• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1577-1580
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• 2004

This paper proposes a design and implementation of transversal adaptive digital filter using LMS (Least Mean Squares) adaptive algorithm. The filter structure is based on Distributed Arithmetic (DA) which is able to calculate the inner product by shifting and accumulating of partial products and storing in look-up table, also the desired adaptive digital filter will be multiplierless filter. In addition, the hardware implementation uses VHDL (Very high speed integrated circuit Hardware Description Language) and synthesis using FLEX10K Altera FPGA (Field Programmable Gate Array) as target technology and uses Leonardo Spectrum and MAX+plusII program for overall development. The results of this design are shown that the speed performance and used area of FPGA. The experimental results are presented to demonstrate the feasibility of the desired adaptive digital filter.

• The Transactions of the Korean Institute of Power Electronics
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• v.26 no.5
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• pp.307-314
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• 2021

Even though the LCL filters have superior harmonic attenuation ability to L filters, stability has always been an issue. The system could be unstable because of the resonance phenomenon, especially when digital controller is used. Adding a notch filter to the compensator is one approach to solve the problem. Resonance phenomenon can be inhibited by aligning notch frequency to system resonance frequency. However, resonance frequency variation can be obtained because the actual system has a nonstationary characteristic. Therefore, the system could be unstable, where the system parameters are changed when the conventional notch filter is used. An adaptive digital notch filter that stabilizes the system even system parameters are changed. Simulation and experiment results are provided to verify the validity of the proposed adaptive filter.

• The Journal of the Acoustical Society of Korea
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• v.6 no.2
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• pp.15-18
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• 1987

Multiplication free adaptive digital filtering algorithms are discussed. The proposed. The proposed algorithm uses delta modulation digital filter and the relevant filter weights are updated using the SIGN algorithms to realize an adaptive digital filter without multiplication operations. It is shown that the resulting algorithm can be implemented using simple up/down counting operations. The convergence characteristics of the proposed adaptive digital filtering algorithm and .others are investigated for a system identification problem.

• Journal of Electrical Engineering and Technology
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• v.7 no.3
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• pp.451-458
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• 2012

The voluntary EMG (vEMG) signal from electrically stimulated muscle is very useful for feedback control in functional electrical stimulation. However, the recorded EMG signal from surface electrodes has unwanted stimulation artifact and M-wave as well as vEMG. Here, we propose an event-synchronous adaptive digital filter for the suppression of stimulation artifact and M-wave in this application. The proposed method requires a simple experimental setup that does not require extra hardware connections to obtain the reference signals of adaptive digital filter. For evaluating the efficiency of this proposed method, the filter was tested and compared with a least square (LS) algorithm using previously measured data. We conclude that the cancellation of both primary and residual stimulation artifacts is enhanced with an event-synchronous adaptive digital filter and shows promise for clinical application to rehabilitate paretic limbs. Moreover because this algorithm is far simpler than the LS algorithm, it is portable and ready for real-time application.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2035-2038
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• 2002

The necessity of the filter coefficients compensation for the LTJ adaptive filter was explained generally and easily by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%

• Journal of Biomedical Engineering Research
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• v.22 no.3
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• pp.285-291
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• 2001

In this paper, we designed two preprocessing adaptive filter - high pass filter and notch filter - using FPGA. For minimizing the calculation load of multi-channel and high-resolution ECG system, we utilize FPGA rather than digital signal processing chip. To implement the designed filters in FPGA, we utilize FPGA design tool(Altera corporation, MAX-PLUS II) and CSE database as test data. In order to evaluate the performance in terms of processing time, we compared the designed filters with the digital filters implemented by ADSP21061(Analog Devices). As a result, the filters implemented by FPGA showed better performance than the filters based on ADSP21061. As a consequence of examination, we conclude that FPGA is a useful solution in multi-channel and high-resolution signal processing.

• The Journal of the Acoustical Society of Korea
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• v.27 no.3E
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• pp.95-103
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• 2008

In this paper, the new hybrid adaptive Volterra filter was proposed to be applied for compensating the nonlinear distortion of memoryless nonlinear systems with saturation characteristics. Through computer simulations as well as the analytical analysis, it could be shown that it is possible for both conventional Volterra filter and proposed hybrid Volterra filter, to be applied for linearizing the memoryless nonlinear system with nonlinear distortion. Also, the simulations results demonstrated that the proposed hybrid filter may have faster convergence speed and better capability of compensating the nonlinear distortion than the conventional Volterra filter.

• 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.

• 제어로봇시스템학회:학술대회논문집
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• 2002.10a
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• pp.87.3-87
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• 2002

1. Introduction 2. Overview of Collected Data 3. Adaptive Digital Filter 4. Experiment and Results 5. Conclusion