• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.1
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• pp.25-35
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• 2022

The SAP(salt-and-pepper) noise changes the pixel value to the maximum and minimum values of the dynamic region of the pixel. For this reason, unlike white Gaussian noise, SAP noise can predict the ratio of noise relatively easily. Because the condition of the neighboring pixels that can be referenced changes according to the noise ratio, it is necessary to apply different noise reduction methods according to the noise ratio. This paper proposes an adaptive switching filtering algorithm which can eliminates the SAP noise. It consists of two phases. It first detects the location of the SAP noise and calculates the noise ratio. After that, the image is reconstructed using different methods depending on which of the three sections the calculated noise ratio belongs to. As a result of the experiment, the proposed method showed superior objective and subjective image quality compared to the previous methods such as MF, AFSWMF, NAMF and RWMF.

• Journal of the Korea institute for structural maintenance and inspection
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• v.23 no.7
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• pp.9-15
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• 2019

The purpose of present study is to assess the maximum width of the surface cracks using the histogram analysis of image processing techniques in concrete structures. For this purpose, the concrete crack image is acquired by the camera. The image is Grayscale coded and Binary coded. After Binary coded image is Dilate and Erode coded, the image is then recognized as separated objects by applying Labeling techniques. Over time, dust and stains may occur naturally on the surface of concrete. The crack image of concrete may include shadows and reflections by lighting depending on a surrounding conditions. In general, concrete cracks occur in a continuous pattern and noise of image appears in the form of shot noises. Bilateral Blurring and Adaptive Threshold apply to the Grayscale image to eliminate these effects. The remaining noises are removed by the object area ratio to the Labeled area. The maximum numbers of pixels and its positions in the crack objects without noises are calculated in x-direction and y-direction by Histogram analysis. The widths of the crack are estimated by trigonometric ratio at the positions of the pixels maximum numbers for the Labeled objects. Finally, the maximum crack width estimated by the proposed method is compared to the crack width measured with the crack gauge. The proposed method by the present study may increase the reliability for the estimation of maximum crack width using image processing techniques in concrete surface images.

• The Journal of the Acoustical Society of Korea
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• v.10 no.1
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• pp.20-29
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• 1991

A fundamental assumption in conventional linear predictive coding (LPC) analysis procedure is that the input to an all-pole vocal tract filter is white process. In the case of periodic inputs, however, a pitch bias error is introduced into the conventional LP coefficient. Multi-pulse (MP) LP analysis can reduce this bias, provided that an estimate of the excitation is available. Since the prediction error of conventional LP analysis can be modeled as the sum of an MP excitation sequence and a random noise sequence, we can view extracting MP sequences from the prediction error as a classical detection and estimation problem. In this paper, we propose an algorithm in which the locations and amplitudes of the MP sequences are first obtained by applying a likelihood ratio test (LRT) to the prediction error, and LP coefficients free of pitch bias are then obtained from the MP sequences. To verify the performance enhancement, we iterate the above procedure with adaptive threshold at each step.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1068-1077
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• 2007

This paper proposes a method that extracts characteristics of cracks such as length, thickness and direction from a concrete slab surface image with image processing techniques. These techniques extract the cracks from the concrete surface image in variable conditions including bad image conditions) using the ART2-based RBF network to recognize the dominant directions -45 degree, 45 degree, horizontal and vertical) of the extracted cracks from the automatically calculated specifications like the lengths, directions and widths of the cracks. Our proposed extraction algorithms and analysis of the concrete cracks used a Robert operation to emphasize the cracks, and a Multiple operation to increase the difference in brightness between the cracks and background. After these treatments, the cracks can be extracted from the image by using an iterated binarization technique. Noise reduction techniques are used three separate times on this binarized image, and the specifications of the cracks are extracted form this noiseless image. The dominant directions can be recognized by using the ART2-based RBF network. In this method, the ART2 is used between the input layer and the middle layer to learn, and the Delta learning method is used between the middle layer and the output layer. The experiments using real concrete images showed that the cracks were effectively extracted, and the Proposed ART2-based RBF network effectively recognized the directions of the extracted cracks.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.18 no.2 s.117
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• pp.177-182
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• 2007

This paper presents the InGaP/GaAs HBT differential LC VCO with low phase noise performance for adaptive feedback interference cancellation system(AF-lCS). The VCO is verified with enhanced tank structure including filtering technique. The output tuning range for proposed VCO using asymmetric inductor and symmetric capacitors withlow pass filtering technique is 207 MHz. The output powers are -6.68 including balun and cable loss. The phase noise of this VCO at 10 kHz, 100 kHz and 1 MHz are -102.02 dBc/Hz, -112.04 dBc/Hz and -130.40 dBc/Hz. The VCO is designed within total size of $0.9{\times}0.9mm^2$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9A
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• pp.905-915
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• 2007

This paper proposes a filter-based algorithm to adaptively adjust the contention window in IEEE 802.11 DCF. The proposed mechanism is focused on the general and realistic environments that have various conditions regarding to noise, media types and network load. For this flexible adaptation, Filter-based DCF(FDCF) takes a more realistic policy such as median filter concept in the image processing technologies. We can handle these various environments by adjusting the contention window size according to the result of filtering based on history-buffer. We can ignore temporarily and randomly occurred transmission failures due to noise errors and collisions in noisy environments. In addition, by changing the reference number and history-buffer size, FDCF can be extended as a general solution including previous proposed mechanism. We have confirmed that the proposed mechanism can achieve the better performance than those of previous researches in aspects of the throughput and the delay in the realistic environments.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.18 no.8
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• pp.18-24
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• 2017

The process of transmitting messages through a medium with a limited bandwidth or channel dispersion inevitably involves signal distortion and noise influxes, resulting in the degradation of transmission quality due to the inter-symbol interference and additional noise, which increases the error rate of the received symbols. The main role of the equalizer is to remove the channel distortion and noise from the received signal to recover the transmitted messages. A number of studies on the equalizer composed of a combination of linear filter and error control coding have shown that they played a key role in enhancing the transmission efficiency, which is essential for digital communication. This paper proposes a new algorithm to correct the residual symbol errors in the message signal. In general, equalizer performance improvement algorithms were developed to improve the initial convergence speed or steady-state error. In this paper, however, the equalizer input signal was reconstructed using the equalizer decision symbols and the channel estimates to directly correct the decision errors by analyzing the statistical characteristics of the difference signal between the actual received signal and the reconstructed signal.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.289-294
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• 2002

A valid speech-sound block can be classified to provide important information for speech recognition. The classification of the speech-sound block comes from the MRA(multi-resolution analysis) property of the DWT(discrete wavelet transform), which is used to reduce the computational time for the pre-processing of speech recognition. The merging algorithm is proposed to extract valid speech-sounds in terms of position and frequency range. It needs some numerical methods for an adaptive DWT implementation and performs unvoiced/voiced classification and denoising. Since the merging algorithm can decide the processing parameters relating to voices only and is independent of system noises, it is useful for extracting valid speech-sounds. The merging algorithm has an adaptive feature for arbitrary system noises and an excellent denoising SNR(signal-to-nolle ratio).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.23 no.4
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• pp.431-437
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• 2019

PLC(Power Line Communication) is being used in various ways in smart grid system because of the advantages of low cost and high data throughput. However, power line channel has many problems due to impulse noise and various studies have been conducted to solve the problem. Recently, ACDL(Adaptive Cannonical Differential Limiter) which is based on an adaptive clipping with analog nonlinear filter, has been proposed and performs better than the others. In this paper, we show that ACDL is similar to the detection of slew rate with oversampled digital signal by simplification and analysis. Through the simulation under the PRIME standard it is shown that the proposed performs equal to or better than that of ACDL, but significantly reduce the complexity to implement. The BER performance is equal but the complexity is reduced to less than 10%.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.417-427
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• 2021

Long-range communication in deep waters must overcome the low data rate due to limited bandwidth. This paper presents the performance of Multiple-Input-Multiple-Output (MIMO) system to increase the data rate. In MIMO system, communication performance is degraded by crosstalk between users and an adaptive passive Time Reversal Processing (TRP) is widely used to eliminate this. In October 2018, long-range underwater acoustic communication experiment was conducted in deep water (1,000 m ~) off the east of Pohang, South Korea. During the experiment, a vertical line array was utilized and communication signals modulated by binary phase shift keying and quadrature phase shift keying with a symbol rate of 512 sps were transmitted. To generate MIMO communication signals, received signals from ranges of 26 km and 30 km is synthesized. Compared to the conventional passive TRP, the adaptive passive TRP eliminates the crosstalk between users and achieves error-free performance with an increase of output signal-to-noise ratio. Therefore, two users separated by 4 km in range achieves an aggregate data rate of 1,024 symbols/s.