• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.30-32
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be Improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.2
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• pp.143-150
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• 2004

In this paper, a recursive digital filter realization for an ionospheric channel model is proposed. This realization is in the form of a cascade of identical second-order all-pass filters, and is determined by only three parameters; two coefficients of an all-pass section, and the number of sections. The values of these parameters are optimized by a nonlinear optimization algorithm called the "downhill simplex method", so that the resulting time delay function closely approximates that of the ionospheric channel model. Comparing with the nonrecursive digital filter realization, it can be shown that the proposed recursive-digital-filter-realization is advantageous in points of view for the numbers of filter coefficients and the realization.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.6
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• pp.19-28
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• 1995

In this paper, measured iris transmission loss data and simulated data by using 3-dimension full-wave analysis S/W are presented and compared with Marcuvitz's theory. And by using measured iris data, dual-mode narrow-band channel filters can be successfully implemented. This paper shows that there is severe difference between the transmission loss of iris calculated by using Marcuvitz's equation to calculate iris dimension, if the length of slot iris is longer than .lambda./.pi., and in the long urn the response of channel filter is distorted. Experimental result shows that the characteristic response of implemented channel filter by using the iris transmission loss graph presented here matches well the design specfications. In conclusion, iris transmission loss measurement method will be very useful to design channel filter.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.11 no.5
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• pp.723-730
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• 2000

In this paper, we perform a performance analysis of serial acquisition scheme using AWGN rejection filter in Rayleigh fading channel and propose robust acquisition scheme using Reference filter, which is utilized to vary threshold at fading rate, in Rayleigh fading channel. AWGN rejection filter is utilized to evaluate running average for compensating channel gain. The JAKE model, which a channel model, is used for the analysis. The simulation result shows that the mean acquisition time of the proposed acquisition system is minimized than acquisition system using AWGN rejection filter and serial-search acquisition system.

• Journal of Communications and Networks
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• v.18 no.1
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• pp.8-18
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• 2016

In this paper, we propose a number of blind equalization approaches for time-varying andmulti-path channels. The approaches employ cost reference particle filter (CRPF) as the symbol estimator, and additionally employ either least mean squares algorithm, recursive least squares algorithm, or $H{\infty}$ filter (HF) as a channel estimator such that they are jointly employed for the strategy of "Rao-Blackwellization," or equally called "mixture filtering." The novel feature of the proposed approaches is that the blind equalization is performed based on direct channel estimation with unknown noise statistics of the received signals and channel state system while the channel is not directly estimated in the conventional method, and the noise information if known in similar Kalman mixture filtering approach. Simulation results show that the proposed approaches estimate the transmitted symbols and time-varying channel very effectively, and outperform the previously proposed approach which requires the noise information in its application.

• Journal of electromagnetic engineering and science
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• v.3 no.2
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• pp.140-146
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• 2003

In this paper, the design of a channel filter and its group-delay-and-amplitude equalizer is carried out for the Ka-band satellite transponder subsystem. The 8th order dual-mode filter is employed for high selectivity around the band-edges with an elliptic-integral function response and has an in-line configuration. The 2-pole, reflection-type, group-delay equalizer is designed and manufactured to reduce the group-delay and amplitude variation, which can be large for such a high order filter. It is noted that in both the filter and equalizer, adopting the dual-mode coupling mechanism leads to less mass and volume. Through measurement, the performance of the realized group-delay-equalized filter is shown to meet the equipment requirements and to be appropriate for the satellite input multiplexer.

• 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.

• Journal of Communications and Networks
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• v.9 no.3
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• pp.219-228
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• 2007

In this paper, we present a novel blind adaptive multiuser detector structure and three robust recursive filters to improve the performance in CDMA environments: Sigma point kalman filter (SPKF), particle filter (PF), and Gaussian mixture sigma point particle filter (GMSPPF). Our proposed robust recursive filters have superior performance over a conventional extended Kalman filter (EKF). The proposed multiuser detector algorithms initially use Kalman prediction form to estimated channel parameters, and unknown data symbol be predicted. Second, based on this predicted data symbol, the robust recursive filters (e.g., GMSPPF) is a refined estimation of joint multipaths and time delays. With these estimated multipaths and time delays, data symbol detection is carried out (Kalman correction form). Computer simulations show that the proposed algorithms outperform the conventional blind multiuser detector with the EKF. Also we can see it provides a more viable means for tracking time-varying amplitudes and time delays in CDMA communication systems, compared to that of the EKF for near-far ratio of 20 dB. For this reason, it is believed that the proposed channel estimators can replace well-known filter such as the EKF.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.57 no.3
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• pp.260-264
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• 2008

In this paper, a dual mode baseband analog channel selection filter is described which is designed for the Bluetooth and WCDMA wireless communications. Using the presented current-mode integrator, a dual mode channel selection filter is designed. To verify the current-mode integrator circuit, Hspice simulation using 1.8V Hynix $0.18{\mu}m$ standard CMOS technology was performed and achieved $50.0{\sim}4.3dB$ gain, $2.29{\sim}10.3MHz$ unity gain frequency. The described third-order dual mode analog channel selection filter is composed of the current-mode integrator, and used SFG(Signal Flow Graph) method. The simulated results show 0.51, 2.40MHz cutoff frequency which is suitable for the Bluetooth and WCDMA baseband block each.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.5
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• pp.1172-1177
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• 2008

To compensate voltage error of the channel selecting filter, a current comparison type voltage tuning circuit is designed. Because the proposed current comparison type voltage tuning circuit is not need to attach another subcircuit, the chip size can be reduced, therefore the proposed circuit is very useful in the low voltage and low power channel filter. We used three channels including bluetooth communication system as application circuits of the proposed tuning circuit. As the results of HSPICE simulation using $0.18{\mu}m$ CMOS technology verify that the proposed tuning circuit respectively can be operated in $12{\mu}s$, $13{\mu}s$ and $15{\mu}s$ in three channel.