• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.296-299
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• 2003

In this paper, an underwater acoustic digital transceiver is designed and implemented by a multiple DSPs system. We have designed a QPSK transmitter based on look-up table and 13-symbols Barker code is used for frame synchronization. Channel distortions are compensated by a wide-band beamformer based on FIR filter and an adaptive equalize. with RLS algorithm. Uniform linear array (ULA) with four elements is used for the spartial signal processing. 1/2 convolutional code and Viterbi decoder are implemented to overcome time-varying multi-path fading. Also, we show experimental results in the underwater anechoic basin at KRISO/KORDl and Goseong, Donghae and Soyang lake of Kangwon-do.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.817-818
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• 2006

In Magnetic Resonance Imaging(MRI), the QRS complex of ECG is used as a trigger signal for MRI scan. But, gradient and RF(radio frequency) artifacts which are caused to static and dynamic field in MRI scanner cause interference in the ECG. Also, the signal shape of theses artifacts can be similar to the QRS-complex, causing possible misinterpretation during patient monitoring and false gating of the MRI. In case of using general FIR or IIR band-pass filters for minimizing the artifacts, artifact-reduction-ratio is not excellent. So, an adaptive real-time digital filter is proposed for reduction of noise by gradient and RF(radio frequency) artifacts. The proposed filter for MRI-Gating is based on the noise-canceller with NLMS(Normalized Least Mean Square) algorithm. The reference signals of the adaptive noise canceller are a combination of the noisy three channel ECG signals. In conclusions, the proposed method showed the acceptable quality of ECG signal with sufficient SNR for gating the MRI and possibility of real time implementation.

• Journal of the Society of Naval Architects of Korea
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• v.39 no.3
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• pp.57-63
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• 2002

This paper describes a control algorithm for the buffer of a deep-sea mining system, in which the buffer is connected to a long slender pipe and then to a surface ship on one end, and to a collector on sea floor through a flexible hose on the other end. A mathematical modeling is established for designing the controller for buffer thrusters, in which the dynamic response of the long pipe is taken into account based on the mode superposition method. The fluid loading acting on the pipe is estimated by using Morison's formula. For simplicity, the surface ship is assumed to be kept stationary, the reaction from the flexible hose is ignored and only the lateral motions are considered. In order to guide the buffer to react only to the low-frequency motion of the surface vessel, the FIR digital filter is introduced to a PID-based controller It can be shown numerically that the high frequency component of the ship's motion can be effectively filtered out by using the FIR low pass filter.

• Journal of Korea Society of Digital Industry and Information Management
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• v.8 no.2
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• pp.49-58
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• 2012

This paper is on an improved orthogonal projection method which can cancel the acoustic feedback signals in the digital hearing aids. Comparing with the NLMS algorithm which is widely used for simplicity and stability, it shows that this method has the improvement of the convergence performances, and has small computational quantities, for signals with the large auto-correlation as speech signals. This uses the improved orthogonal projection algorithm which reduces the correlation of signals. To verify the convergence characteristics of the proposed algorithm, we simulated about various input signals. The acoustic feedback canceller has a 12-bit resolution with 64-tap adaptive FIR filter. And we compared the results of simulation for this algorithm with the ones for the NLMS algorithm. By these works, it is proved that the feedback canceller adopting the proposed algorithm shows about 3.5dB more high SNR than the NLMS algorithm in the colored input signals.

• Progress in Superconductivity
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• v.4 no.1
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• pp.53-58
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• 2002

Noise from electronic instrumentation is invariably present in biomedical signals, although the art of instrumentation design is such that this noise source may be negligible. And sometimes signals of interest are contaminated or degraded by signals of similar type from another source. Biomedical signals are omni-presently contaminated by these background noises that span nearly all frequency bandwidths. In the magneto-cardiogram (MCG), several digital filters have been designed for the elimination of the power-line interference, broadband white noise, surrounding magnetic noise, and baseline wondering. In addition to the introduced FIR filter, notch, adaptive filter using the least mean square (LMS) algorithm, and recurrent neural network (RNN) filter, a new filtering method for effective noise canceling in MCG signals is proposed in this paper, which is realized by the wavelet packets. The experimental results show that the proposed filter using wavelet packet performs efficiently with respect to noise rejection. To verify this, two characteristics were analyzed and compared with LMS adaptive filter, SNR of filtered signal and attractor pattern using the nonlinear dynamics.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.4
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• pp.591-598
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• 2021

The digital phase-locked loops(DPLL) is a circuit used for phase synchronization and has been generally used in various fields such as communication and circuit fields. State estimators are used to design digital phase-locked loops, and infinite impulse response state estimators such as the well-known Kalman filter have been used. In general, the performance of the infinite impulse response state estimator-based digital phase-locked loop is excellent, but a sudden performance degradation may occur in unexpected situations such as inaccuracy of initial value, model error, and disturbance. In this paper, we propose a minimum variance finite impulse response filter with optimal horizon for designing a new digital phase-locked loop. A numerical method is introduced to obtain the measured value interval length, which is an important parameter of the proposed finite impulse response filter, and to obtain a gain, the covariance matrix of the error is set as a cost function, and a linear matrix inequality is used to minimize it. In order to verify the superiority and robustness of the proposed digital phase-locked loop, a simulation was performed for comparison and analysis with the existing method in a situation where noise information was inaccurate.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.36 no.5
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• pp.367-376
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• 1987

In this paper we have disigned and implemented a real time ALE (adaptive line enhancer) using a high speed digital processor,NEC 7720. For the ALE system, we have employed an adaptive LMS(least mean square) algorithm proposed by Widrow and Hoff and a 32-order FIR(finite impulse response) filter. Extensive computer simulations have been performed to investigate the peformance of the ALE and to determine necessary parameters for hardware design. The developed software for an NEC 7720 was tested in real time operation using an NEC7720 hardware emulator. The ALE has been tested by sinusoidal waves and real CW (continuous wave) signals. It was found that the experimental results were well agreed with the computer simulation results. Thus it can be concluded that the ALE is useful for detection and enhancement of a sinusoidal signal which is corrupted by an additive Gaussian noise.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.10
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• pp.188-198
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• 2013

Maximally flat (MAXFLAT) half-band filters usually have wider transition band than other filters. This is due to the fact that the maximum possible number of zeros at $z={\pm}1$ is imposed, which leaves no degree of freedom, and thus no independent parameters for direct control of the frequency response. This paper describes a novel method for the design of FIR halfband filters with an explicit control of the transition-band width. The proposed method is based on a generalized Lagrange halfband polynomial (g-LHBP) with coefficients parametizing a 0-th coefficient $h_0$, and allows the frequency response of this filter type to be controllable by adjusting $h_0$. Then, $h_0$ is modeled as a steepness parameter of the transition band and this is accomplished through theoretically analyzing a polynomial recurrence relation of the g-LHBP. This method also provides explicit formulas for direct computation of design parameters related to choosing a desired filter characteristic (by trade-off between the transition-band sharpness and passband & stopband flatness). The examples are shown to provide a complete and accurate solution for the design of such filters with relatively sharper transition-band steepness than MAXFLAT half-band filters.

• ETRI Journal
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• v.32 no.1
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• pp.33-43
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• 2010

The increasing demand for non-sinusoidal currents affects the quality of distribution networks. Harmonic detection is a crucial step in the cancellation of those components by active power filters. In this paper, the discrete cosine transform (DCT) is compared with different implementations based on Fourier transforms, demonstrating their equivalences and the advantages provided by the former. We demonstrate that the phase error in the presence of grid frequency deviations and the transient length are reduced by half in comparison to the discrete Fourier transform. A novel algorithm is developed to provide frequency adaptation to the DCT, taking advantage of its good features. The window width is adjusted in real time according to the actual value of the grid fundamental frequency by means of a phase-locked loop. A technique based on dithering is employed to overcome the limitation caused by the truncation of the window number of samples, so the frequency resolution is enhanced. The theoretical approach is verified by simulated and experimental results.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2004.10a
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• pp.1406-1409
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• 2004

The Resolver usually used in industry is the absolute angle analog sensor that must be in order to driving BLDC (brushless DC) motor, and it needs RDC(Resolver-to-Digital converter) for changing the output signal to digital to be applied to the SVPWM(Space Vector Pulse Width Modulation) algorithm. Commonly used S/W RDC needs trigonometric function. What it takes a lot of calculation time of processor is gotten at weak point. In this paper, S/W RDC is realized except trigonometric functions as a result of feedback resolver outputs after filtering using FIR filter. thus, processing time is reduced. So, One-chip DSP Controller operating the Vector Control, RDC, and SVPWM can be designed.