• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.5
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• pp.126-131
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• 2004

As a noninvasive imaging method, optical coherence tomography system has been extensively studied because it has some advantages such as imaging of high resolution, low cost, and compact size configuration. In order to improve the SNR of OCT system, two types of interferometers were configured and then, we compared simulation with measurement of reference sample. In the OCT system is configured with Michelson interferometer, the contrast of cross-sectional image is reduced with low SNR detection which is due to loss of feedback interference signal from light source part. Also, in order to image measured data with real time, image processing program is constructed. From results of simulation, it is confirmed that improved Michelson interferometer is improved about 10[dB] with a 50 : 50 fiber coupler. And from the measurement of reference sample, about 5[dB] is improved with a 50 : 50 fiber coupler. It is confirmed that the OCT system is configured with the improved Michelson interferometer which has a good distinctive cross-sectional image due to higher contrast.

• Proceedings of the KIEE Conference
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• 2000.07c
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• pp.1605-1607
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• 2000

In this paper, the design of class AB BiCMOS log-domain filter for low-voltage and low-power was proposed. This filter is consist of a log-domain integrator using folded junctions with capacitor connected to emitter and it's class AB structure. A comparison between the proposed class AB BiCMOS log-domain filter and classical class A BiCMOS log-domain filter is drawn on the basis of SNR, THD and the frequency response. This comparison shows proposed filter are more than good SNR, THD and frequency characteristics than more class A log-domain filter for low voltage and low power.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.687-690
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• 1999

Many high-resolution algorithms based on the eigen-decomposition analysis of observed covariance matrix, such as MVE, MUSIC, and EVM, have been proposed. However, the resolution of spectral estimates for these algorithms is severely degraded when Signal-to-Noise Ratio (SNR) is low and arrival angles of signal are close to each other. And EVM and MUSIC is powerful for the characteristic of SNR. But have the limitation that the number of signals presented is known. While MVE is bad the characteristic of SNR. In this study, we propose a modified MVE to enhance the resolution for Direction-Of-Arrival (DOA) estimation of underwater acoustic signal. This is to remove the limitation that existing algorithms should know the information for the number of signals. Because the algorithms founded on the eigen value estimate DOA with only the noise subspace, they have the high-resolution characteristic. And then, with the method reducing the effect of the signal subspace, we are to reduce the degradation because of complementary relationship between the signal subspace and the noise subspace. This paper, with using the simulation data, we have estimated the proposed algorithms, compared it with other high-resolution algorithms. The simulation results show that the modified MVE proposed is accurate and has a better resolution even though SNR is low, under the same condition.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.9C
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• pp.683-690
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• 2008

By using inherent stopping criteria of LDPC codes, the average number of iterations can be substantially reduced at high signal to noise ratio (SNR). However, we encounter a problem when hybrid automatic repeat request (H-ARQ) systems are applied. Frequent failures of decoding at low SNR region imply that the decoder leaches the maximum number of iterations frequently and thus the decoding complexity increases. In this paper, we propose a combination of stopping criteria using the syndrome weight of tentative codeword. By numerical analysis, it is shown that the decoding complexity of given H-ARQ system is reduced by 70-80% with the proposed algorithms.

• ETRI Journal
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• v.33 no.4
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• pp.528-536
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• 2011

In this paper, we analyze the location accuracy of real-time locating systems (RTLSs) in multipath environments in which the RTLSs comply with the ISO/IEC 24730-2 international standard. To analyze the location error of RTLS in multipath environments, we consider a direct path and indirect path, in which time and phase are delayed, and also white Gaussian noise is added. The location error depends strongly on both the noise level and phase difference under a low signal-to-noise ratio (SNR) regime, but only on the noise level under a high SNR regime. The phase difference effect can be minimized by matching it to the time delay difference at a ratio of 180 degrees per 1 chip time delay (Tc). At a relatively high SNR of 10 dB, a location error of less than 3 m is expected at any phase and time delay value of an indirect signal. At a low SNR regime, the location error range increases to 8.1 m at a 0.5 Tc, and to 7.3 m at a 1.5 Tc. However, if the correlation energy is accumulated for an 8-bit period, the location error can be reduced to 3.9 m and 2.5 m, respectively.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.40 no.5
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• pp.289-297
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• 2003

The performance of a bimodal system is affected by the accuracy of the endpoint detection from the input signal as well as the performance of the speech recognition or lipreading system. In this paper, we propose the endpoint detection method which detects the endpoints from the audio and video signal respectively and utilizes the signal to-noise ratio (SNR) estimated from the input audio signal to select the reliable endpoints to the acoustic noise. In other words, the endpoints are detected from the audio signal under the high SNR and from the video signal under the low SNR. Experimental results show that the bimodal system using the proposed endpoint detector achieves satisfactory recognition rates, especially when the acoustic environment is quite noisy.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.935-941
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• 2007

In this paper, we suggest the power allocation method which enhances the performance in high SNR for LDPC codes. In this method, bit power is unequally allocated proportionally to the difference of the degree distributions of variable and check nodes of Tanner graph between practically used codes and the codes optimized by density evolution. Simulation is performed to the codes in IEEE 802.16e standards, and the results show that the proposed method works well in high SNR.

• The Bulletin of The Korean Astronomical Society
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• v.36 no.1
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• pp.87.2-87.2
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• 2011

We report our 110 ks Chandra observations of the supernova remnant (SNR) 0104-72.3 in the Small Magellanic Cloud (SMC). The X-ray morphology shows two prominent lobes along the northwest-southeast direction and a soft faint arc in the east. Previous low resolution X-ray images attributed the unresolved emission from the southeastern lobe to a Be/X-ray star. Our high resolution Chandra data clearly shows that this emission is diffuse, shock-heated plasma, with negligible X-ray emission from the Be star. The eastern arc is positionally coincident with a filament seen in optical and infrared observations. Its X-ray spectrum is well fit by plasma of normal SMC abundances, suggesting that it is from shocked ambient gas. The X-ray spectra of the lobes show overabundant Fe, which is interpreted as emission from the reverse-shocked Fe-rich ejecta. The overall spectral characteristics of the lobes and the arc are similar to those of Type Ia SNRs, and we propose that SNR 0104-72.3 is the first case for a robust candidate Type Ia SNR in the SMC. On the other hand, the remnant appears to be interacting with dense clouds toward the east and to be associated with a nearby star-forming region. These features are unusual for a standard Type Ia SNR. Our results suggest an intriguing possibility that the progenitor of SNR 0104-72.3 might have been a white dwarf of a relatively young population.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.1
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• pp.101-107
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• 2011

The background noises and distortions by channel are major factors that disturb the practical use of speech recognition. Usually, noise reduce the performance of speech recognition system DSR(Distributed Speech Recognition) based speech recognition also bas difficulty of improving performance for this reason. Therefore, to improve DSR-based speech recognition under noisy environment, this paper proposes a method which detects accurate speech region to extract accurate features. The proposed method distinguish speech and noise by using entropy and detection of spectral energy of speech. The speech detection by the spectral energy of speech shows good performance under relatively high SNR(SNR 15dB). But when the noise environment varies, the threshold between speech and noise also varies, and speech detection performance reduces under low SNR(SNR 0dB) environment. The proposed method uses the spectral entropy and harmonics of speech for better speech detection. Also, the performance of AFE is increased by precise speech detections. According to the result of experiment, the proposed method shows better recognition performance under noise environment.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2480-2482
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• 1998

In this paper, the implementation of dual rate ADPCM using G.723 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows the same SNR as the high rate 40Kbps coding algorithm, while the low rate 16Kbps coding algorithm shows the lower SNR than the high rate 40Kbps coding algorithm for large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are tested for good trade off data rate and speech quality. Also the low pass filter effect of speech input and output devices is simulated at several cut-off frequencies. To simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.