• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF 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 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 yields more accurate detection of the start point of the NET signal.

• Journal of the korean Society of Automotive Engineers
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• v.14 no.5
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• pp.95-106
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• 1992

Active control of acoustic noise is an application area of adaptive digital signal processing with increasingly interest along the last year. This work studies the implementation of the multichannel LMS filter and the application of this algorithm for the reduction of the noise inside a vechicle cabin using a number of 'secondary sources' drived by adaptive filtering of a reference noise source. Firstly, we propose the use of an adaptive method for the time-varient optimal convergence factor. Secondly, we propose the use of adaptive delayed inverse model to estimate the elastic-acoustic transfer function presented in vechicle cabin. The original, primary source is often periodic, with a known fundamental frequency. A suitably filtered reference signal can thus be used to drive the secondary sources. An algorithm is presented for adapting the coefficients of an FIR filter feeding such a secondary source in such a way as to minimize the output of a suitably placed microphone. In this algorithm, the coefficients of adaptive filter driving an array of secondary sources can be adapted to minimize the sum of the squares of the outputs of a number of error microphones. The multichannel LMS algorithm displays that such an algorithm is considered suitable to used for the global suppression of noise in vehicle cabin.

• Economic and Environmental Geology
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• v.18 no.1
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• pp.41-47
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• 1985

The filtering ability of $J_0$ and $J_1$ digital linear filters is compared by means of an adaptive linear filter. Any $J_0$ domain Hankel transform integral can be transformed mathematically into its corresponding $J_1$ domain integral. The apparent resistivities for any electrode configuration employed in resistivity soundings can be evaluated with a single $J_1$ filter. The $J_1$ filter usually has similar accuracy to, but shorter length than, the corresponding $J_0$ filter. The domain transformation from $J_0$ to $J_1$ enables us to use effective expressions of apparent resistivity, involving $J_1$ alone, not only for Schlumberger but also for dipole-dipole array.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.7
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• pp.332-334
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• 2006

In this paper, we present an equalization scheme for 8-VSB signals for the ATSC DTV system. We propose a complex feedback filter and complex feedback sample generator for DFE to equalize 8-VSB signals in order to efficiently remove multipath distortions causing leakages from the qudrature component. We show that the proposed structure outperforms the conventional DFE used for the digital VSB which uses a real-valued feedback filter with real-valued decisions.

• Journal of Power Electronics
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• v.15 no.6
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• pp.1619-1627
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• 2015

This paper presents a power system harmonic elimination using the mixed adaptive linear neural network and variable step-size leaky least mean square (ADALINE-VSSLLMS) control algorithm based active power filter (APF). The weight vector of ADALINE along with the variable step-size parameter and leakage coefficient of the VSSLLMS algorithm are automatically adjusted to eliminate harmonics from the distorted load current. For all iteration, the VSSLLMS algorithm selects a new rate of convergence for searching and runs the computations. The adopted shunt-hybrid APF (SHAPF) consists of an APF and a series of 7^th tuned passive filter connected to each phase. The performance of the proposed ADALINE-VSSLLMS control algorithm employed for SHAPF is analyzed through a simulation in a MATLAB/Simulink environment. Experimental results of a real-time prototype validate the efficacy of the proposed control algorithm.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.6
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• pp.28-34
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• 1993

An algorithm for MMADF(modified multiplication-free adaptive filter) which need not to multiplication arithmatic operation is proposed to improve the performance of FSE (fractionally spaced equalizer) which reduce the ISI(intersymbol interference) in signal transfer channel. The input signals are quantized using DPCM and the reference signals is processed using a first-order linear prediction filter. The convergence properties of Sign. MADF and M-MADF algorithm for updating of the coefficients of a FIR digital filter of the fractionally spaced equalizer (FSE) are investigated and compared with one another. The convergence properties are characterized by the steady state error and the convergence speed. It is shown that the convergence speed of M-MADF is almost same as Sign algorithm and is faster than MADF in the condition of same steady state error. Especially it is very useful for high correlated signals.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.498-503
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• 2005

This paper is concerned with the implementation of adaptive positive position feedback controller using a digital signal processor and microcontroller The main advantage of the positive position feedback controller is that it can control a natural mode of interest by tuning the filter frequency of the positive position feedback controller to the natural frequency of the target mode. However, the positive position feedback controller loses its advantage when mistuned. In this paper, the fast fourier transform algorithm is implemented on the microcontroller whereas the positive position feedback controller is implemented on the digital signal processor. After calculating the frequency which affects the vibrations of structure most the result is transferred to the digital signal processor. The digital signal processor updates the information on the frequency to be controlled so that it can cope with both internal and external changes. The proposed scheme was installed and tested using a beam equipped with piezoceramic sensor and actuator. The experimental results show that the adaptive positive position feedback controller proposed in this paper can suppress vibrations even when the target structure undergoes structural change thus validating the approach.

• Journal of Korea Multimedia Society
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• v.4 no.3
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• pp.222-229
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• 2001

Many techniques have been proposed for digital image enlargement, which use spatially neighbored pixels information in a still image. In this paper, we propose the digital image interpolation method that improves edge characteristics by selectively transposing the sub-windows of pseudomedian filter, which results in relatively better performance than others. We have simulated the proposed algorithm using Visual C++ and verified performance of the algorithm by PSNR(Peak signal Noise Ratio) and edge characteristics. Finally, we have designed the adaptive pseudomedian by using synopsys VHDL(Very high speed integrated circuit Hardware Description Language).

• Proceedings of the KIPE Conference
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• 2009.11a
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• pp.209-211
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• 2009

A digital grid synchronization technique is needed for distributed generation system to make output current sinusoidal even if the grid voltage is distorted by harmonics. In this paper, a digital grid synchronization technique based on adaptive notch filter is proposed. The analysis of proposed technique is performed through the consideration of grid synchronization technique based on PLL and FLL, and the validity of the proposed method was confirmed by simulation results.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.272-279
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• 2016

In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.