• Molecules and Cells
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• v.44 no.9
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• pp.627-636
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• 2021

The three-dimensional organization of chromatin and its time-dependent changes greatly affect virtually every cellular function, especially DNA replication, genome maintenance, transcription regulation, and cell differentiation. Sequencing-based techniques such as ChIP-seq, ATAC-seq, and Hi-C provide abundant information on how genomic elements are coupled with regulatory proteins and functionally organized into hierarchical domains through their interactions. However, visualizing the time-dependent changes of such organization in individual cells remains challenging. Recent developments of CRISPR systems for site-specific fluorescent labeling of genomic loci have provided promising strategies for visualizing chromatin dynamics in live cells. However, there are several limiting factors, including background signals, off-target binding of CRISPR, and rapid photobleaching of the fluorophores, requiring a large number of target-bound CRISPR complexes to reliably distinguish the target-specific foci from the background. Various modifications have been engineered into the CRISPR system to enhance the signal-to-background ratio and signal longevity to detect target foci more reliably and efficiently, and to reduce the required target size. In this review, we comprehensively compare the performances of recently developed CRISPR designs for improved visualization of genomic loci in terms of the reliability of target detection, the ability to detect small repeat loci, and the allowed time of live tracking. Longer observation of genomic loci allows the detailed identification of the dynamic characteristics of chromatin. The diffusion properties of chromatin found in recent studies are reviewed, which provide suggestions for the underlying biological processes.

• Mass Spectrometry Letters
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• v.1 no.1
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• pp.17-20
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• 2010

As background significantly affects measurement accuracy and a detection limit in determination of the trace amounts of uranium, it is necessary to minimize the impurities in the filaments used for thermal ionization mass spectrometry (TIMS). We have varied the degassing condition such as the heating currents and duration times to reduce the backgrounds from the filaments prepared with zone-refined rhenium tape. The most efficient degassing condition of the heating current and the duration time was determined as 3.5 A and 60 min, respectively. The TIMS measurement combined with the isotope dilution mass spectrometry (IDMS) technique showed that the uranium backgrounds were determined to be in a few fg level from blank rhenium filaments. The background minimized filaments were utilized to measure the uranium isotope ratios of a U030 (NIST) standard sample. The excellent agreement of the measurement with the certified isotope ratios showed that the degassing procedure optimized in this study efficiently reduced the impurity signals of uranium from blank rhenium filaments to a negligible level.

• Journal of information and communication convergence engineering
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• v.10 no.2
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• pp.162-167
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• 2012

This paper proposes a speech processing system based on a model of the human auditory system and a noise reduction neural network with fast Fourier transform (FFT) amplitude and phase spectrums for noise reduction under background noise environments. The proposed system reduces noise signals by using the proposed neural network based on FFT amplitude spectrums and phase spectrums, then implements auditory processing frame by frame after detecting voiced and transitional sections for each frame. The results of the proposed system are compared with the results of a conventional spectral subtraction method and minimum mean-square error log-spectral amplitude estimator at different noise levels. The effectiveness of the proposed system is experimentally confirmed based on measuring the signal-to-noise ratio (SNR). In this experiment, the maximal improvement in the output SNR values with the proposed method is approximately 11.5 dB better for car noise, and 11.0 dB better for street noise, when compared with a conventional spectral subtraction method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.2
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• pp.327-329
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• 2021

Underwater acoustic communication channel is influenced by environmental parameters such as multipath, background noise and scattering. Therefore, a transmitted signal is influenced by the sea surface and the sea bottom boundaries, and a received signal shows a delay spread. These factors create a noise in the image and degrade the quality of underwater acoustic communication. To solve these problems, in this paper, we evaluate the performance of an underwater acoustic communication model using a denoising auto-encoder used for unsupervised learning. Noise images generated by the underwater multipath channel were collected and used as training data. Experimental results were analyzed as a PSNR parameter that expressed the noise ratio of the two images.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1004-1016
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• 2007

In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).

• 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.

• Proceedings of the Korea Committee for Ocean Resources and Engineering Conference
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• 2004.11a
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• pp.204-209
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• 2004

It is well known fact that acoustic positioning systems are absolutely needed for various underwater operations. According to the distances between their sensors they are classified into three parts: long baseline(LBL), short baseline(SBL), and ultra-short baseline(USBL). Among them the USBL system is widely used because of its simplicity, although it is the most inaccurate. Recently, in order to increase the positioning accuracy, various USBL systems using broadband signal such as MFSK(Multiple Frequency Shift Keying) are produced. However, their positioning accuracy is still limited by background noise and reflected waves. Therefore, there is difficulty in applying the USBL system using MFSK signal in a shallow water with noisy conditions. In order to examine the effect of the noise and wave reflections this paper analyze position errors for various conditions using numerical simulations. The simulation results say that tile SNR must be greater than 20dB and errors in the vertical direction are slightly increased by wave reflections by upper and lower boundaries.

• Journal of ILASS-Korea
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• v.2 no.4
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• pp.7-14
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• 1997

Laser Raman scattering method has been applied to measure equivalence ratio of methane/air mixture in injected spray. We used high power KrF excimer laser$(\lambda=248nm)$ and a high gain ICCD camera to capture low intensity signal. Raman shifts and Raman scattering cross -sections of $H_2,\;O_2,\;N_2,\;CO_2,\;CH_4\;and\;C_3H_8$ are measured precisely. Our results show an excellent agreement with those of other groups. Mole fraction measurement of $O_2\;and\;N_2$ from air shows that $O_2:N_2=0.206:0.794$. We used gas injector which was operated at 1 bar. Methane is used as a fuel. Spray region is $10mm\times37mm$ and this region is divided into 80 points. In Raman signals are obtained and ensemble averaged for each point. 3-d and contour plot of distribution of equuivalence ratio is presented. Our measured results show that the equivalence ratio of methane/air mixture in methane-rich region is reasonable. However, more study is necessary for methane-lean region because background noise level is almost same as Raman intensity of methane.

• Journal of Biomedical Engineering Research
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• v.26 no.3
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• pp.177-184
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• 2005

In this paper, we proposed a new hearing aid algorithm to improve SNR(signal to noise ratio) of noisy speech signal and speech perception. The proposed hearing aid algorithm is a multi-band loudness compensation based independent component analysis (ICA). The proposed algorithm was compared with a conventional spectral subtraction algorithm on behind-the-ear type hearing aid. The proposed algorithm successfully separated a target speech signal from background noise and from a mixture of the speech signals. The algorithms were compared each other by means of SNR. The average improvement of SNR by ICA based algorithm was 16.64dB, whereas spectral subtraction algorithm was 8.67dB. From the clinical tests, we concluded that our proposed algorithm would help hearing aid user to hear clearly a target speech in noisy conditions.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.