• Journal of the Korea Institute of Military Science and Technology
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• v.1 no.1
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• pp.201-210
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• 1998

Inertial Navigation System(INS) can provide the vehicle position and velocity information using inertial sensor outputs without the use of external aids. Unfortunately INS navigation error increases with time due to inertial sensor errors, and therefore it is desirable to combine INS with external aids such as GPS, TACAN, OMEGA, and etc.. In this paper we propose an integration algorithm of commercial GPS/GLONASS and INS where an adaptive filter for signal processing of GPS/GLONASS receiver and the 12th order Kalman filter for aided strapdown INS(SDINS) we employed. Simulation results show that the proposed adaptive filter can effectively remove a randomly occurring abrupt jump due to sudden corruption of the received satellite signal and that the Kalman filter performs satisfactorily.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.8A
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• pp.785-793
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• 2007

This paper presents the digital signal processing submodule of a 100Base-TX Ethernet receiver to support 100Mbps at TP cable channel. The proposed submodule consists of programmable gain controller, timing recovery, adaptive equalizer and baseline wander compensator. The measured Bit Error Rate is less than $10^{-12}BER$ when continuously receiving data up to 150m. The proposed signal processing submodule is implemented in digital circuits except for PLL and amplifier. The performance improvement of the proposed equalizer and BLW compensator is measured about 1dB compared with the existing architecture that removes BLW using errors of an adaptive equalizer. The architecture has been modeled using Verilog-HDL and synthesized using samsung $0.18{\mu}m$ cell library. The implemented digital signal processing submodule operates at 142.7 MHz and the total number of gates are about 128,528.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.1
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• pp.20-27
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• 2010

In this paper, we presented a signal model of STAP and actual environment of clutter. The novel estimation method of clutter covariance matrix using median value is proposed to overcome serious performance degradation after NHD in nonhomogeneous clutter. Eigen value characteristic is improved through diagonal loading. Target detection ability and SINR loss of the proposed method though MSMI statistic is also compared with conventional method using average value. The simulation results, confirm the proposed method has better performance than others.

• Speech Sciences
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• v.9 no.4
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• pp.275-287
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• 2002

In man-machine communication by speech under the noisy environment, the quality of speech may be degraded severely for the machine to recognize correctly. Especially when the corrupting noise occupies the same band as the speech, the conventional fixed filters cannot filter out the noise effectively. In recent, to resolve such a problem adaptive noise canceller (ANC) is frequently used, which is based upon adaptive filters. The Adaptive recursive filters perform better than adaptive nonrecursive filters due to the added poles, but the stability may be severely threatened. In this paper an ANC system employing the adaptive recursive filter is proposed to enhance the speech corrupted by noise. And the stability of the adaptive recursive filter is guaranteed by employing the adaptive compensator.

• Journal of IKEEE
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• v.21 no.3
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• pp.202-210
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• 2017

Recently, some variable length adaptive filters which employ variable lengths taps for the input signal statistics are proposed [1-5]. In this paper, a new subband adaptive filter with variable filter tap length is proposed. The proposed subband variable length adaptive filters can optimize filter length for each subband which can result less computational complexities with respect to the conventional full band adaptive filters. When the signal in the full band has narrow spectrum, the conventional full band adaptive requires very long filter taps, whereas the proposed subband variable filter requires less taps with the spectrum split in subband. The computer simulation results reveals that in many case, in system identification with narrow band system estimation, the proposed adaptive filter has less computational complexities with faster convergence.

• Journal of the Institute of Convergence Signal Processing
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• v.8 no.2
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• pp.130-135
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• 2007

This paper deals with a controller design for a 2 dimensional aeroelatic model which has unknown parameters including polynomial type nonlinearity. Actually in case of state and acuator signal having magnitude, rate and bandwidth limitations, the controller can't be implemented and so in each case, a filter is used for implementation. First, error signals are defined upon the backstepping theory, and tracking error signals are also defined due to command signal and filter signals and then compensated tracking error signals are defined. Lastly, a Lyapunov function is defined for the stabilization and from this method, an adaptive law is derived. Simulations are done for the demonstrtion of the effectiveness of the algorithms.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2005.11a
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• pp.45-48
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, We know the proposed algorithm is prefer than the existent algorithm.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.9 no.4
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• pp.294-300
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• 2009

The neural network with radial basis function is introduced for position tracking control of AC servo drive with the existence of system uncertainties. An adaptive robust term is applied to overcome the external disturbances. The proposed controller is implemented on a high performance digital signal processing DSP TMS320C6713-300. The stability and the convergence of the system are proved by Lyapunov theory. The validity and robustness of the controller are verified through simulation and experimental results

• Journal of Korea Society of Digital Industry and Information Management
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• v.5 no.3
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• pp.99-105
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• 2009

This paper suggests novel signal processing methods for optimal beamforming of smart antenna system in CDMA2000 mobile communication environments. This method utilize characteristics of the reverse pilot channel of CDMA2000 mobile communication systems, and applies them to improve the performance of an adaptive algorithm, which is used to a smart antenna system for beamforming. To perform the best beamforming, it is important to get an exact beamforming algorithm. This paper proposed an algorithm based on Laglange multiplier which has such an adaptive process, and also proposed the method to demodulate the received signal through array antenna using pilot channel in CDMA2000 environment. This paper analysed the enhanced performance of proposed algorithm in various signal environment through signal modeling of physical layer in CDMA2000 reverse link.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.46-49
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• 2003

The use of confidence measures for word/utterance verification has become art essential component of any speech input application. Confidence measures have applications to a number of problems such as rejection of incorrect hypotheses, speaker adaptation, or adaptive modification of the hypothesis score during search in continuous speech recognition. In this paper, we present a new utterance verification method using vowel string. Using subword HMMs of VCCV unit, we create anti-models which include vowel string in hypothesis words. The experiment results show that the utterance verification rate of the proposed method is about 79.5%.