• The Journal of the Acoustical Society of Korea
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• v.37 no.1
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• pp.66-72
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• 2018

In this paper, we propose an adaptive beamforming algorithm of triplet arrays for active sonar systems. The proposed algorithm consists of three steps: matched filters, cardioid beamforming, and line array beamforming. First, we apply a matched filter of a transmitted pulse to received individual sensor signals and obtain filterd signals. Then, we perform the fast Fourier transform to the matched filter results, and make a cardioid beam for each triplet data, respectively. Finally, we apply an adaptive beamforming by assuming that the cardioid beams are input signals of a line array. Experimental results demonstrate that the proposed algorithm provides better performances than conventional algorithms.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.12
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• pp.5647-5654
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• 2012

In this paper, we developed an electro impedance spectroscopy (EIS) device, which are primarily used for the analysis of fuel cells or batteries, to widen its coverage to the next generation super capacitor EDLC characterization. The developed system was composed of a signal generator that can generate various signal patterns, a potentiostatic generator, and a high speed digital filter for signal processing and measurement program. The developed system is portable, which is not only suitable laboratory use but also for mass production line. The special features of the system include a patterned output signal from 0.01 to 20 kHz, and a fast Fourier transform (FFT) analysis of current signals, both of which are acquired simultaneously. Our tests showed similar results after comparing the analysis from our newly-developed device showing the characteristics of EDLC complex impedance and the analysis from an equivalent impedance which was applied to an equivalent circuit. Now, we can expect a fast inspection time from the application of the present system to the super capacitor production line, based on time-varying changes in electrochemical impedance.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.4
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• pp.166-172
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• 2012

The standard wave height measurement system is usually based on spectrum analysis for measuring wave height. The spectrum analysis is complicated because of the FFT, and the FFT is not for real time processing since it requires the saved data segments. In this paper, we carried out the performance evaluation of real-time and simpler wave height measurement system using the kalman filter and inertial sensors. The kalman filter theory is complicated, but its algorithm is simpler than the FFT and the kalman filter is used to estimate wave height by integrating acceleration data. But the accumulated error is occurred when the acceleration data is integrated. We developed the algorithm using the wind wave characteristic to decrease the accumulated error. In this paper, the performance evaluation of the wave height measurement system is carried out for various wind wave conditions. Through the experiments, we verified that it shows high measurement performance with the 3.5% margin of error in wind wave condition.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.4
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• pp.214-219
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• 2009

This paper proposes a watermarking algorithm considering the frequency characteristics of Haegeum sounds for copyright protection of digital Haegeum sound contents. The harmonics of Haegeum sounds commonly have large magnitude values in 1500Hz~2000Hz and 2800Hz~3500Hz so that those bands are selected to embed a watermark. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and embeds the watermark bits generated by PN (pseudo noise) sequence into the harmonics in the selected bands. Furthermore, the proposed method is robust to lowpass filter, bandpass filter, cropping, noise addition, MP3 compression attacks and the maximum BER (bit error rate) is 1.41% after lowpass filter attack. To measure the quality of the watermarked sound, subjective listening test, MUSHRA (multiple stimuli with hidden reference and anchor), was conducted. The mean value of MUSHRA listening test is bigger than 98 and 96.67 for every Haegeum sounds and Korean classical music with Haeguem, respectively.

• Journal of the Semiconductor & Display Technology
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• v.22 no.3
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• pp.28-35
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• 2023

Moiré patterns emerge due to the interference between the non-emission area of the LED screen and the grid line in an image sensor of a video recording device when taking a video in the presence of the LED screen. To reduce the moiré intensity, we have fabricated an anti-moiré filter using hollow glass microspheres (HGMs) by slot-die coating. The LED screen has a large non-emission area because of a large pitch (distance between LED chips), causing more severe moiré phenomenon, compared with a display panel having a very narrow black matrix (BM). It is shown that HGMs diffuse light in such a way that the periodicity of the screen is broken and thus the moiré intensity weakens. To quantitatively analyze its moiré suppression capability, we have calculated the spatial frequencies of the moiré fringes using fast Fourier transform. It is addressed that the moiré phenomenon is suppressed and thus the amplitude of each discrete spatial frequency term is reduced as the HGM concentration is increased. Using the filter with the HGM concentration of 9 wt%, the moiré fringes appeared depending sensitively on the distance between the LED screen and the camera are almost completely removed and the visibility of a nature image is enhanced at a sacrifice of luminance.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.8 s.101
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• pp.954-961
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• 2005

This paper is concerned with the implementation of auto-tuning positive position feedback controller using a digital signal processor and microcontroller. The main advantage of the positive position feedback controller is that it can control a natural mode of interest by tuning the filter frequency of the positive position feedback controller to the natural frequency of the target mode. However, the positive position feedback controller loses its advantage when mistuned. In this paper, the fast fourier transform algorithm is implemented on the microcontroller whereas the positive position feedback controller is implemented on the digital signal processor. After calculating the frequency which affects the vibrations of structure most, the result is transferred to the digital signal processor. The digital signal processor updates the information on the frequency to be controlled so that it can cope with both internal and external changes. The proposed scheme was installed and tested using a beam equipped with piezoceramic sensor and actuator. The experimental results show that the auto-tuning positive position feedback controller proposed in this paper can suppress vibrations even when the target structure undergoes structural change thus validating the approach.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.498-503
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• 2005

This paper is concerned with the implementation of adaptive positive position feedback controller using a digital signal processor and microcontroller The main advantage of the positive position feedback controller is that it can control a natural mode of interest by tuning the filter frequency of the positive position feedback controller to the natural frequency of the target mode. However, the positive position feedback controller loses its advantage when mistuned. In this paper, the fast fourier transform algorithm is implemented on the microcontroller whereas the positive position feedback controller is implemented on the digital signal processor. After calculating the frequency which affects the vibrations of structure most the result is transferred to the digital signal processor. The digital signal processor updates the information on the frequency to be controlled so that it can cope with both internal and external changes. The proposed scheme was installed and tested using a beam equipped with piezoceramic sensor and actuator. The experimental results show that the adaptive positive position feedback controller proposed in this paper can suppress vibrations even when the target structure undergoes structural change thus validating the approach.

• Journal of Electrical Engineering and Technology
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• v.8 no.6
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• pp.1380-1388
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• 2013

The conventional graphical representation of the instantaneous compensation power flow for single-phase active power filters(APFs) simply represents the active power flow and the reactive power flow which flowing between the power source and the active filter / the load. But, this method does not provide the information about the rectification mode and the compensation mode of APFs, especially, the loss for each mode was not considered at all. This is very important to understand the compensation operation characteristics of APFs. Therefore, this paper proposes the graphical representation of the instantaneous compensation power flow for single-phase APFs considering the instantaneous rectification mode and the instantaneous inversion mode. Three cases are verified in this paper - without compensation, with compensation of the active power 'p' and the fundamental reactive power 'q', and with compensation of only the distorted power 'h'. To ensure the validity of the proposed approach, PSIM simulation is achieved. As a result, we could confirm that the proposed approach was easy to explain the instantaneous compensation power flow considering the instantaneous rectification mode and the instantaneous inversion mode of APFs, also, Total Harmonic Distortion(THD)/Power Factor (P.F) and Fast Fourier Transform(FFT) analysis were compared for each case.

• Speech Sciences
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• v.10 no.2
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• pp.99-109
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• 2003

The purpose of this study was to analyze and compare head register with chest register of singers acoustically. Fifteen healthy tenor major students were participated. Fifteen healthy untrained adults were chosen as the control group for this study. Long term average (LTA) power spectrum using the Fast Fourier transform (FFT) algorithm and Linear predictive coding (LPC) filter response were made with /a/ sustained in both head (G4, 392 Hz) and chest registers (C3, 131 Hz). Statistical analysis was performed using the Mann-Whitney test. In the LTA power spectrum, head register of singers increased in the level of energy gain within the frequency of 2.2-3.4 kHz (p<0.01), and 7.5-8.4 kHz (p<0.01, p<0.05). Chest register of singers increased in the frequency of 2.2-3.1 kHz (p<0.01), 7.8-8.4 kHz (p<0.05) and around 9.6 kHz (p<0.01). The LTA power spectrum revealed a peak of acoustic energy around 2,500 Hz, known as the singer's formant and another peak of acoustic energy around 8,000 Hz in the singer's voice.

• Transactions on Electrical and Electronic Materials
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• v.9 no.1
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• pp.38-43
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• 2008

Using an arterial pressure-volume (APV) model, we performed an analysis of the conventional blood pressure estimation method using an oscillometric sphygmomanometer with computer simulation. Traditionally, the maximum amplitude algorithm (MAA) has been applied to the oscillation waveforms of the APV model to obtain the mean arterial pressure and the characteristic ratio. The estimation of mean arterial pressure and characteristic ratio was significantly affected by the shape of the blood pressure waveforms and the cutoff frequency of high-pass filter (HPF) circuitry. Experimental errors result from these effects when estimating blood pressure. To determine an algorithm independent of the influence of waveform shapes and parameters of HPF, the volume oscillation of the APV model and the phase shift of the oscillation with fast Fourier transform (FFT) were tested while increasing the cuff pressure from 1 mmHg to 200 mmHg (1 mmHg/s). The phase shift between ranges of volume oscillation was then only observed between the systolic and the diastolic blood pressures. The same results were obtained from simulations performed on two different arterial blood pressure waveforms and one hyperthermia waveform.