• 제어로봇시스템학회:학술대회논문집
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• 1997.10a
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• pp.1379-1382
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• 1997

We introduce a fast implementation of the adaptive transversal filter which uses least-mean-square(LMS) algorithm. The fast Hadamard transform(FHT) is used for the implementation of the filter. By using the proposed filter we can get the significant time reduction in computatioin over the conventional time domain LMS filter at the cost of a little performance. By computer simulation, we show the comparison of the propsed Hadamard-domain filter and the time domain filter in the view of multiplication time, mean-square error and robustness for noise.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.4
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• pp.673-677
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• 2003

In this study, we propose the new filter bank that is adaptive filter bank using neural networks in time domain. Also, we proposed a new filter neuron as neuron with filter window, the structure and algorithm for filter banks. The performance of neural filter banks is shown from two examples. It show characteristics the simple structure and higher speed processing than traditional methods (filter banks in frequency domain, etc.). In many applications, the proposed method will provide the high performance to features detection of signals in time domain.

• Journal of the Korean Institute of Telematics and Electronics
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• v.22 no.2
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• pp.18-24
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• 1985

Above certain filter order, the frequency domain -adaptive filter is superior to the time domain-adaptive filter in computational complexity. In this paper a new type algorithm, $\mu$-FLMs algorithm, is proposed for the frequency domain- adaptive filter and the characteristics of the proposed algorithm is compared with that of the time domain- adaptive filter algorithm($\mu$-FLMS algorithm). The simulation results showed that under the same convergence rate , the frequency domain-adaptive filter is efficient in compu tational burden.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.33 no.7
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• pp.59-69
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• 2005

A position domain Hatch filter is proposed as an efficient carrier-smoothed-code processing algorithm for real-time kinematic differential global satellite navigation systems. The well-known range domain Hatch filter is newly interpreted with a stochastical point of view. The interpretation result is extended to derive the position domain Hatch filter. By a covariance simulation, it is shown that Hatch gain is, in general, more efficient than Kalman-type gain in carrier-smoothed-code processing and the proposed position domain Hatch filter is more advantageous than the conventional range domain Hatch filter if the visible satellite constellation changes during the positioning task.

• Journal of Communications and Networks
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• v.9 no.1
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• pp.75-83
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• 2007

Orthogonal frequency division multiplexing (OFDM) systems with multiple transmit antennas can exploit space-time block coding on each subchannel for reliable data transmission. Spacetime coded OFDM systems, however, are very sensitive to time variant channels because the channels need to be static over multiple OFDM symbol periods. In this paper, we propose to mitigate the channel variations in the frequency domain using a linear filter in the frequency domain that exploits the sparse structure of the system matrix in the frequency domain. Our approach has reduced complexity compared with alternative approaches based on time domain block-linear filters. Simulation results demonstrate that our proposed frequency domain block-linear filter reduces computational complexity by more than a factor of ten at the cost of small performance degradation, compared with a time domain block-linear filter.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.34 no.2
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• pp.48-55
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• 2006

Performance characteristics of the position-domain Hatch filter is analyzed for differential global navigation satellite systems. It is shown that the position-domain Hatch filter generates white measurement residual sequences, which is beneficial property for fault detection. It is also shown that the position-domain Hatch filter yields more accurate a priori state estimate than the position-domain Kalman-type filter. Thus, it can be concluded that the position-domain Hatch filter is beneficial in wide application areas where fault-tolerance and accuracy are required at the same time.

• Korean Journal of Fisheries and Aquatic Sciences
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• v.30 no.1
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• pp.16-30
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• 1997

Application of digital filter to the wave analysis is studied using the observed data by wave gauge. Sea wave data obtained from wave gauge always include long wave frequency components. In order to estimate the sea wave parameters, we must re-analyzed wave data by using a digital filter and the concept of mean sea level correction method. By the wave by wave analysis and spectral methods, sea wave parameters on the basis of wave data obtained by the conventional method and digital filter are compared. The best-fitted design filter determined by the necessary conditions of frequency responses, can be obtained by calculating various transfer functions. Thus, to get the best the digital filter design, both Butterworth filter and Savitzky-Golay filter of digital filter are used in the frequency and time domain, respectively. Three cases of observation wave data are calculated by applying digital filter. The components of different frequency bands in the surf zone are coexisted in three cases. The wave data for wind wave components is computed using the digital filter the surf zone and off-surf zone, and based on the filtered data, wave parameters are calculated by the spectral analysis and wave by wave analysis methods, respectively. As a results, when sea wave data observed by wave gauge are analyzed, the Savitzky-Golay method is recommended which can well appear cut-off frequency by experimental choosing filter length in the time domain. The better mean sea level correction method is the Butterworth filter in the frequency domain.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.9
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• pp.1188-1197
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• 1995

This paper describes an algorithm that reduces computational requirement of the Kalman filter and estimates the signal efficiently. The reference signals are mapped onto the orthogonal wavelet transform domain so that the eigenvalue spread of its autocorrelation matrix could be smaller than that in the time domain. In the wavelet transform domain the autocorrelation matrix is nearly diagonal. Therefore, the transformed signal can be decomposed each orthogonal elements. The Kalman filter can be applied to each orthogonal elements and computational requirement is reduced. The possibility of applying the parallel Kalman filter was verified through the theory and simulation. The eigenvalue spread in the wavelet transform domain is smaller 8.35 times than that in the time domain and the computational requirement is reduced from 1.4 times to 2. 93 times than that of the conventional Kalman filter.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2E
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• pp.38-46
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• 1997

The objective of the paper is to compare the computational complexity of five algorithms for computing time-domain second-order Volterra filter outputs in terms of number of real multiplication and addition operations required for implementation. This study shows that if the filter memory length is greater that or equal to 16, the fast algorithm using the overlap-save method and the frequency-domain symmetry properties of the quadratic coefficients is the most efficient among the algorithms investigated in this paper, When the filter memory length is less than 16, the algorithm using the time-domain symmetry properties is better than any other algorithm.

• MALSORI
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• no.67
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• pp.195-216
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• 2008

This paper addresses the problem of single-channel speech separation to extract the speech signal uttered by the speaker of interest from a mixture of speech signals. We propose to apply time-frequency smoothing to the existing statistical single-channel speech separation algorithms: The soft mask and the minimum-mean-square-error (MMSE) algorithms. In the proposed method, we use the two smoothing later. One is the uniform mask filter whose filter length is uniform at the time-Sequency domain, and the other is the met-scale filter whose filter length is met-scaled at the time domain. In our speech separation experiments, the uniform mask filter improves speaker-to-interference ratio (SIR) by 2.1dB and 1dB for the soft mask algorithm and the MMSE algorithm, respectively, whereas the mel-scale filter achieves 1.1dB and 0.8dB for the same algorithms.