• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.166-174
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• 1998

Neural networks, having highly nonlinear dynamics by virtue of the distributed nonlinearities and the learing ability, have the potential for the adaptive prediction of nonstationary signals. This paper describes the nonlinear prediction of these signals in two ways; using a nonlinear module and the cascade combination of nonlinear and linear modules. Fully-connected recurrent neural networks (RNNs) and a conventional tapped-delay-line (TDL) filter are used as the nonlinear and linear modules respectively. The dynamic behavior of the proposed predictors is demonstrated for chaotic time series adn speech signals. For the relative comparison of prediction performance, the proposed predictors are compared with a conventional ARMA linear prediction model. Experimental results show that the neural networks based adaptive predictor ourperforms the traditional linear scheme significantly. We also find that the cascade combination predictor is well suitable for the prediction of the time series which contain large variations of signal amplitude.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.3
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• pp.29-35
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• 2010

This paper discusses a method that can enhance the exactness of depth estimation of an image by PCA(Principle Component Analysis) based on feature reduction through learning algorithm. In estimation of the depth of an image, hyphen such as energy of pixels and gradient of them are found, those selves and their relationship are used for depth estimation. In such a case, many features are obtained by various filter operations. If all of the obtained features are equally used without considering their contribution for depth estimation, The efficiency of depth estimation goes down. This paper proposes a method that can enhance the exactness of depth estimation of an image and its processing speed is considered as the contribution factor through PCA. The experiment shows that the proposed method(30% of an feature vector) is more exact(average 0.4%, maximum 2.5%) than using all of an image data in depth estimation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.61-72
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• 2005

For the improvement of coding efficiency, the H.264 standard uses new coding tools such as VBS, 1/4-pel accurate ME, multiple references, intra prediction, loop filter, etc. Using these coding tools, H.264 has achieved significant improvements from rate-distortion point of view compared to existing standards. However, the encoder complexity is greatly increased due to these coding tools. We focus on the complexity reduction method of MB mode selection. Among all modes which can be selected, $8{\times}8$ and intra $4{\times}4$ mode have higher complexity than the others. So we propose the methods for reduction of the $8{\times}8$ and intra $4{\times}4$ mode complexity by using information of other modes with relatively low complexity. Simulation results show that the proposed methods save up to $54.6{\%}$ of total encoding time while keeping the average decrease about 0.012dB in PSNR.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.49-56
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• 2017

This paper proposes a method to improve the left right discrimination performance by eliminating the imaginary target based on the frequency features of the beam pattern for bow array. The beamwidth of the imaginary target is wider than that of the real target. If an azimuth axis is considered as a time axis, the real and the imaginary targets can be assumed as high and low frequencies, respectively. To eliminate the imaginary target which has a low frequency component, we design a cut-off frequency of the High Pass Filter (HPF) using the back-lobe imaginary beamwidth. The real target is estimated by eliminating the imaginary target by applying HPF to the entire power of the beamformer output. Computer simulations show that the proposed method can increase the left right discrimination performance above 8 dB on average.

• Journal of the Ergonomics Society of Korea
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• v.30 no.6
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• pp.689-694
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• 2011

Objective: The aim of this study is to evaluate the fluctuation of signal amplitude during repetitive dynamic contraction based on surface electromyography(EMG). Background: The most previous studies were considered isometric muscle contraction and they were difference to smoothing window length by moving average filter. In practical, the human movement is dynamic state. Dynamic EMG signal which indicated as the nonstationary pattern should be analyzed differently compared with the static EMG signal. Method: Ten male subjects participated in this experiment, and EMG signal was recorded by biceps brachii, anterior/posterior deltoid, and upper/lower trapezius muscles. The subject was performed to repetitive right horizontal lifting task during ten cycles. This study was considered three independent variables(muscle, amplitude processing technique, and smoothing window length) as the within-subject experimental design. This study was estimated muscular activation by means of the linear envelope technique(LE). The dependent variable was set coefficient of variation(CV) of LE for each cycle. Results: The ANOVA results showed that the main and interaction effects between the amplitude processing technique and smoothing window length were significant difference. The CV value of peak LE was higher than mean LE. According to increase the smoothing window length, this study shows that the CV trend of peak LE was decreased. However, the CV of mean LE was analyzed constant fluctuation trend regardless of the smoothing window length. Conclusion: Based on these results, we expected that using the mean LE and 300ms window length increased reproducibility and signal noise ratio during repetitive dynamic muscle contraction. Application: These results can be used to provide fundamental information for repetitive dynamic EMG signal processing.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.44-50
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• 2002

Conventional acoustic echo cancelers using ES(Exponentially weighted Stepsize) algorithm have simple operational configuration and fast convergence speed batter then NLMS algorithm, but they are very weak in external noise because ES algorithm updates filter taps using an average energy reduction rate of room impulse response in specific acoustical condition. So, a new configuration of acoustic echo canceler with stepsize generator and selector is proposed in this thesis. The proposed stepsize generator and selector improve conventional acoustic echo canceler's weakness in external noise and improve the system robustness. The stepsize generator generates additional stepsize value using moving averager, which is the residual noise energy of error signal multiplied by constant ${\gamma}$. The stepsize selector selects the stepsize value that has better performance in an acoustic echo canceler using a coefficient decision factor ${\Delta}_{differ}$ The simulation results show that the proposed algorithm reduces residual error by 5[dB] to 10[dB], improves misadjustment regardless of external noise's SNR. 

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.36-42
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• 2001

This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5％ average accuracy for sound source detection.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.134-142
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• 2016

In this paper, a method synthesizing speech signal using the 40 kHz ultrasonic signals reflected from the articulatory muscles was introduced and performance was evaluated. When the ultrasound signals are radiated to articulating face, the Doppler effects caused by movements of lips, jaw, and chin observed. The signals that have different frequencies from that of the transmitted signals are found in the received signals. These ADS (Acoustic-Doppler Signals) were used for estimating of the speech parameters in this study. Prior to synthesizing speech signal, a quantitative correlation analysis between ADS and speech signals was carried out on each frequency bin. According to the results, the feasibility of the ADS-based speech synthesis was validated. ADS-to-speech transformation was achieved by the joint Gaussian mixture model-based conversion rules. The experimental results from the 5 subjects showed that filter bank energy and LPC (Linear Predictive Coefficient) cepstrum coefficients are the optimal features for ADS, and speech, respectively. In the subjective evaluation where synthesized speech signals were obtained using the excitation sources extracted from original speech signals, it was confirmed that the ADS-to-speech conversion method yielded 72.2 % average recognition rates.

• Korean Journal of Optics and Photonics
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• v.10 no.3
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• pp.226-232
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• 1999

We have investigated the output power performance of a continuous-wave chromium-doped YAG laser ($Cr^{4+}$:YAG laser), pumped by a Nd: YAG laser. With an output coupler of 1% transmission, we obtained as much as 400 mW of the average output power at 1.450 ${\mu}{\textrm}{m}$. The power slope efficiency of 8%, when the gain medium was cooled to 2$0^{\circ}C$. We tuned the wavelength of the $Cr^{4+}$:YAG laser, at pump power of 6 W, by using birefringent filter. As a result, lasing wavelength was tuned from 1.399 ${\mu}{\textrm}{m}$ to 1.532 ${\mu}{\textrm}{m}$, with the maximum output power of 340 mW at 1.4903 ${\mu}{\textrm}{m}$ and the linewidth was 0.21 nm.

• 한국태양에너지학회:학술대회논문집
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• 2009.11a
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• pp.62-67
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• 2009

This paper describes the characteristics of scroll type stirling engine receiver. Scroll type stirling engine operated scroll compressor and expander instead of piston. Pass dimension of the receiver was $14(W){\times}14(H)$ mm and total pass length was 5,049 mm. External dimension of the receiver was $300{\times}300mm$. The experimental facility consisted of parabolic dish concentrator, compressor to supply air, triplex air filter, and flowmeter. In this study, basic experimental conditions were set at a inlet pressure of 5 bar and volume flow rate of $25m^3/hr$. As a result, air temperature in receiver at each measuring position of point 1, 2, 3 were $241^{\circ}C$, $465^{\circ}C$, and $542^{\circ}C$ respectively at inlet pressure of 5.5 bar and volume flow rate of $24.6m^3/hr$. As DNI increasing, heat transfer coefficient of the receiver changed from $695W/m^2K$ to $827W/m^2K$. Average heat transfer coefficient of receiver in the experiment was $798W/m^2K$. In addition, receiver efficiency became about 83%.