• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1571-1576
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• 2013

This paper proposes an improvement algorithm of signal-to-noise ratios (SNRs) for speech signals under noisy environments. The proposed algorithm first estimates the SNRs in a low SNR, mid SNR and high SNR areas, in order to improve the SNRs in the speech signal from background noise, such as white noise and car noise. Thereafter, this algorithm subtracts the noise signal from the noisy speech signal at each bands using a spectrum sharpening method. In the experiment, good signal-to-noise ratios (SNR) are obtained for white noise and car noise compared with a conventional spectral subtraction method. From the experiment results, the maximal improvement in the output SNR results was approximately 4.2 dB and 3.7 dB better for white noise and car noise compared with the results of the spectral subtraction method, in the background noisy environment, respectively.

• Korean Journal of Optics and Photonics
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• v.30 no.2
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• pp.48-58
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• 2019

We investigate the signal-to-noise ratios (SNRs) of time-domain (i.e. pulsed illumination) and frequency-domain (i.e. chirped illumination) photoacoustic signals measured by a spherically focused ultrasound transducer for spherical absorbers. The simulation results show that the time-domain photoacoustic SNR is higher than that of frequency-domain photoacoustic signals, as reported in the previous literature. We understand the reason for this SNR gap between the two measurement modes by analyzing photoacoustic-signal spectra, considering the incident beam energy controlled by the maximum permissible exposure. As the result of this approach, we find that filtering off the DC term in the chirped signal's spectrum improves frequency-domain photoacoustic SNRs by up to approximately 5 dB. In particular, it is observed that photoacoustic SNRs are highly sensitive to an upper-frequency value of frequency filtering functions, and the optimal upper-frequency values maximizing the SNR are different in time- and frequency-domain photoacoustic measurements.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.31-37
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• 2009

In the two-way relay channel, the relay employ Amplify-and-Forward (AF) or Decode-and-Forward (DF) protocol, and broadcast the network-coded signal to both user. In the system, DF protocol provides maximum throughput at low signal to noise ratio(SNR). On the other hand, at high SNR, AF protocol provides maximum throughput. The paper propose active transmission scheme which employ Amplify-and-Forward or Decode-and-Forward protocol based on received SNR at the relay over Two-way relay channel. The optimal threshold is investigated numerically for switching the protocol. Through numerical results, we confirm that the proposed scheme outperforms conventional schemes over two-way relay channel.

• Journal of Digital Contents Society
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• v.17 no.2
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• pp.89-96
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• 2016

SVC(Scalable Video Coding), which is one form among video encoding technologies, makes video streaming with the various frame rate, resolution, and video quality by combining three different scalability dimensions: temporal, spatial, and video quality scalability. As the above SVC-encoded video streaming consists of one base layer and several enhancement layers, and a wireless AP(Access Point) chooses and sends a suitable layer according to the received power from the receiving terminals in the changeable wireless network environment, the receiving terminals supporting SVC are able to receive video streaming with the appropriate resolution and quality according to their received powers. In this paper, after the performance analysis for the received power, packet loss rate, PSNR(Required Peak Signal to Noise Ratio), video quality level and amount of received video data based on the number of SVC layers was performed, an efficient method for selecting the number of SVC layer satisfying the RSNR and minimizing the amount of received video data is proposed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9C
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• pp.840-845
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• 2006

This paper presents a partial response maximum likelihood (PRML) detection method using partial response signal-to-noise ratio (PRSNR) that evaluates the signal quality of asymmetric optical recording channel. It is confirmed that the equalizer maximizing the PRSNR value can be most properly adapted to the asymmetric optical recording channel. The proposed PRML detection using this result has 1.8dB SNR gain at $8.5{\times}10-5$ bit error rate compared to PRML detection using typical adaptive equalizer.

• Journal of Digital Convergence
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• v.13 no.10
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• pp.377-383
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• 2015

This study evaluates the convergence radiation protection performance by measuring the PSNR(peak signal-to-noise ratio) values of the image J in the image evaluation program based on increased relative to this exposure of radiation workers.The aim of this study was to evaluate radiation protection performance of apron for design of it's basic information. Method was used to PSNR of Image J program and good condition apron was more than 27dB, the PSNR value of poor condition apron appeared to be less than 24dB. The result is the normality were satisfied distribution and T-test values were statistically significant with p<0.001. Results of evaluation of the performance protective apron through the more easily accessible experimental conditions and methods in the clinical was confirmed distinctly different. in order to reduce the radiation exposure we need to evaluate convergence protection performance and to be having a good performance apron.

• Journal of the Society of Naval Architects of Korea
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• v.38 no.3
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• pp.117-122
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• 2001

The shock test of shipboard equipments is performed for the evaluation of the shock-resistant. capability by analyzing the maximum acceleration, the effective time duration and the shock response spectrum, etc. But some measured signals have impulsive noise and gaussian white noise because of the ambient noise, the acquisition equipment error and the transient movement of cables during the shock test. The improved transient signal analysis method which removes the noise of measured signal using the threshold policy of the median filter and the orthogonal wavelet coefficients is proposed. It was verified that the signal-to-noise ratio was improved about 30dB by the numerical simulation. And the shock response spectrum was extracted using the denoised shock response signal which was applied by this proposed method.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.723-726
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• 2009

In this research, We simulated performances of error reduction algorithm for the speech signal based on the microphone array-based beamforming method in speech recognition system and analyzed its performance. Also, we processed speech signal adopted from microphone array and maximum signal to noise ratio for each channel, and then compared them with signal to noise ratio of speech signal. Speech recognition rate is improved from 54.2% to 61.4% in case 1 and is improved from 41.2% to 50.5% in case 2 of the lower signal to noise ratio. Therefore the average reduction rates are showed 15.7% in case 1.

• Journal of the Korean Institute of Intelligent Systems
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• v.24 no.1
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• pp.28-32
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• 2014

For a signal such as speech showing piece-wise linear shape in a very short time period, a nonuniform sampling method based on the inflection point detection (IPD) is proposed to reduce data rate. The method exploits the geometrical characteristics of signal further than the existing local maxima/minima detection (MMD) based sampling method. As results, the reconstructed signal by the interpolation of the IPD based sampled data resembles the original speech more. Computer simulation shows that the proposed IPD based method produces about 9~23 dB improvement over the existing MMD method. To show the usefulness of the IPD technique, it is applied to speech coding, and compared to the continuously variable slope delta modulation (CVSD). The nonuniformly sampled data is binary coded with one bit flag set "1". Noninflection samples are not sent, but only flag bits set 0 are sent. The method shows 0.3 ~ 9 dB SNR and 0.5 ~ 1.3 mean opinion score (MOS) improvements over the CVSD.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.6A
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• pp.913-922
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• 2001

Multi-Carrier CDMA 시스템에서 불완전한 채널 예측과 다중사용자 간섭은 시스템의 성능 저하를 유발한다. 본 논문에서는 역방향 링크에서 파일럿 심볼을 이용하여 채널을 예측하는 Multi-Carrier CDMA BPSK 시스템에서 불완전한 채널 예측과 다중사용자 간섭의 영향을 연구하였다. 더욱이, 라이시안 페이딩 채널의 역방향 링크에서 최대비 합성 공간 다이버시티와 선택 합성 공간 다이버시티 기법을 적용한 성능 개선에 대하여 연구하였다. 수치계산 결과, 파일럿 심볼을 이용하여 채널을 예측하는 Multi-Carrier CDMA BPSK 시스템의 BER 성능은 등가 잡음 대역폭의 분산($\sigma$$^2$$_{c}$)에 매우 민감하나 무선 채널에서의 신호 전력 대 잡음 전력비에는 그리 심하지 않았다. 불완전한 채널 예측에 의한 BER 열화와 파일럿 심볼 간격의 최적화는 라이시안 페이딩 채널에서 Multi-Carrier CDMA BPSK 시스템의 BER 계산을 통하여 얻을 수 있었다. 그리고 불완전한 채널 예측에 의한 BER 열화는 신호 전력 대 잡음 전력비와 채널 예측 필터의 등가 잡음 대역폭에 관계하고 있음을 알 수 있었다.