• The Korean Journal of Nuclear Medicine Technology
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• v.16 no.1
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• pp.34-37
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• 2012

Purpose: To verify the optimal scan time per bed for clinical application, we evaluated the quality of $^{18}F$-FDG images with varying scan times in a phantom and 20 patients with 38 lesions using a Philips (TOF) PET/CT scanner. Materials and Methods: The PET/CT images of a NEMA IEC body phantom and 20 patients (16 males, 4 females) were acquired for 5 different scan times of 20-100 sec per bed with intervals of 20 sec. The activity ratio of hot spheres (diameter of 17 [H1], 22 [H2] and 28 [H3] mm) to the background region in the IEC body phantom was 8-to-1. The contrast recovery coefficient (CRC) and standard uptake value (SUV) based on ROIs of hot spheres and background region were calculated. The noise in each background region was estimated as the ratio of SD of counts to the mean counts in the background region. On the patient image, the injected dose of $^{18}F$-FDG was $444{\pm}74$ MBq and the SUVs in the 38 hot lesions were measured. Results: The two scan time groups (LT-60 [<60 sec] and GT-60 [${\geq}60$ sec]) were compared. In the phantom study, the coefficient of deviations (CVs, %) of CRC and SUV in LT-60 (H1: 14.2 and 7.3, H2: 11.4 and 7.8, H3: 4.9 and 3.2) were higher than GT-60 (H1: 8.9 and 2.8, H1: 8.2 and 5.0, H3: 2.0 and 1.6). In the patient study, the mean CV of CRC and SUV in LT-60 (4.0) was higher than GT-60 (1.2). Conclusion: This study showed that noise increased as the scan time decreased. High noise for the scan time <60 sec per bed yielded high variation of SUV and CRC. Therefore, considering PET/CT image quality, the scan time per bed in the TOF PET/CT scanner should be at least ${\geq}60$ sec.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.10 s.103
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• pp.1160-1168
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• 2005

Floor impact noise isolation performance of reinforced concrete floors was investigated through new measurement method using impact bail. Strong impact force in Bow frequency band below 63 Hz of bang machine is not similar to human impact source and causes some problem in evaluating heavy-weight Impact noise but heavy-weight impact noise measurement and evaluation using impact ball which is very similar to human impact is more reliable than bang machine. Correction value on the background noise and sensitivity of residents should be considered on the floor impact noise evaluation classes.

• Journal of the Korean earth science society
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• v.37 no.1
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• pp.40-51
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• 2016

The problem has been pointed out that the domestic design response spectrum does not reflect site amplification, particularly in the high frequency bands, including the fact that site specific response spectrum from the observed ground motions appears relatively higher than design response spectrum. Among various methods, this study applied H/V spectral ratio of ground motion for estimating site amplification. This method, originated from S waves and Rayleigh waves, recently has been extended to Coda waves and background noise for estimating site amplification. For limited time of periods, 4 electric substation sites had operated seismic stations at two separate locations (bedrock and borehole) within each substation site. H/V spectral ratio of S wave, Coda wave, and background noise, was applied to 36 accelerations of 3 macro earthquakes (Odaesan, Jeju and Gongju earthquakes), larger than magnitude 3.4. observed simultaneously at each bedrock location within 4 electric substation sites. Site amplifications at the bedrock location of 4 sites were compared among S wave, Coda wave energy, and background noise, and then compared to the previous results from the borehole location data. The site classification was also tried using resonancy frequency information at each site and location. The results suggested that all the electric substation sites showed similar site amplification patterns among S wave, Coda wave, and background noise. Each station showed its own characteristics of site amplification property in low, high and specific resonance frequency ranges. Comparison of this study to other results using different method can give us much more information about dynamic amplification of domestic sites characteristics and site classification.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2
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• pp.76-85
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• 1994

A speech enhancement algorithm which attenuates nonstationary colored noise is presented In this paper. The algorithm consists of a stationary Kalman filter and the simple speech/nonspeech detector. While the conventional enhancement systems are focused on a stationary and/or white background noise, this study Is focused on the mort realistic nonstationary and nonwhite noise. An AR model-based vector Kalman filter is used as a noise suppression system and a short-time energy threshold logic is used as a speech/nonspeech classifier. For Kalman filtering. noise coefficients are estimated in the nonspeech frame, and speech coefficients are estimated by applying the EM iteration algorithm. Simulation results using the car noise are presented based on the signal-to-noise ratio and informal listening tests. According to the experimental results, background noises in the nonspeech frames are eliminated almost completely, while some distortions are noticed in the speech frames. The distortion becomes severer as the SNR is reduced to 0dB and -5dB. Intelligibility, however, is not degraded significantly.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.379-385
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• 2008

In this paper, we propose an effective speech reinforcement technique under the near-end and the far-end noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. Specifically, based on the estimated background noise spectrum of the near-end, we reinforce the far-end speech spectrum by incorporating the more general cases under the near-end with background noise. Also, we propose the novel approach to reinforce the actual speech signal except for the noise signal in the far-end noisy speech signal. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with the conventional method.

• The Journal of the Acoustical Society of Korea
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• v.25 no.2
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• pp.78-87
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• 2006

In this paper. we propose an effective mask estimation scheme for missing-feature reconstruction in order to achieve robust speech recognition under unknown noise environments. In the previous work. colored noise is used for training the mask classifer, which is generated from the entire frequency Partitioned signals. However it gives a limited performance under the restricted number of training database. To reflect the spectral events of more various background noise and improve the performance simultaneously. a new Bayesian classifier for mask estimation is proposed, which works independent of other frequency bands. In the proposed method, we employ the colored noise which is obtained by combining colored noises generated from each frequency band in order to reflect more various noise environments and mitigate the 'sparse' database problem. Combined with the cluster-based missing-feature reconstruction. the performance of the proposed method is evaluated on a task of noisy speech recognition. The results show that the proposed method has improved performance compared to the Previous method under white noise. car noise and background music conditions.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.402-409
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• 2010

In this paper, we propose an efficient integrated suppression algorithm based on combined power of acoustic echo and background noise. The proposed method combines the acoustic echo and noise power by the weighting parameter derived from the decision rule based on the estimated echo to noise power ratio. Therefore, in the proposed approach, the acoustic echo and noise signal are able to be reduced through only one suppression filter based on the estimated combined power. The proposed unified structure improves the problems of the residual echo and noise resulted from the conventional unified structure where the noise suppression (NS) operation is placed after the acoustic echo suppression (AES) algorithm or vice versa. The performance of the proposed algorithm is evaluated by the objective test under various environments and yields better results compared with the conventional scheme.

• Journal of Electrical Engineering and Technology
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• v.11 no.6
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• pp.1793-1801
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• 2016

Speech enhancement is a challenging problem due to the diversity of noise sources and their effects in different applications. The goal of speech enhancement is to improve the quality and intelligibility of speech by reducing noise. Many research works in speech enhancement have been accomplished in English and other European Languages. There has been limited or no such works or efforts in the past in the context of Tamil speech enhancement in the literature. The aim of the proposed method is to reduce the background noise present in the Tamil speech signal by using wavelets. New modified thresholding function is introduced. The proposed method is evaluated on several speakers and under various noise conditions including White Gaussian noise, Babble noise and Car noise. The Signal to Noise Ratio (SNR), Mean Square Error (MSE) and Mean Opinion Score (MOS) results show that the proposed thresholding function improves the speech enhancement compared to the conventional hard and soft thresholding methods.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1183-1186
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• 1998

Adaptive nois canceller can extract the noiseremoved spech in noisy speech signal by adapting the filter-coefficients to the background noise environment. A kind of LMS algorithm is one of the most popular adaptive algorithm for noise cancellation due to low complexity, good numerical property and the merit of easy implementation. However there is the matter of increasing misadjustment at voiced speech signal. Therefore the demanded speech signal may be extracted. In this paper, we propose a fast and noise robust wavelet packet adaptive noise canceller with NLMS-SUM method LMS combined algorithm. That is, we decompose the frequency of noisy speech signal at the base of the proposed analysis tree structure. NLMS algorithm in low frequency band can efficiently dliminate the effect of the low frequency noise and SUM method LMS algorithm at each high frequency band can remove the high frequency nosie. The proposed wavelet packet adaptive noise canceller is enhanced the more in SNR and according to Itakura-Satio(IS) distance, it is closer to the clean speech signal than any other previous adaptive noise canceller.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.334-337
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• 2005

The speech intelligibility is one of the most important factors in communication system. The speech intelligibility is related with speech to noise ratio. To enhance the speech to noise ratio, background noise reduction techniques are being developed. As a part of solution to noise reduction, this paper introduces directional microphone using beamforming method and speech filtering method. The directional microphone narrows the spatial range of processing signal into the direction of the target speech signal. The noise signal located in the same direction with speech still remains in the processing signal. To sort this mixed signal into speech and noise, as a following step, a speech-filtering method is applied to pick up only the speech signal from the processed signal. The speech filtering method is based on the characteristics of speech signal itself. The combined directional microphone and speech filtering method gives enhanced performance to speech intelligibility in communication system.