• 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.

• Journal of the Korea society of information convergence
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• v.5 no.2
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• pp.101-108
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• 2012

In the present paper, it removed the low frequency noise under 1Hz which get up base wandering from the various noise which is included in ECG signals. It used wavelet filter, FIR filter and Adaptive FIR filter and compared the efficiency of the filter. The set condition of 3 kind filters which are the comparative object is the next contents. Used wavelet case, used generating functions db7 and after decomposition, the low frequency of 8 phases (cA8) replaced at 0. FIR filter case, filter coefficient set 1400, cutoff frequency (${\omega}$) set 0.002. Adaptive FIR filter case, collecting coefficients (${\mu}$) with 0.005. The comparative result from the output wave shape and FT spectrum, wavelet is excellent in base wandering removals compared FIR filter and Adaptive FIR filter. And SNR comparisons, wavelet filter(44.16) is high compare with other two filters(25.19 and 15.94).

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.3
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• pp.1-8
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• 1996

This paper proposes a novel transversal filter and an optimal LMS algorithm, and show how these can be realized as an adaptive equalizer. The transversal filter consists of a walsh and block pulse functions. in the LMS algorithm with equalizers, the convergence factor is an improtant design parameter because it governs stability and convergence speed. The conventional adaptation techniques use a fixed time constant convergence factor by the trial and error method. In this paper, an optimal method in the choice of the convergence factor is proposed. The proposed algorithm is obtrained that is tailored for each filter tap and is updated at each iteration. The performance of the proposed algorithm is compared iwth those of the conventional TDL and DFT equalizers by computer simulations.

• Journal of the Korean Institute of Telematics and Electronics
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• v.24 no.4
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• pp.719-726
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• 1987

In this paper, an adaptive IIR (infinite impulse response) notch filter of lattice type is constructed and its adaptation algorithm is proposed for the detection and retrieval of a sine wave signal embedded in noise. A modified method which adapts only one coefficient of the filter is also suggested. All these methods adapt the coefficients while keepting the poles of the filter inside the unit circle on z-plane, and thus they satisfy the condition on the stability of the IIR filter after it has converged. To investigate the convergence characteristics of these methods such as convergence speed and output S/N ratio, intensive computer simulation has been performed by varying the frequency of the sine wave and the input S/N ratio. And the results of the simulation have been compared to those of Rao and Kung's which shows relatively fast convergence speed. The methods proposed here, especially the second one. shows faster convergence speed and higher output S/N ratio than the Rao and Kung's.

• 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.

• The Pure and Applied Mathematics
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• v.12 no.2 s.28
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• pp.105-124
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• 2005

In this paper convergence of fuzzy filters and graded fuzzy filters have been studied in graded L-fuzzy topological spaces.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.45-48
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• 2001

In this paper, the characteristic of the equalizer using Lattice filter was investigated in channel ISI (Inter-Symbol Interference) in OFDM (orthogonal frequency division multiplexing) system. The equalizer using lattice filter has more fast convergence and little equalization error characteristic in two number of tap by orthogonal effect of each tap than another equalizers. The filter coefficient convergency and static BER (bit error ratio) characteristic was analysed by computer simulation. In this paper, it is shown that the equalizer using lattice filter has the better performance than a equalizer which makes use of another equalization method.

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.3
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• pp.238-243
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• 2010

A new implementation method for MPDSAP(Maximally Polyphase Decomposed Subband Affine Projection) adaptive filter is proposed. The affine projection(AP) adaptive filter achieves fast convergence speed, however, its implementation is so expensive because of the matrix inversion for a weight-updating of adaptive filter. The maximally polyphase decomposed subband filtering allows the AP adaptive filter to avoid the matrix inversion, moreover, by using a pipelining technique, the simple subband structured AP is suitable for VLSI implementations concerning throughput, power dissipation and area. Computer simulations are presented to verify the performance of the proposed algorithm.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.9 no.1
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• pp.59-64
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• 2009

In this paper, we study the relations between L-fuzzy topologies and L-filters on a strictly two-sided, commutative quantale lattice L. We define an L-fuzzy neighborhood filter and introduce the notion of L-filter convergence in L-fuzzy topological spaces.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.357-360
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• 2000

In this paper, we want to implement a noise canceller by using FIR filter on DSP(Digital Signal Processor). The FIR filter was designed by Blackman window together with desired band width and center frequency. We adopt Motorola DSP56002 and Crystal CS4215 (A/D and D/A converter) for our purpose. we generate input sinusoidal signals and noises by differential equations and pseudo random sequences on DSP also. The input signal including sinusoidal and noise passes through the FIR filter. The FIR filer output is a sinusoidal signal with noise reduced.