• Journal of information and communication convergence engineering
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• v.12 no.3
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• pp.140-144
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• 2014

A general linearly constrained adaptive array is examined in the weight vector space to illustrate the array performance with respect to the gain factor. A narrowband linear adaptive array is implemented in a coherent signal environment. It is shown that the gain factor in the general linearly constrained adaptive array has an effect on the linear constraint gain of the conventional linearly constrained adaptive array. It is observed that a variation of the gain factor of the general linearly constrained adaptive array results in a variation of the distance between the constraint plane and the origin in the translated weight vector space. Simulation results are shown to demonstrate the effect of the gain factor on the nulling performance.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.240-240
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• 2000

This paper presents two algorithms based on the concept of the frequency domain adaptive filter(FDAF). First the frequency domain recursive least squares(FRLS) algorithm with the overlap-save filtering technique is introduced. This minimizes the sum of exponentially weighted square errors in the frequency domain. To eliminate discrepancies between the linear convolution and the circular convolution, the overlap-save method is utilized. Second, the sliding method of data blocks is studied Co overcome processing delays and complexity roads of the FRLS algorithm. The size of the extended data block is twice as long as the filter tap length. It is possible to slide the data block variously by the adjustable hopping index. By selecting the hopping index appropriately, we can take a trade-off between the convergence rate and the computational complexity. When the input signal is highly correlated and the length of the target FIR filter is huge, the FRLS algorithm based on the block processing technique has good performances in the convergence rate and the computational complexity.

• Journal of Power Electronics
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• v.2 no.1
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• pp.1-10
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• 2002

An advanced active power filter for the compensation of instantaneous harmonic current components in nonlinear current load is presented in this paper. A novel signal processing technique using an adaptive neural network algorithm is applied for the detection of harmonic components generated by three-phase nonlinear current loads and this method can efficiently determine the instantaneous harmonic components in real time. The control strategy of the switching signals to compensate current harmonics of the three-phase inverter is also discussed and its switching signals are generated with the space voltage vector modulation scheme. The validity of this active filtering processing system to compensate current harmonics is substantiated on the basis of simulation results.

• Proceedings of the KSME Conference
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• 2004.11a
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• pp.585-590
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• 2004

Many machine failures are not detected well in advance due to the masking of background noise and attenuation of the source signal through the transmission mediums. Advanced signal processing techniques using adaptive filters and higher order statistics have been attempted to extract the source signal from the measured data at the machine surface. In this paper, blind deconvolution using the eigenvector algorithm (EVA) technique is used to recover a damaged bearing signal using only the measured signal at the machine surface. A damaged bearing signal corrupted by noise with varying signal-to-noise (s/n) was used to determine the effectiveness of the technique in detecting an incipient signal and the optimum choice of filter length. The results show that the technique is effective in detecting the source signal with an s/n ratio as low as 0.21, but requires a relatively large filter length.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.20 no.2
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• pp.97-102
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• 2020

This paper relates with the Hybrid-SE-MMA (Signed-Error MMA) that is possible to improving the equalization performance by using the adaptive modulus and adaptive step size in SE-MMA adaptive equalizer for the minimizing the intersymbol interference. The equalizer tap coefficient is updatted use the error signal in MMA algorithm for adaptive equalizer. But the sign of error signal is used for the simplification of arithmetic operation in SE-MMA algorithm in order to updating the coefficient. By this simplification, we get the fast convergence speed and the reduce the algorithm processing speed, but not in the equalization performance. In this paper, it is possible to improve the equalization performance by computer simulation applying the adaptive modulus to the SE-MMA which is proposional to the power of equalizer output signal. In order to compare the improved equalization performance compared to the present SE-MMA, the recovered signal constellation that is the output of the equalizer, residual isi, MD(maximum distortion), MSE and the SER perfomance that means the robustness to the external noise were used. As a result of computer simulation, the Hybrid-SE-MMA improve equalization performance in the residual isi and MD, MSE, SER than the SE-MMA.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.1
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• pp.24-30
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• 2008

In this paper we analyze the performance of an adaptive sector cell system, which is adopted to maintain the traffic balance between sectors and to utilize the cell resources effectively, using the data collected from real channel environments. In the data measurements, we transmitted the QPSK modulated signal with carrier frequency of 1.95GHz and received the signals using the 8x4 array antenna equipped on the top of buildings in the urban area. We analyzed the angular distribution and the delay spread of a user signal and analyzed angular distribution of mobile users in a cell using the collected data. Also, we propose the vector channel modeling using the estimated pdf(probability distribution function) of the analyzing results. Through the proposed channel modeling the improvement of the call blocking rate was analyzed when using the adaptive sector cell system, and computer simulations show that the call blocking rate of the adaptive sector cell system was much lower than that of the fixed sector cell system. Additionally, it shows that the call blocking rate increases severely in the fixed sector cell system while the difference of the call blocking rate was smaller in the adaptive sector cell system, as the user density of the spatial distribution increases.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.272-279
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• 2008

Shot change detection is an important technique for effective management of video data, so detection scheme requires adaptive detection techniques to be used actually in various video. In this paper, we propose an adaptive shot change detection algorithm using the mean of feature value on variable reference blocks. Our algorithm determines shot change detection by defining adaptive threshold values with the feature value extracted from video frames and comparing the feature value and the threshold value. We obtained better detection ratio than the conventional methods maximally by 15% in the experiment with the same test sequence. We also had good detection ratio for other several methods of feature extraction and could see realtime operation of shot change detection in the hardware platform with low performance was possible by implementing it in TVUS model of HOMECAST company. Thus, our algerian in the paper can be useful in PMP(portable multimedia player) or other portable players.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.54-62
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• 2000

Adaptive filtering method is one of signal processing area which is frequently used in the case of statistical characteristic change in time-varing situation. The performance of adaptive filter is usually evaluated with complexity of its structure, convergence speed and misadjustment. The structure of adaptive filter must be simple and its speed of adaptation must be fast for real-time implementation. In this paper, we propose chirp discrete cosine transform (CDCT), which has the characteristics of CZT (chrip z-transform) and DCT (discrete cosine transform), and then CDCTLMS (chirp discrete cosine transform LMS) using the above mentioned algorithm for the improvement of its speed of adaptation. Using loaming curve, we prove that the proposed method is superior to the conventional US (normalized LMS) algorithm and DCTLMS (discrete cosine transform LMS) algorithm. Also, we show the real application for the ultrasonic signal processing.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.3
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• pp.216-222
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• 2004

Frequency selective fading is a limiting factor for transmission rate and performance in high-speed multimedia communications. In this paper, we propose a turbo coded adaptive quadrature amplitude modulation (QAM) system for efficient high-speed transmission. By making use of a user-friendly simulation platform of SPW, the proposed turbo coded adaptive QAM system(TuAQAM) is developed and its performance is evaluated in terms of throughput and BER performance. Two channel models having delay spreads of 700ns and 1400ns are created for the simulations. It is shown that the proposed TuAQAM system provides a performance improvement of approximately 3dB and improved throughput for the channel model whose delay spread is 700ns. Similarly, a performance improvement is also achieved for the channel model whose delay spread is 1400ns.

• Journal of the Institute of Convergence Signal Processing
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• v.23 no.4
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• pp.185-193
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• 2022

This paper proposes an adaptive sliding mode control with radial basis function neural network(RBFNN) scheme to enhance the performance of position and attitude tracking control of quadrotor UAV. The RBFNN is utilized on the approximation of nonlinear function in the UAV dynmic model and the weights of the RBFNN are adjusted online according to adaptive law from the Lyapunov stability analysis to ensure the state hitting the sliding surface and sliding along it. In order to compensate the network approximation error and eliminate the existing chattering problems, the sliding mode control term is adjusted by adaptive laws, which can enhance the robust performance of the system. The simulation results of the proposed control method confirm the effectiveness of the proposed controller which applied for a nonlinear quadrotor UAV is presented. Form the results, it's shown that the developed control system is achieved satisfactory control performance and robustness.