• Nuclear Engineering and Technology
• /
• 제56권2호
• /
• pp.715-727
• /
• 2024

During fast neutron imaging, besides the dark current noise and readout noise of the CCD camera, the main noise in fast neutron imaging comes from high-energy gamma rays generated by neutron nuclear reactions in and around the experimental setup. These high-energy gamma rays result in the presence of high-density gamma white spots (GWS) in the fast neutron image. Due to the microscopic quantum characteristics of the neutron beam itself and environmental scattering effects, fast neutron images typically exhibit a mixture of Gaussian noise. Existing denoising methods in neutron images are difficult to handle when dealing with a mixture of GWS and Gaussian noise. Herein we put forward a deep learning approach based on the Swin Transformer UNet (SUNet) model to remove high-density GWS-Gaussian mixture noise from fast neutron images. The improved denoising model utilizes a customized loss function for training, which combines perceptual loss and mean squared error loss to avoid grid-like artifacts caused by using a single perceptual loss. To address the high cost of acquiring real fast neutron images, this study introduces Monte Carlo method to simulate noise data with GWS characteristics by computing the interaction between gamma rays and sensors based on the principle of GWS generation. Ultimately, the experimental scenarios involving simulated neutron noise images and real fast neutron images demonstrate that the proposed method not only improves the quality and signal-to-noise ratio of fast neutron images but also preserves the details of the original images during denoising.

• 대한후두음성언어의학회지
• /
• 제28권2호
• /
• pp.100-105
• /
• 2017

Background and Objectives : Adductor type spasmodic dysphonia (ADSD) is neurogenic disorder and focal laryngeal dystonia, while muscle tension dysphonia (MTD) is caused by functional voice disorder. Both ADSD and MTD may be associated with excessive supraglottic contraction and compensation, resulting in a strained voice quality with spastic voice breaks. The aim of this study was to determine the utility of spectrogram analysis in the differentiation of ADSD from MTD. Materials and Methods : From 2015 through 2017, 17 patients of ADSD and 20 of MTD, underwent acoustic recording and phonatory function studies, were enrolled. Jitter (frequency perturbation), Shimmer (amplitude perturbation) were obtained using MDVP (Multi-dimensional Voice Program) and GRBAS scale was used for perceptual evaluation. The two speech therapist evaluated a wide band (11,250 Hz) spectrogram by blind test using 4 scales (0-3 point) for four spectral findings, abrupt voice breaks, irregular wide spaced vertical striations, well defined formants and high frequency spectral noise. Results : Jitter, Shimmer and GRBAS were not found different between two groups with no significant correlation (p>0.05). Abrupt voice breaks and irregular wide spaced vertical striations of ADSD were significantly higher than those of MTD with strong correlation (p<0.01). High frequency spectral noise of MTD were higher than those of ADSD with strong correlation (p<0.01). Well defined formants were not found different between two groups. Conclusion : The wide band spectrograms provided visual perceptual information can differentiate ADSD from MTD. Spectrogram analysis is a useful diagnostic tool for differentiating ADSD from MTD where perceptual analysis and clinical evaluation alone are insufficient.

• 임상이비인후과
• /
• 제29권2호
• /
• pp.212-222
• /
• 2018

Background and Objectives : The purpose of this study was to compare the measured values of acoustic and auditory perceptual assessments between normal and functional dysphonia (FD) groups. Materials and Methods : 102 subjects with FD and 59 normal voice groups were participated in this study. Mid-vowel portion of the sustained vowel /a/ and two sentences of 'Sanchaek' were edited, concatenated, and analyzed by Praat script. And then auditory-perceptual (AP) rating was completed by three listeners. Results : The FD group showed higher acoustic voice quality index version 2.02 and version 3.01 (AVQIv2 and AVQIv3), slope, Hammarberg index (HAM), grade (G) and overall severity (OS), values than normal group. Additionally, smoothed cepstral peak prominence in Praat (PraatCPPS), tilt, low-to high spectral band energies (L/H ratio), long-term average spectrum (LTAS) in FD group were lower than normal voice group. And the correlation among measured values ranged from -0.250 to 0.960. In ROC curve analysis, cutoff values of AVQIv2, AVQIv3, PraatCPPS, slope, tilt, L/H ratio, HAM, and LTAS were 3.270, 2.013, 13.838, -22.286, -9.754, 369.043, 27.912, and 34.523, respectively, and the AUC of each analysis was over .890 in AVQIv2, AVQIv3, and PraatCPPS, over 0.731 in HAM, tilt, and slope, over 0.605 in LTAS and L/H ratio. Conclusions : In conclusion, AVQI and CPPS showed the highest predictive power for distinguishing between normal and FD groups. Acoustic analyses and AP rating as noninvasive examination can reinforce the screening capability of FD and help to establish efficient diagnosis and treatment process plan for FD.

• 한국통신학회논문지
• /
• 제40권6호
• /
• pp.1175-1184
• /
• 2015

3D 영상의 획득을 위해 이용하는 깊이 지도는 영상의 깊이 정보를 가지고 있다. 그러나 일반적으로 고해상도이며 잡음이 제거된 고화질의 깊이 지도를 카메라로부터 직접 획득하는 것은 어렵다. 그래서 저해상도의 깊이 지도를 획득하고 업샘플링 및 전/후 영상처리를 통해 높은 품질의 고해상도 깊이 지도를 획득하는 기법들이 연구되고 있다. 하지만 기존의 연구는 영상의 질에 큰 변수로 작용하는 에지 부분의 효과적 업샘플링이 미흡하다. 그래서 본 논문은 에지 부분을 차별적으로 고려하는 인지적인 특성을 반영한 영상품질향상 연구에 초점을 맞춰 결합 양방향 필터의 가중치를 적응적으로 조절함으로써 깊이 지도와 합성 영상을 개선한 고해상도의 깊이 지도를 얻는 업샘플링 방법을 제안하였다. 제안 방식을 기존의 방식과 비교하였을 때 PSNR 측면과 주관적 품질에서 이득이 있음을 보였다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2007년도 추계학술대회논문집
• /
• pp.1037-1041
• /
• 2007

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral psycho-acoustic filters considering masking effects are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for realtime implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• 전기학회논문지
• /
• 제60권6호
• /
• pp.1196-1203
• /
• 2011

The reduced frequency selectivity is a typical phenomenon of sensorineural hearing loss. In this paper, we compared two modeling methods for reduced frequency selectivity of hearing impaired people. The two models of reduced frequency selectivity were made using LPC(linear prediction coding) algorithm and bandwidth control algorithm based on ERB(equivalent rectangular bandwidth) of auditory filter, respectively. To compare the effectiveness of two models, we compared the result of PESQ (perceptual evaluation of speech quality) and LLR(log likelihood ratio) using 36 Korean words of two syllables. To verify the effect on noise condition, we mixed white and babble noise with 0dB and -3dB SNR to speech words. As the result, it is confirmed that the PESQ score of bandwidth control algorithm is higher than the score of LPC algorithm, on the other hands, and the LLR score of LPC algorithm is lower than the score of bandwidth control algorithm. It means that both non-linearity and widen auditory filter characteristics caused by reduced frequency selectivity could be more reflected in bandwidth control algorithm than in LPC algorithm.

• IEIE Transactions on Smart Processing and Computing
• /
• 제5권2호
• /
• pp.129-142
• /
• 2016

In Wyner-Ziv coding, video signals are reconstructed by correcting side information generated by block-based motion estimation/compensation at the decoder. The correction is not always accurate due to the limited number of parity bits and early stopping of low-density parity check accumulate (LDPCA) decoding in distributed video coding, or due to the limited number of measurements in distributed compressive video sensing. The blocking artifacts caused by block-based processing are usually conspicuous in smooth areas and degrade the perceptual quality of the reconstructed video. Conventional deblocking filters try to remove the artifacts by treating both sides of the block boundary equally; however, coding errors generated by block-based processing are not necessarily the same on both sides of the block boundaries. Such a block-wise difference is exploited in this paper to improve deblocking for Wyner-Ziv frameworks by designing a filter where the deblocking strength at each block can be non-identical, depending on the reliability of the reconstructed pixels. Test results show that the proposed filter not only improves subjective quality by reducing the coding artifacts considerably, but also gains rate distortion performance.

• ETRI Journal
• /
• 제31권4호
• /
• pp.408-418
• /
• 2009

Adopting an encryption function in voice over Wi-Fi service incurs problems such as additional power consumption and degradation of communication quality. To overcome these problems, a partial encryption (PE) algorithm for compressed speech was recently introduced. However, from the security point of view, the partial encryption sets (PESs) of the conventional PE algorithm still have much room for improvement. This paper proposes a new selection method for finding a smaller PES while maintaining the security level of encrypted speech. The proposed PES selection method employs the perceptual evaluation of the speech quality (PESQ) algorithm to objectively measure the distortion of speech. The proposed method is applied to the ITU-T G.729 speech codec, and content protection capability is verified by a range of tests and a reconstruction attack. The experimental results show that encrypting only 20% of the compressed bitstream is sufficient to effectively hide the entire content of speech.

• 대한전자공학회논문지SP
• /
• 제40권6호
• /
• pp.124-131
• /
• 2003

본 논문에서는 최근 비디오 스트리밍 응용을 위해 만들어진 FGS((me granular scalability) 방식에서 주어진 비트율에서 인간 시각에 좀더 나은 화질을 얻기 위해 주파수 가중 방법을 제안한다. 제안된 방식에서는 각 DCT 계수의 사람 눈에 미치는 중요도를 고려하여 구해진 주파수 가중행렬을 비트평면 부호화를 사용하는 FGS 상향계층 부호화 동작에 적용될 수 있도록 주파수 천이 행렬로 바꾸어서 가중치를 부여한다. 본 논문에서 제안된 주파수 가중 방법을 통해 화질의 향상, 더 미세한 계위 부호화, 시간 영역에서 화질의 균일화를 얻을 수 있었다. 또한, 주파수 가중 방법을 통해 얻은 화질 개선을 설명하기 위해 기존에 사용하던 PSNR(peak signal to noise ratio) 대신에 인간 시각적인 관점에서의 오차만을 고려한 JNDE(just noticeable difference ewer)를 Weber의 법칙을 근거로 하여 새롭게 제안하였다.

• 한국소음진동공학회논문집
• /
• 제18권2호
• /
• pp.231-237
• /
• 2008

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral real-time masking blocks are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for real-time implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.