• The Journal of the Acoustical Society of Korea
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• v.21 no.3E
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• pp.119-125
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• 2002

In the conventional generalized side-lobe canceller (GSC), the output is an estimated error signal that causes the adaptive filter weights do not converge to the optimal value. This paper presents a new side-lobe canceller with adaptive compensator th it reduces the misadjustment of the adaptive filter coefficients for the structural problem in the GSC. The adaptive compensator separates the output signal from the estimated error. The newly estimated error signal converges to the zero while the output signal tracks the target signal. This paper shows improvement of the performance by comparing the computer simulation of the output signal of GSC with the output signal of the proposed algorithm.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.56.3-56
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• 2001

This study describes the jump error smoothing scheme with fuzzy logic based on the scalar adaptive filter. The scalar adaptive filter is an useful algorithm for smoothing abrupt jump errors. However, the performances of scalar adaptive algorithm depend on the variance of real signal. So to design an effective algorithm, many informations of real and jump signal are required. In this paper, the fuzzy rules are designed by the analysis of scalar adaptive filter, and then the improved and simplified scheme is developed for smoothing the jump error. Simulations to INS/GPS integrated system show that the proposed method is effective.

• 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.

• 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 Biomedical Engineering Research
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• v.22 no.3
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• pp.285-291
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• 2001

In this paper, we designed two preprocessing adaptive filter - high pass filter and notch filter - using FPGA. For minimizing the calculation load of multi-channel and high-resolution ECG system, we utilize FPGA rather than digital signal processing chip. To implement the designed filters in FPGA, we utilize FPGA design tool(Altera corporation, MAX-PLUS II) and CSE database as test data. In order to evaluate the performance in terms of processing time, we compared the designed filters with the digital filters implemented by ADSP21061(Analog Devices). As a result, the filters implemented by FPGA showed better performance than the filters based on ADSP21061. As a consequence of examination, we conclude that FPGA is a useful solution in multi-channel and high-resolution signal processing.

• 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.

• 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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.

• The Transactions of the Korean Institute of Power Electronics
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• v.9 no.6
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• pp.592-601
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• 2004

The performance of active power filter depends on the inverter characteristic, the control method, and the accuracy of reference signal generator. The accuracy of reference signal generator is the most critical item to determine the performance of active power filter. This paper introduces a novel reference signal generator composed of improved adaptive predictive filter. The performance of proposed reference signal generator was verified by means of simulation with MATLAB. The application feasibility was evaluated by building and experimenting a single-phase active power filter based on the proposed reference generator, which was implemented in the DSP(digital signal processor) TMS320C31. Both simulation and experimental results confirm that the proposed reference signal generator can be utilized for the active power filter.

• Journal of Information Processing Systems
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• v.15 no.4
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• pp.748-764
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• 2019

In order to deal with the filtering delay problem of least mean square adaptive filter noise reduction algorithm and music noise problem of spectral subtraction algorithm during the speech signal processing, we combine these two algorithms and propose one novel noise reduction method, showing a strong performance on par or even better than state of the art methods. We first use the least mean square algorithm to reduce the average intensity of noise, and then add spectral subtraction algorithm to reduce remaining noise again. Experiments prove that using the spectral subtraction again after the least mean square adaptive filter algorithm overcomes shortcomings which come from the former two algorithms. Also the novel method increases the signal-to-noise ratio of original speech data and improves the final noise reduction performance.