• The Journal of the Acoustical Society of Korea
• /
• v.40 no.5
• /
• pp.460-465
• /
• 2021

In active sonar, time delay estimation is used to find the distance between the target and the sonar. Among the time delay estimation methods for active sonar, estimation in the frequency domain is widely used. When estimating in the frequency domain, the time delay can be thought of as a frequency estimator, so it can be used relatively easily. However, this method is prone to rapid increase in error due to noise. In this paper, we propose a new method which applies weighted dictionary and sparsity in order to reduce this error increase and we extend it to two receivers to propose an algorithm for estimating the time delay between two receivers. And the case of applying the proposed method and the case of not applying the proposed method including the conventional frequency domain algorithm and Generalized Cross Correlation-Phase transform (GCC-PHAT) in a white noise environment were compared with one another. And we show that the newly proposed method has a performance gain of about 15 dB to about 60 dB compared to other algorithms.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.9 no.4
• /
• pp.542-551
• /
• 1998

In this study, a method of error compensation for channel gain imbalance, phase imbalance and local oscillator leakage in the modulator of frequency direct conversion is suggested. The compensation of channel imbalance can be carried out by using the received power after transmitting test signal. By applying this method, the phase imbalance conversion with frequency can be easily compensated since this method is rarely affected by the transmission channel. It is confirmed that the algorithm proposed in this study(iteration coefficient=11) converges faster than conventional algorithm(iteration coefficient=43). From the numerical results, the DC-offset, channel gain, phase imbalance compensation coefficient and iteration number converges into($f_1$=0.0199999, $f_2$=-0.050001, $C_{22}$=0.9133, $C_{12}$=-0.0524, N=13) when the local oscillator leakage is not considered. However, it converges into($f_1$=-0.02, $f_2$=-2.2476, $C_{22}$=0.9133, $C_{12}$=-0.0524, N=16) when the local oscillator leakage is considered.

• Journal of Power Electronics
• /
• v.18 no.1
• /
• pp.234-250
• /
• 2018

This paper presents a new load sharing control for use between paralleled three-phase inverters in an islanded microgrid based on the online line impedance estimation by the use of a Kalman filter. In this study, the mismatch of power sharing when the line impedance changes due to temperature, frequency, significant differences in line parameters and the requirements of the Plug-and-Play mode for inverters connected to a microgrid has been solved. In addition, this paper also presents a new droop control method working with the line impedance that is different from the traditional droop algorithm when the line impedance is assumed to be pure resistance or pure inductance. In this paper, the line impedance estimation for parallel inverters uses the minimum square method combined with a Kalman filter. In addition, the secondary control loops are designed to restore the voltage amplitude and frequency of a microgrid by using a combined nominal value SOGI-PLL with a generalized integral block and phase lock loop to monitor the exact voltage magnitude and frequency phase at the PCC. A control model has been simulated in Matlab/Simulink with three voltage source inverters connected in parallel for different ratios of power sharing. The simulation results demonstrate the accuracy of the proposed control method.

• The KIPS Transactions:PartB
• /
• v.17B no.1
• /
• pp.69-78
• /
• 2010

Spectral modeling synthesis (SMS) has been used as a powerful tool for musical sound modeling. This technique considers a sound as a combination of a deterministic plus a stochastic component. The deterministic component is represented by the series of sinusoids that are described by amplitude, frequency, and phase functions and the stochastic component is represented by a series of magnitude spectrum envelopes that functions as a time varying filter excited by white noise. These representations make it possible for a synthesized sound to attain all the perceptual characteristics of the original sound. However, sometimes considerable phase variations occur in the deterministic component by using the conventional SMS for the complex sound such as whale sounds when the partial frequencies in successive frames differ. This is because it utilizes the calculated phase to synthesize deterministic component of the sound. As a result, it does not provide a good spectrum matching between original and synthesized spectrum in higher frequency region. To overcome this problem, we propose a modified SMS that provides good spectrum matching of original and synthesized sound by calculating complex residual spectrum in frequency domain and utilizing original phase information to synthesize the deterministic component of the sound. Analysis and simulation results for synthesizing whale sounds suggest that the proposed method is comparable to the conventional SMS in both time and frequency domain. However, the proposed method outperforms the SMS in better spectrum matching.

• Proceedings of the KIEE Conference
• /
• 1998.11a
• /
• pp.286-288
• /
• 1998

A digital distance relaying algorithm immune to the effect of the fault resistance in single-phase to ground fault is proposed. The power frequency components of relaying signals are extracted by the 1 cycle DFT. To compensate the magnitude and phase of the estimated impedance, which is calculated by the general method, this algorithm uses phase difference between the zero sequence impedance of networks on both sided of the system.

• Transactions of the Korean Society of Automotive Engineers
• /
• v.22 no.6
• /
• pp.23-30
• /
• 2014

This paper proposes a dead time compensation method for an AC motor drive using phase current polarity information which is detected based on a digital programmable low-pass filter (PLPF). The polarity detection using the PLPF is an alternative solution of a conventional method which uses a general low-pass filter (LPF) and hysteresis bands in order to avoid jittering due to noises. The PLPF not only adjusts its cutoff frequency according to the synchronous frequency of AC motors but also eliminates a gain attenuation and phase delay which are main problems of the general LPF. Through the PLPF, a fundamental component signal without gain and phase distortions is extracted from the measured raw current signal with noise. By use of the fundamental component, the polarity of current is effectively detected by reducing the hysteresis band. Finally, the proposed method compensates the dead time effects by adding or subtracting average voltage value to voltage references of the controller according to the detected current polarity information. The proposed compensation method is experimentally verified by compared with the conventional method.

• Journal of the Computational Structural Engineering Institute of Korea
• /
• v.30 no.1
• /
• pp.17-22
• /
• 2017

In this study, we designed the metallic reception part of a patch antenna using the phase field design method. The design object function is formulated with the S-parameter value which represent the return loss so that it is targeted to maximize radiation efficiency at a target frequency. The initial model of a patch antenna was designed via the ordinary theory based approach and its performance was enhanced by changing the structural configuration of the metallic part using the phase field design method combined with the double well potential functions. The final shape was proposed by removing the gray scale area along the structural boundary by employing a cut-off method. The proposed shape shows that the radiation efficiency at target frequency is significantly improved compared with the initial patch shape. The finite element analysis and optimization precess was performed using the commercial package COMSOL and Matlab programming.

• Journal of Broadcast Engineering
• /
• v.20 no.3
• /
• pp.408-413
• /
• 2015

Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. The specific method is to make the original sound accompaniment to remove only the voice of the singer in the singer music album. In this paper, a system to separate vocal components from music accompaniment for stereo recordings were proposed. Proposed system consists of two stages. The first stage is a vocal detection. This stage classifies an input into vocal and non vocal portions by using SVM with MFCC. In the second stage, selective frequency subtractions were performed at each frequency bin in vocal portions. In this case, it is determined in consideration not only the energies for each frequency bin but also the phase of the each frequency bin at each channel signal. Listening test with removed vocal music from proposed system show relatively high satisfactory level.

• The Transactions of the Korean Institute of Power Electronics
• /
• v.5 no.2
• /
• pp.154-162
• /
• 2000

A technique to suppress the low frequency ripple voltage of the DC output ${\gamma}$oltage in three-phase buck d diode rectifiers is presented in this paper. The proposed pulse frequency modulation method is employed to r regulate the output voltage of the buck diode rectifiers and guarantee zero-current switching of the switch over the Vvide load range. The pulse frequency control method used in tIns paper shows generally good p performance such as low THD of the input line current and unity power factor. In addition, the pulse f freιluency method can be effectively used to suppress the low frequency voltage ripple appeared in the dc output voltage. The proposed technique illustrates its validity and effectiveness through the respective s simulations and experiments.

• ETRI Journal
• /
• v.32 no.6
• /
• pp.911-920
• /
• 2010

This paper presents a method of designing hybrid analog/asymmetrical square-root (SR) FIR filters. In addition to the conventional frequency domain constraints, the proposed method considers time-domain constraints as well, including the inter-symbol interference (ISI) and the opening of the eye pattern at the receiver output. This paper also reviews a systematic way to find the discrete-time equivalence of analog parts in a band-limited digital communication system. Thus, a phase equalizer can be easily realized to compensate for the nonlinear phase responses of the analog components. With the hybrid analog/SR FIR filter co-design, examples show that using the proposed method can result in a more robust ISI performance in the presence of the receiver clock jitter.