• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.335-340
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• 2014

In this paper, we propose a new sign subband adaptive filter to improve the convergence rate of the conventional sign subband adaptive filter which has been proposed to deal with colored input signal under the environment with impulsive noise. The existing sign subband adaptive filter does not increase the convergence speed by increasing the number of subband because each subband input signal is normalized by $l_2-norm$ of all of the subband input signals. We devised a new sign subband adaptive filter that normalizes each subband input signal with $l_2-norm$ of each subband input signal and increases the convergence rate by increasing the number of subband. We carried out a performance comparison of the proposed algorithm with the existing sign subband adaptive filter using a system identification model. It is shown that the proposed algorithm has faster convergence rate than the existing sign subband adaptive filter.

• Journal of information and communication convergence engineering
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• v.8 no.3
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• pp.267-272
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• 2010

For the last decade, recurrent neural networks (RNNs) have been commonly applied to communications channel equalization. The major problems of gradient-based learning techniques, employed to train recurrent neural networks are slow convergence rates and long training sequences. In high-speed communications system, short training symbols and fast convergence speed are essentially required. In this paper, the derivative-free Kalman filter, so called the unscented Kalman filter (UKF), for training a fully connected RNN is presented in a state-space formulation of the system. The main features of the proposed recurrent neural equalizer are fast convergence speed and good performance using relatively short training symbols without the derivative computation. Through experiments of nonlinear channel equalization, the performance of the RNN with a derivative-free Kalman filter is evaluated.

• Journal of electromagnetic engineering and science
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• v.16 no.1
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• pp.7-12
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• 2016

This paper presents the design of a negative group delay circuit (NGDC) using the filter synthesis approach. The proposed design method is based on a frequency transformation from a low-pass filter (LPF) to a bandstop filter (BSF). The predefined negative group delay (NGD) can be obtained by inserting resistors into resonators. To implement a circuit with a distributed transmission line, a circuit conversion technique is employed. Both theoretical and experimental results are provided for validating of the proposed approach. For NGD bandwidth and magnitude flatness enhancements, two second-order NGDCs with slightly different center frequencies are cascaded. In the experiment, group delay of $5.9{\pm}0.5ns$ and insertion loss of $39.95{\pm}0.5dB$ are obtained in the frequency range of 1.935-2.001 GHz.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.3
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• pp.224-229
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• 2009

In this paper, an efficient signal interference control technique to improve the convergence speed of Adaptive transversal filter with LMS algorithm is introduced. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the average squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS Algorithms.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.296-299
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• 2003

This paper shows a novel and simple IIR/FIR QMF(quadrature mirror filter) filter banks, mixed IIR and FIR structure. Here, FIR filters used for phase compensation. In this paper, we introduced analysis and synthesis filter banks, which used FIR linear phase filters and all pass filters. In result, phase response of analysis and synthesis filter banks become approximately linear characteristic. Simultaneously, a liasing distortion can be completely canceled.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.6
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• pp.815-823
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• 1993

In this paper, the structure of modified multiplication-free adaptive filter(M-MADF) and convergence analysis are presented. To evaluate the performance of proposed M-MADF algorithm, fractionally spaced equalizer (FSE) is used. The input signals are quantized using DPCM and the reference signals is processed using a first-order linear prediction filter, and the outputs are processed by a conventional adaptive filter. The filter coefficients are updated using the Sign algorithm. Under the assumption that the primary and reference signals are zero mean, wide-sense stationary and Gaussian, theoretical results for the coefficient misalignment vector and its autocorrelation matrix of the filter are driven. The convergence properties of Sign. MADF and M-MADF algorithm for updating of the coefficients of a 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 that MADF in the condition of same steady error. Especially it is very useful for high correlated signals.

• ETRI Journal
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• v.28 no.3
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• pp.393-396
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• 2006

This letter proposes a new impedance matching scheme of a traveling wave electro-absorption modulator (TWEAM) module for a 60 GHz band radio-over-fiber (ROF) link. A microstrip band pass filter (BPF) was used to achieve impedance matching at the 60 GHz band, and termination resistance was carefully designed to obtain an input impedance close to $50\;{\Omega}$. Also, a bias circuit for the device was designed in the module. The measured return loss and frequency response show that the modulator module observes the characteristics of a filter without the need of a further tuning process.

• Particle and aerosol research
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• v.6 no.4
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• pp.173-183
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• 2010

The high temperature pleated filter bags which were used during this study were made of pleated nonwoven fabric of heat and acid resistant polysulfonate fibers which can withstand the heat up to $300^{\circ}C$ and have a filtration area which is 3 to 5 times larger than the conventional round filter bags. Cartridge module packed with 3 kind of the sulfur impregnated activated-carbon based sorbents were inserted in the inner of the pleated filter bag. This type of pleated filter bag was designed to remove not only the particulate matter but also the gaseous elemental mercury. The electrostatic precipitator part can enhance the particulate removal efficiency and reduce the pressure drop of the pleated filter bag by agglomerated particles to form a more porous dust layer on the surface of the pleated bag which is increased the filter bag cleaning efficiency. In addition, the most of particles are separated from the flue gas stream through the cyclone and the electrostatic precipitator part which were installed at the lower part and main body part of the convergence particulate collector, respectively. Thus reduce particulate loading of the high temperature pleated filter bags were applied in this study to analyze the removal characteristics of particulate matter and gaseous elemental mercury.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.59-63
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• 2011

We present a lattice transversal joint (LTJ) adaptive filter with fixed reflection coefficients to achieve fast convergence with low complexity. The reflection coefficients of the filter are given by the statistics of speech signals, and the proposed order of the lattice predictor is one. Experimental results confirm that as compared to the adaptive transversal filter, the proposed adaptive filter achieves fast convergence with a negligible increase in complexity. The proposed adaptive filter converges around six times faster than the adaptive transversal filter in case of the band-limited voiced signal from the ITU-T G.168 standard.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.8
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• pp.3647-3653
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• 2012

In active noise control filter, IIR filter structure which used for control filter assures the stability property. The stability characteristics of IIR filter structure is mainly determined by pole location of control filter within unit disc, so stable selection of the value of control filter coefficient is very important. In this paper, we proposed novel adaptive stabilized Filtered_U LMS algorithms with IIR filter structure which has better convergence speed and less computational burden than conventional FIR structures, for multi-channel active noise control with vehicle enclosure signal case. For better convergence speed in adaptive algorithms, fuzzy LMS algorithms where convergence coefficient computed by a fuzzy PI type controller was proposed.