• Journal of electromagnetic engineering and science
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• v.11 no.4
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• pp.274-281
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• 2011

High-speed telecommunication systems are influenced by electromagnetic environments because they need a wide bandwidth to transmit signals. Immunity tests of telecommunication equipment are effective for improving its immunity to electromagnetic environments. However, immunity tests are expensive to carry out because there are several different tests. The correlation among the tests should therefore be examined in order to reduce the kinds of tests that are necessary. This paper investigates the correlation between the electrical fast transient/burst (EFTB) test and the radio frequency (RF) conductive immunity test. Imitation equipment was constructed with a balun, and a baseband signal was transmitted from the associated equipment to the imitation equipment. Then, disturbances were applied to the equipment, and the telecommunication quality was evaluated by using the bit error rate (BER). The results from the EFTB test indicated that the BER was less than $6{\times}10^{-5}$ and the value was independent of the peak value. The results from the RF conductive immunity test indicated that the BER was affected by the longitudinal conversion loss (LCL).

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.10a
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• pp.65-67
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• 2017

Due to the development of mobile computing and wearable technology, various wearable devices for measuring bio-signals in everyday life have been developed and popularized, and healthcare services utilizing bio-signals are attracting attention. In recent years, healthcare services have been developed and studied using various bio-signal analysis tools. Most bio-signal analysis studies utilize Matlab. However, in order to apply the algorithm developed in Matlab to the system, it is necessary to convert the source. We want to provide a smart interface that can skip source conversion. In this paper, we develop an interface to run the source file itself in the system by omitting the conversion technique for applying the algorithm developed in Matlab to the system.

• Journal of the Korean Institute of Telematics and Electronics D
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• v.36D no.2
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• pp.81-89
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• 1999

Among various wavelength conversion technologies, that using the cross-gain modulation in laser diode makes it possible to deal with the high speed signal quite simply and efficiently. In this paper, presented was the applicability of an improved time-domain large-signal dynamic model as a CAD tool to analyzed the characteristics of SSGDBR(Superstructure Grating Distributed Bragg Reflector) laser diodes used for wavelength converters. Using this model, it was shown that this kind of wavelength converter can provide the widely tunable wavelength conversion of the high speed data above 10 Gbps. We also investigated the effect of input optical power and the bias current on the characteristics of the device such as extinction ration and eye diagram. The modeling results show very similar trend to the experimental reports.

• 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.

• Proceedings of the KSR Conference
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• 2003.10c
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• pp.247-251
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• 2003

The velocity of train which is measured from pulse generator attached to TM is used for displaying or control signal of inverter and so on. Measured signals increase and decrease step-by-step by pulse counting or monotonously by F/V conversion. But noises and signal distortions by measuring error like alias make it difficult to provide correct velocity infomation and estimate the acceleration. In this paper, we investigated the performance of Smoothing method for suppressing the noises in velocity signals. And the difference between Smoothed signal and origin velocity signals is inspected and the comparison with low pass filtering show applicable of Smoothing method for noise rejection and the estimation of signal. Finally, acceleration curves estimated from Smoothing method are compared with real accelerator signal attached to train.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.11
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• pp.17-23
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• 2004

The implementation and design of the wideband IFIU using aperture open loop resonator and reversed phase technique to reduce the local oscillator leakage signal was represented in this paper. The local oscillator leakage signal is generated in stage of frequency conversion, especially in frequency conversion of fully digital modulation signal close to DC signal. The leakage signal and spurious signals, which have effects on adjacent channel or in-band channel as interference signals, were reduced below -60 dBc for 45 Mbps and 155 Mbps IF interface units. The group delay for both IFIUs shows low ripple characteristics of 15 ns and 8 ns, respectively. Also, the amplitude ripple characteristic in 150 MHz bandwidth with L-band center frequency satisfies the required specification of 2 dB. The implemented IFIU provides the required specifications for wideband satellite communication system.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.49 no.11
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• pp.933-940
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• 2021

The space-borne SAR(Synthetic Aperture Radar) system radiates the microwave signal and receives the backscattered signal. The received signal is converted to digital at the Digital Receiver, which is implemented at the end of the SAR sensor receiving chain. The converted signal is formated after signal processing such as filtering and data compression. Two quantization are conducted in the Digital Receiver. One quantization is an analog to digital conversion at ADC(Analog-Digital Converter). Another quantization is the BAQ(Block Adaptive Quantization) for data compression. The quantization process is a conversion from a continuous or higher bit precision to a discrete or lower bit precision. As a result, a quantization noise is inevitably occurred. In this paper, the impact of two quantization processes are analyzed in a view of SNR degradation.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.6
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• pp.1103-1108
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• 2023

In this paper, we propose a new algorithm for attenuating the background noises in acoustic signal. This algorithm improves the noise attenuation performance by using the FNN(: Full-connected Neural Network) deep learning algorithm instead of the existing adaptive filter after wavelet transform. After wavelet transforming the input signal for each short-time period, noise is removed from a single input audio signal containing noise by using a 1024-1024-512-neuron FNN deep learning model. This transforms the time-domain voice signal into the time-frequency domain so that the noise characteristics are well expressed, and effectively predicts voice in a noisy environment through supervised learning using the conversion parameter of the pure voice signal for the conversion parameter. In order to verify the performance of the noise reduction system proposed in this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed. As a result of the experiment, the proposed deep learning algorithm improved Mean Square Error (MSE) by 30% compared to the case of using the existing adaptive filter and by 20% compared to the case of using the STFT(: Short-Time Fourier Transform) transform effect was obtained.

• Journal of Broadcast Engineering
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• v.20 no.2
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• pp.265-272
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• 2015

A new technology on video frame rate up-conversion (FRUC) is presented by combining the median filter and motion estimation (ME) with an occlusion detection (OD) method. First, ME is performed to have a motion vector. Then, the OD method is used to refine motion vector in the occlusion region. Since the wrong motion vector can be obtained with high possibility in the occluded area, a median filtering that less depends on the motion vector is applied to that area, and since the motion vector is continuous and robust in the non-occluded area, BDMC(Bi-Directional Motion Compensated interpolation) is applied to obtain interpolated image in that area. BDMC using the bi-directional motion vectors achieves good results when continuity and robustness of the motion vector is higher. Experimental results show that the proposed algorithm provides better performance than the conventional approach. The average gain of PSNR (Peak Signal to Noise Ratio) is approximately 0.16 dB in the test sequences compared with BDMC.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.34-44
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• 2010

In this study, the algorithms for multichannel format conversion and robust representation of spatial sound information are proposed. In the spatial analysis, the directional information of sound source is estimated and sound sources are separated from stereo signal. In the spatial resynthesis, the multichannel matrixing with spatial repanning and post-scaling method are applied to represent a spatial sound. The conventional method about channel format conversion has the problem that the energy of sound source and the spatial information are not preserved in the desired channel format. Because the proposed method is designed in consideration of the target multichannel format and its resynthesized signal, the robust representation of spatial sound can be achieved in the multichannel format conversion.