• Journal of The Korean Association of Information Education
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• v.15 no.3
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• pp.517-524
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• 2011

This paper presents a fast blind deconvolution method that produces a deblurring result from a single image in only a few seconds. The high speed of our method is enabled by considering the Discrete Fourier Transform (DFT), and its relation to filtering and convolution, and fast computation of Moore-Penrose inverse matrix. How can we predict the behavior of an arbitrary filter, or even more to the point design a filter to achieve certain specifications. The idea is to study the frequency response of the filter. This concept leads to an useful convolution formula. A Matlab implementation of our method usually takes less than one minute to deblur an image of moderate size, while the deblurring quality is comparable.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.5 no.1
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• pp.13-20
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• 2005

This paper addresses a new blind channel equalization method using fourth-order cumulants of channel inputs and a three-layer neural network equalizer. The proposed algorithm is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum-phase characteristic of the channel. The transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel inputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple recordering and scaling. By using this estimated deconvolution matrix, which is the inverse of the over-sampled unknown channel, a three-layer neural network equalizer is implemented at the receiver. In simulation studies, the stochastic version of the proposed algorithm is tested with three-ray multi-path channels for on-line operation, and its performance is compared with a method based on conventional second-order statistics. Relatively good results, withe fast convergence speed, are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

• The Journal of the Acoustical Society of Korea
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• v.36 no.6
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• pp.378-386
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• 2017

Accurate distance measurement between maneuver target in underwater and measuring devices is required to perform quantitative test evaluation in marine weapons system R&D process. In general, the target distance is measured using a one-way ToA (Time of Arrival) method that calculates the time difference between transmitted and received signals from the two accurately synchronized devices. However, the distance estimation performance is degraded because of the multi-path environments. In this paper, the time-variant transfer function of complex underwater environment is estimated from each received data frame using RBD (Ray-based Blind Deconvolution), and the estimated time-variant transfer function is then used to get rid of the effect about complex underwater environment and to recover the data signal using PTRM (Passive Time Reversal Mirror). The result from the simulation and experimental data show that the suggested method improve the distance estimation performance when comparing with the conventional ToA method.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.375-382
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• 2016

This paper presents the results for the performance improvement of underwater communication in a passive time reversal mirror (PTRM) using ray-based blind deconvolution (RBD). In conventional PTRM, the signal to be recovered is found from matched-filtering the received probe signal. However, the communication performance is degraded because the time-varying impulse response for each data frame is not reflected in the received probe signal. In this study, the time-variant transfer function is estimated from each received data frame using RBD, and the estimated time-variant transfer function is then used to recover the data signal using PTRM. The results from the experimental data show that the suggested method improves the communication performance when comparing with the conventional PTRM.

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.242-253
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• 2022

When the source waveform is unknown, the Green's function can be estimated by Ray-based Blind Deconvolution (RBD) based on the simple array signal processing. In previous papers, RBD was successfully demonstrated using simulation and experiments in shallow water environment. In this paper, we investigate the applicability of RBD for a long-range communication (e.g., 30 km, 60 km, and 90 km) in a deep water environment (1,000 m ~), using experimental data conducted in the east of Pohang, South Korea, in October 2018. Data results are presented to demonstrate Green's function estimation of a communication signal (2.2 kHz ~ 2.9 kHz) using a 16-element, 42-m long vertical array. The results show that the Green's function estimated from RBD is comparable to that of matched filter result. Additional communication performance at a maximum range of 90 km will be also presented.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.839-845
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• 2003

This paper introduces multichannel blind source separation (BSS) and multichannel blind deconvolution (MBD) based on higher order statistics of signals from convolutive mixtures. In particular, we are concerned with the case that the number of inputs is the same as the number of outputs. Simulations for two input two output cases are carried out and their performances are assessed. One of the major applications of those sequential algorithms (BSS and MBD) is demonstrated through the fault signal detection from only a single measurement of rotating machine, which offers a certain degree of practicability in the engineering field such as machine health monitoring or condition monitoring.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.3
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• pp.15-20
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• 2002

In this paper, the method of processing a blurred noisy signal has been researched. The conventional method of processing signal has faults, which are slow-convergence speed and long time-consuming process at the singular point and/or in the ill condition. There is the process, the Gauss-Seidel's method to remove these faults, but it takes too much time because it processes signal repeatedly. For overcoming the faults, this paper shows a signal process method which takes shorter than the Gauss-Seidel's by comparing the Gauss-Seidel's with proposed algorithm and accelerating convergence speed at the singular point and/or in the ill condition. 

• The Journal of the Acoustical Society of Korea
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• v.38 no.1
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• pp.120-127
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• 2019

Passive source depth discrimination has been studied for decades since the source depth can be used for discriminating whether the target is near the surface or submerged. In this thesis, an algorithm for source depth discrimination is proposed based on CIR (Channel Impulse Response) from target-radiated noise (or signal). In order to extract CIR without a known source signal, Ray-based blind deconvolution is used. Subsequently, intersections of CIR pattern, which is characterized by ray arrival time difference, is utilized for discriminating source depth. The proposed algorithm is demonstrated through numerical simulation in ocean waveguide, and verified via the experimental data.

• Journal of Korea Multimedia Society
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• v.7 no.6
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• pp.781-790
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• 2004

This paper presents a fourth-order cumulants based iterative algorithm for blind channel equalization. It is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum phase characteristic of the channel. In this approach, the transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel outputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple reordering and scaling. Both a closed-form and a stochastic version of the proposed algorithm are tested with three-ray multi-path channels in simulation studies, and their performances are compared with a method based on conventional second-order cumulants. Relatively good results are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

• Structural Engineering and Mechanics
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• v.17 no.6
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• pp.819-828
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• 2004

A new process for estimating the natural frequency and the corresponding damping ratio in large structures is discussed. In a practical situation, it is very difficult to analyze large structures precisely because they are too complex to model using the finite element method and too heavy to excite using the exciting force method; in particular, the measured signals are seriously influenced by ambient noise. In order to identify the structural impulse response associated with the information of natural frequency and the corresponding damping ratio in large structures, the analysis process, a so-called "multiresolution blind system identification algorithm" which combines Mallat algorithm and the bicepstrum method. High time-frequency concentration is attained and the phase information is kept. The experimental result has demonstrated that the new analysis process exploiting the natural frequency and the corresponding damping ratio of structural response are useful tools in structural analysis application.