displaced Gaussian smoothing filters Is proposed as an approximate operator for the DroG(flrst derivative of Gaussian) known as a sub-op-timal step edge detector. The performance of the proposed edge detector Is very close to that of the DroG with the performance criteria . signal to noise ratio, locality, and multiple response. And the computational complexity can be reduced almost by a half of that of DroG, because of the use of common 2-D smoothing filter for DroG and LoG ( Laplacian of Gausslan) systems.

• MALSORI
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• no.67
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• pp.195-216
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• 2008

This paper addresses the problem of single-channel speech separation to extract the speech signal uttered by the speaker of interest from a mixture of speech signals. We propose to apply time-frequency smoothing to the existing statistical single-channel speech separation algorithms: The soft mask and the minimum-mean-square-error (MMSE) algorithms. In the proposed method, we use the two smoothing later. One is the uniform mask filter whose filter length is uniform at the time-Sequency domain, and the other is the met-scale filter whose filter length is met-scaled at the time domain. In our speech separation experiments, the uniform mask filter improves speaker-to-interference ratio (SIR) by 2.1dB and 1dB for the soft mask algorithm and the MMSE algorithm, respectively, whereas the mel-scale filter achieves 1.1dB and 0.8dB for the same algorithms.