• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.8C
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• pp.696-702
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• 2007

This paper proposes a regional feature preserving adaptive interpolation algorithm for natural images. The algorithm can be used in resolution enhancement, arbitrary rotation and other applications of still images. The basic idea is to first scan the sample image to initialize a 2D array which records the edge direction of all four-pixel squares, and then use the array to adapt the interpolation at a higher resolution based on the edge structures. A hybrid approach of switching between bilinear and triangulation-based interpolation is proposed to reduce the overall computational complexity. The experiments demonstrate our adaptive interpolation and show higher PSNR results of about max 2 dB than other traditional interpolation algorithms.

• Journal of Sensor Science and Technology
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• v.1 no.2
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• pp.155-163
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• 1992

The phased array antenna has the ability to perform adaptive sampling by directing the radar beam without inertia in any direction. The adaptive sampling capability of the phased array antenna allows each sampling time interval to be varied for each target, depending on the acceleration of each target at any time. In this paper we design a three-dimensional adaptive tracking algorithm for the phased array radar system with a given set of measurement parameters. The tracking algorithm avoids taking unnecessarily frequent samples, while keeping the angular prediction error within a fraction of antenna beamwidth so that the probability of detection will not be degraded during a track update illuminations. In our algorithm, the target model and the sampling rate are selected depending on the target range and the target maneuver status which is determined by a maneuver detector. A detailed simulation is conducted to test the validity of our tracking algorithm for encounter geometries under various conditions of maneuver.

• The Journal of the Acoustical Society of Korea
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• v.37 no.1
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• pp.66-72
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• 2018

In this paper, we propose an adaptive beamforming algorithm of triplet arrays for active sonar systems. The proposed algorithm consists of three steps: matched filters, cardioid beamforming, and line array beamforming. First, we apply a matched filter of a transmitted pulse to received individual sensor signals and obtain filterd signals. Then, we perform the fast Fourier transform to the matched filter results, and make a cardioid beam for each triplet data, respectively. Finally, we apply an adaptive beamforming by assuming that the cardioid beams are input signals of a line array. Experimental results demonstrate that the proposed algorithm provides better performances than conventional algorithms.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.29 no.5
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• pp.375-382
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• 2018

To eliminate strong jamming signals, a radar acquires a relatively weak target signal by using a side-lobe canceller (SLC) algorithm. This paper presents a novel adaptive SLC algorithm that is applicable to a fully digital active array radar. First, a covariance matrix is obtained from the SLC beam. Then, an adaptive SLC coefficient is extracted after calculating the correlation matrix between the main beam signal and the SLC beam signal. Finally, the target signal is estimated and the jamming signal is removed through the operation with the main beam signal. The application results from simulated radar signals demonstrated that the proposed algorithm is effective in an SLC system. Moreover, we analyzed various considerations and improved systematic usability.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.26 no.4
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• pp.424-434
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• 2015

Adaptive beamforming algorithms have been widely used to remove interference and jamming in the phased array radar system. Advances in the field programmable gate array(FPGA) technology now make possible the real time processing of adaptive beamforming (ABF) algorithm. In this paper, the FPGA based real-time implementation method of adaptive beamforming system(beamformer) in the pre-processor module for active electronically scanned array(AESA) radar is proposed. A compact FPGA-based adaptive beamformer is developed using commercial off the shelf(COTS) FPGA board with communication via OpenVPX(Virtual Path Cross-connect) backplane. This beamformer comprises a number of high speed complex processing including QR decomposition & back substitution for matrix inversion and complex vector/matrix calculations. The implemented result shows that the adaptive beamforming patterns through FPGA correspond with results of simulation through Matlab. And also confirms the possibility of application in AESA radar due to the real time processing of ABF algorithm through FPGA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.54-60
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• 2005

In this Paper, an adaptive microphone array system with self-delay estimator is proposed. By showing that the adaptive blocking matrix (ABM) of the generalized sidelobe canceller (GSC) can estimate the relative time delay between each sensor, the proposed system utilizes the ABM not only for blocking target components in the blocked signal path, but also for estimating the relative time delay. Therefore, the proposed system requires only the GSC structure while maintaining the system performance similar to the conventional system using an additional time delay estimator as a preprocessor. Simulation results show that the performance of the proposed system is identical to the conventional system that uses an additional time delay estimation module.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2E
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• pp.40-43
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• 2004

In moving jammer environments, the performance of conventional adaptive beamformer is severely degraded and the robust adaptive beamformer requires additional sensors to obtain desired performances. Therefore, it is necessary to develop efficient algorithm without any additional requirement of the number of sensors, etc. In this paper, we introduce a fast adaptive algorithm with variable forgetting factor, which does not have any additional requirements. From the computer simulations, we obtain the better performances than those of other techniques for the arrays with various aperture lengths.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.559-567
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• 2020

This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm's limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.3
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• pp.142-150
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• 1996

Monopulse technique has been widely used for a bearing estimation of a desired signal. This method can estimate the bearing of the signal with only one pulse, so computational load and complexity are low and its processes are simple. But if the desired signal incidents with interferences or multiple signals exist in the beams, the method fails to estimate correct signal angle. To solve this problem, the method to use adaptive array was proposed. In conventional adaptive array methods, sum beam was formed with fixed weights and ifference beam was formed with adaptive weights determined to minimize beamformer output power with one or more linear constraints, or an algorithm to minimize each sum beam and difference beams simultaneously using adaptively formed one basic beam. Theroretical analyses and simulation results show that the performance of the proposed method is superior to that of conventional algorithms.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.258-262
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• 2005

An adaptive beamforming system based on code filtering and differential correlation approaches is proposed. The differential correlation method was originally proposed for time delay estimation of direct sequence code division multiple access (DS-CDMA) systems under near-far ratio conditions and the code filtering correlation algorithm, on the other hand, was proposed for array response estimation in DS-CDMA systems under perfect power control. In this paper, by combining differential correlation concept with the code filtering beamforming technology, an accurate estimate of the beam forming weights and an enhanced performance of DS-CDMA systems under sever near-far ratio conditions is achieved. The system performance in terms of beam pattern and bit-error-rate (HER) shows that the proposed adaptive beamformer outperforms the conventional code filtering correlation technique.