• Proceedings of the Korean Society of Medical Physics Conference
• /
• 2002.09a
• /
• pp.367-369
• /
• 2002

We have designed ramp profile excitation pulse based on the Shinnar-Le Roux (SLR) algorithm. The algorithm provides many advantages to pulse designers. The first advantage is the freedom of deciding the amplitudes, frequencies, and ripple sizes of stopband, passband, and transition band of pulse profile. The second advantage is the freedom of deciding the pulse phase, more specifically, minimum phase, linear phase, maximum phase, and any phase between them. The minimum phase pulse is the best choice in the case of 3D TOF, because it minimizes the echo time, which implies the best image quality in the same MR examination condition. In addition, the half echo technique is slightly modified in our case. In general, using the half echo technique means that the acquired data size is half and the rest part can be filled with complex conjugate of acquired data. But in our case, the echo center is just shifted to left, which implies the reduction of echo time, and the acquired data size is the same as the one without using the half echo technique. In this case, the increase of right part of data leads to improvement of the resolution and the decrease of left part of data leads to decrease of signal to noise ratio. Since in the case of 3D TOF, the signal to noise ratio is sufficiently high and the resolution is more important than signal to noise ratio, the proposed method appears to be significantly affective and gives rise to the improved high resolution angiograms.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1985.10a
• /
• pp.87-94
• /
• 1985

In this paper, enhancement of speech corrupted by additive white or colored noise is stuided. The nuconstrained frequency-domain block least-mean-square (UFBLMS) adaptation algorithm and its frequency-weighted version are newly applied to speech enhancement. For enhancement of speech degraded by white noise, the performance of the UFBLMS algorithm is superior to the spectral subtraction method or Wiener filtering technique by more than 3 dB in segmented frequency-weighted signal-to-noise ratio(FWSNERSEG) when SNR of speech is in the range of 0 to 10 dB. As for enhancement of noisy speech corrupted by colored noise, the UFBLMS algorithm is superior to that of the spectral subtraction method by about 3 to 5 dB in FWSNRSEG. Also, it yields better performance by about 2 dB in FWSNR and FWSNRSEG than that of time-domain least-mean-square (TLMS) adaptive prediction filter(APF). In view of the computational complexity and performance improvement in speech quality and intelligibility, the frequency-weighted UFBLMS algorithm appears to yield the best performance among various algorithms in enhancing noisy speech corrupted by white or colored noise.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.8 no.4
• /
• pp.555-560
• /
• 2013

This paper first proposes a classification algorithm which detects a voiced, unvoiced, and silence signal using distance measure, logarithm power and root mean square methods at each frame, then a spectrum subtraction algorithm based on a subband filter. The proposed algorithm subtracts spectrums of white noise and street noise from noisy signal based on the subband filter at each frame. In this experiment, experimental results of the proposed spectrum subtraction algorithm demonstrate using the speech and noise data of Aurora-2 database. Based on measuring the speech-to-noise ratio (SNR), experiments confirm that the proposed algorithm is effective for the speech by contaminated the noise. From the experiments, the improvement in the output SNR values was approximately 2.1 dB and 1.91 dB better for white noise and street noise, respectively.

• The Journal of the Acoustical Society of Korea
• /
• v.29 no.4
• /
• pp.270-277
• /
• 2010

In this paper, we proposed an algorithm that improves tone quality of noisy audio signals by using ICA(Independent Component Analysis) algorithm and perceptual filters. Many algorithms have been proposed to eliminate the noise from the audio signals, such as spectral subtraction method, perceptual filter, etc. The perceptual filter uses a noise that is acquired from silent ranges in the input signal. In this case, the improvement rate of tone quality decreases if the noise energy is changed by the environmental variation in a signal frame. But the proposed method estimates a noise that is changed at each frame using ICA algorithm. The estimated noise is applied to perceptual filter. To show the performance of the proposed algorithm, several tests are performed to various input signals. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR), noise-to-mask ratio (NMR) and Degradation Category Rating (DCR) test.

• Journal of the Korean Institute of Telematics and Electronics A
• /
• v.32A no.9
• /
• pp.1257-1263
• /
• 1995

This paper presents a method not only to detect moving targets from ISAR data corrupted by heavy clutter but to effectively extract the moving targets' signitures that are necessary for acquiring the velocities and images of the targets. In this method, input ISAR data are coherently processed such that echoes due to the moving targets are transformed to be in-phase and then integrated. It is shown that targets' signitures are localized in the transformed two-dimensional domain, leading to dramatic improvement of signal-to-noise (SNR) ration. The extent of the improved signal-to-noise ratio is analyzed and results implemented with simulated ISAR data are presented.

• Proceedings of the KSPS conference
• /
• 2007.05a
• /
• pp.127-130
• /
• 2007

The conventional feature recombination technique is very effective in the band-limited noise condition, but in broad-band noise condition, the conventional feature recombination technique does not produce notable performance improvement compared with the full-band system. To cope with this drawback, we introduce a new technique of sub-band likelihood computation in the feature recombination, and propose a new feature recombination method by using this sub-band likelihood computation. Furthermore, the reliable sub-band selection based on the signal-to-noise ratio is used to improve the performance of this proposed feature recombination. Experimental results shows that the average error reduction rate in various noise condition is more than 27% compared with the conventional full-band speaker identification system.

• Proceedings of the Korean Information Science Society Conference
• /
• 1998.10c
• /
• pp.467-468
• /
• 1998

Impulsive noise appears as black and/or white spots in an image. It is usually caused by errors during the image acquisition or transmission through communication channels. This paper presents a study on the impulsive noise reduction filter of digital image. A much more effective method for removing impulse noise is weighted median filtering. But it loses some information by changing center value with no condition. We propose some new technique to change center value with some conditions. In this paper, the performance of conditional weighted median filter is compared to the commonly used median filter, mean filter, max/min filter, and weighted median filter. A quantitative comparison is performed on MSE (Mean Square Error), RMSE (Root Mean Square Error), and SNR (Signal to Noise Ratio). Proposed conditional weighted median filter can yield better performance than regular filters.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.9 no.2
• /
• pp.395-400
• /
• 2008

This paper relates to the performance analysis of DoA estimation algorithm in 2-dimensional circular array antenna for the receiving of GPS signal which is used for the performance improvement by elimination of jammer signal. By performing the spatial filtering after the DoA estimation in array antenna, the quality of receiving signal can improve by the nulling of jammer signal from the undesired direction and the forming of beam from the desired direction. In this paper, the MUSIC and MinNorm algorithm used for DoA estimation were applied after fixing the angle and power of jammer signal in 4 element and 7 element circular array antenna. In order to performance analysis, the estimation result and estimation error were computed by computer simulation. As a result, the MUSIC and MinNorm were fairly good in azimuth and elevation angle estimation of DoA in case of good signal to noise ratio and the MUSIC has better performance compared to MinNorm in case of poor signal to noise ratio.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
• /
• v.2
• /
• pp.399-403
• /
• 2006

Software-based GNSS receivers have the great advantage in flexibility compared with conventional receivers. But it has some problems to processing IF level Signal RAW data, need long time to process long term data and TTFF is long because the process is too slow. So this time, we concentrated on the signal acquisition, and examined the speedup technique. Using this technique, the acquisition was speedup dramatically, and signal-to-noise ratio was improved.

• Journal of Korean Society for Quality Management
• /
• v.23 no.2
• /
• pp.28-42
• /
• 1995

For quality improvement, Taguchi emphasizes the reduction of variation of the quality characteristic, Taguchi has used the signal-to-noise ratios for achieving minimum dispersion of the quality characteristic with its location adjusted to some desired target value ${\tau}$ Lim(1994) proposes a reasonable SN ratio based on a linking relationship of the variance and mean through simple data analysis technique. In this paper we investigate the efficiency of those two SN ratios and variance stabilizing transformations through Monte Carlo Simulations.