• 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.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.49 no.4
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• pp.16-22
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• 2012

In this paper, the forensic marking algorithm is proposed using psychoacoustic model II and MDCT for high-quality audio. The proposed forensic marking method, that inserts the user fingerprinting code of the audio content into the selected sub-band, in which audio signal energy is lower than the spectrum masking level. In the range of the one frame which has 2,048 samples for FFT of original audio signal, the audio forensic marking is processed in 3 sub-bands. According to the average attack of the fingerprinting codes, one frame's SNR is measured on 100% trace ratio of the collusion codes. When the lower strength 0.1 of the inserted fingerprinting code, SNR is 38.44dB. And in case, the added strength 0.5 of white gaussian noise, SNR is 19.09dB. As a result, it confirms that the proposed audio forensic marking algorithm is maintained the marking robustness of the fingerprinting code and the audio high-quality.

• Phonetics and Speech Sciences
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• v.9 no.2
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• pp.119-124
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• 2017

This study investigates how perception of easy sentences varies with prosody pattern, sound intensity, and signal-to-noise ratio(SNR) in young adults with normal hearing who were in their 20's. The results showed that the presence of proper prosody pattern in the sentences increased correct perception rate of the target sentences, and that the lower the intensity and SNR, the lower the sentence perception scores. The results also showed that SNR had a greater effect on the sentence perception scores than sound intensity. There was a significant decrease of perception scores starting at the level of 15 dB and +3 SNR for the sentences with prosody pattern, while starting at the level of 18 dB and +6 SNR for the sentences without prosody pattern, ending up with a very poor perception score as sound intensity and SNR gets lower. There was a significant difference in the perception score of the sentences with prosody pattern between 20 year-old group and 21 year or older group in several listening conditions of sound intensity and SNR.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4C
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• pp.446-452
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• 2009

OFDM system has a disadvantage of sensitiveness about the effect of frequency offset caused by the discord of oscillators in the transmitter and receiver. Either, the frequency offsets in mobile radio channels distort the orthogonality between sub-carriers resulting in the inter-carrier interference(ICI). In this paper, we analyze the effect of the ICI and propose a new method using SC technique. To analyze BER(Bit Error Rate) and CIR(Carrier to Interference Ratio) performance of the proposed method. the simulation program MATLAB is used. By the simulation results, SNR performance is improved by this method. In case the frequency offset is 0.3 and 0.5, SNR gains are over 0.5dB and 1dB in the BPSK modulation and 1dB and 2dB in the QPSK modulation at BER of $10^{-3}$ respectively. In addition, CIR performance is improved over 15dB. As a result, the proposed method is more effective to improve the system performance than the conventional method.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.5
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• pp.21-29
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• 2008

This paper proposes a carrier frequency of set estimation algorithm for IEEE802.11n system. As IEEE802.11n is a multiple input multiple output(MIMO) system, so there are several combining techniques which are used in multiple receive antenna system. In this paper, we propose hybrid carrier frequency offset estimation algorithms using combining techniques in multiple receive antenna systems, and show that the proposed selection combining carrier frequency offset (CFO) estimation algorithm can estimate carrier frequency offset within 1/10 MSE error at SNR 10 dB in channel B and within 1/2 MSE error at SNR 10 dB in channel D rather than the conventional MIMO CFO one.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3C
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• pp.59-63
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• 2005

We propose non-DC-free generalized PRML (GPRML) that are suppressed DC contents for matching to the response of perpendicular magnetic recording channel with a ring-head. In addition, DC-free encoding is considered to prevent low-frequency disturbances. The SNR performance is obtained by combining the various PRML channels with DC-free and non-DC-free codes during the normalized recording density increases from 2.5 to 3.5. The GPRML detections without using DC-free code get SNR gains more than 1dB compared to the conventional PRML systems at 10/sup -5/BER. We confirmed that the rate 127/136 DC-free coded GPRML systems show good performances compared with the 16/17 non-DC-free coded GPRML systems. In results, DC-free coded GPRML detections get gains about 1.4dB and 2.0dB at the density of 3.3 and 3.5, respectively.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.225-230
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• 1998

In this paper, a Kalman filter approach for enhancing speech signals degraded by statistically independent additive nonstationary noise is developed. The autoregressive hidden markov model is used for modeling the statistical characteristics of both the clean speech signal and the nonstationary noise process. In this case, the speech enhancement comprises a weighted sum of conditional mean estimators for the composite states of the models for the speech and noise, where the weights equal to the posterior probabilities of the composite states, given the noisy speech. The conditional mean estimators use a smoothing spproach based on two Kalmean filters with Markovian switching coefficients, where one of the filters propagates in the forward-time direction with one frame. The proposed method is tested against the noisy speech signals degraded by Gaussian colored noise or nonstationary noise at various input signal-to-noise ratios. An app개ximate improvement of 4.7-5.2 dB is SNR is achieved at input SNR 10 and 15 dB. Also, in a comparison of conventional and the proposed methods, an improvement of the about 0.3 dB in SNR is obtained with our proposed method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.11
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• pp.1043-1050
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• 2004

In this paper, Phase noise is analyzed and a novel approach of the nonlinear approximation including second order term of phase noise is presented to analyze and quantize system performance. As results, in QPSK-OFDM system, when PLL loop bandwidth is 5.0 Hz, 1.0 kHB, 0.5 kHz respectively, there are about 0.6 dB, 1.0 dB, 1.7 dB SNR penalties at BER=10$\^$-4/ compared with system without phase noise in AWGN channel. In 16QAM modulation, there are about 1.9 dB, 3.2 dB, 6.7 dB SNR penalties at BER=10$\^$-4/ respectively. At QPSK-OFDM system, comparing the previous linear approximation method with our proposed nonlinear approximation method, there is similar BER performance at phase noise variance lower than 0.02, but certain difference occurs as variance increases more than 0.02. Furthermore, analytical BER results closely match with simulation results in the OFDM system employing QPSK and 16qAM modulation. And, BER performance of QPSK-OFDM system is considerably degraded because of the BER error floor if the phase noise variance becomes larger than 0.03.

• Journal of Internet Computing and Services
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• v.19 no.6
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• pp.1-7
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• 2018

This paper present an 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) coding method integrated with ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding) and LMS-MPC(Least Mean Square - Multi Pulse Coding) used V/UV/S(Voiced / Unvoiced / Silence) switching, compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. In integrating several methods, it is important to adjust the bit rate of voiced and unvoiced sound source to 8kbps while reducing the distortion of the speech waveform. In adjusting the bit rate of voiced and unvoiced sound source to 8 kbps, the speech waveform can be synthesized efficiently by restoring the individual pitch intervals using multi pulse in the representative interval. I was implemented that the ACLMS-MPC method and evaluate the SNR of APC-LMS in coding condition in 8kbps. As a result, SNR of ACLMS-MPC was 15.0dB for female voice and 14.3dB for male voice respectively. Therefore, I found that ACLMS-MPC was improved by 0.3dB~1.8dB for male voice and 0.3dB~1.6dB for female voice compared to existing MPC, ACFBD-MPC and LMS-MPC. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or internet phone. In the future, I will study the evaluation of the sound quality of 6.9kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source.

• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.21-25
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• 2013

The dual-stream based stereoscopic 3D HDTV broadcasting service was launched recently. Although the dual-stream based HDTV service has been successfully provided, the 3D HDTV broadcasting system requires more bandwidth efficient transmission schemes because it should convey both left and right HD resolution images simultaneously in the finite 6MHz bandwidth. In this paper, we consider more advanced ATSC transmission schemes that use higher modulation such as 16-QAM and concatenated RS code and block LDPC codes. Compared with conventional ATSC system and the modified ATSC system in [2], the proposed system has about 2.97dB and 1.12dB SNR gain at the payload data rate of 19.44Mbps compared with the existing ATSC system and the modified ATSC system [2]. Also, the proposed scheme requires only 1.05dB power increase for the 3D HDTV service, which is reasonable SNR increase value and applicable to the advanced 3D high definition broadcasting realization in limited 6MHz bandwidth.