• Title/Summary/Keyword: Acoustic Performance

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An improved regularized particle filter for remaining useful life prediction in nuclear plant electric gate valves

  • Xu, Ren-yi;Wang, Hang;Peng, Min-jun;Liu, Yong-kuo
    • Nuclear Engineering and Technology
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    • v.54 no.6
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    • pp.2107-2119
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    • 2022
  • Accurate remaining useful life (RUL) prediction for critical components of nuclear power equipment is an important way to realize aging management of nuclear power equipment. The electric gate valve is one of the most safety-critical and widely distributed mechanical equipment in nuclear power installations. However, the electric gate valve's extended service in nuclear installations causes aging and degradation induced by crack propagation and leakages. Hence, it is necessary to develop a robust RUL prediction method to evaluate its operating state. Although the particle filter(PF) algorithm and its variants can deal with this nonlinear problem effectively, they suffer from severe particle degeneracy and depletion, which leads to its sub-optimal performance. In this study, we combined the whale algorithm with regularized particle filtering(RPF) to rationalize the particle distribution before resampling, so as to solve the problem of particle degradation, and for valve RUL prediction. The valve's crack propagation is studied using the RPF approach, which takes the Paris Law as a condition function. The crack growth is observed and updated using the root-mean-square (RMS) signal collected from the acoustic emission sensor. At the same time, the proposed method is compared with other optimization algorithms, such as particle swarm optimization algorithm, and verified by the realistic valve aging experimental data. The conclusion shows that the proposed method can effectively predict and analyze the typical valve degradation patterns.

A Novel Approach to COVID-19 Diagnosis Based on Mel Spectrogram Features and Artificial Intelligence Techniques

  • Alfaidi, Aseel;Alshahrani, Abdullah;Aljohani, Maha
    • International Journal of Computer Science & Network Security
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    • v.22 no.9
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    • pp.195-207
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    • 2022
  • COVID-19 has remained one of the most serious health crises in recent history, resulting in the tragic loss of lives and significant economic impacts on the entire world. The difficulty of controlling COVID-19 poses a threat to the global health sector. Considering that Artificial Intelligence (AI) has contributed to improving research methods and solving problems facing diverse fields of study, AI algorithms have also proven effective in disease detection and early diagnosis. Specifically, acoustic features offer a promising prospect for the early detection of respiratory diseases. Motivated by these observations, this study conceptualized a speech-based diagnostic model to aid in COVID-19 diagnosis. The proposed methodology uses speech signals from confirmed positive and negative cases of COVID-19 to extract features through the pre-trained Visual Geometry Group (VGG-16) model based on Mel spectrogram images. This is used in addition to the K-means algorithm that determines effective features, followed by a Genetic Algorithm-Support Vector Machine (GA-SVM) classifier to classify cases. The experimental findings indicate the proposed methodology's capability to classify COVID-19 and NOT COVID-19 of varying ages and speaking different languages, as demonstrated in the simulations. The proposed methodology depends on deep features, followed by the dimension reduction technique for features to detect COVID-19. As a result, it produces better and more consistent performance than handcrafted features used in previous studies.

An acoustic Doppler-based silent speech interface technology using generative adversarial networks (생성적 적대 신경망을 이용한 음향 도플러 기반 무 음성 대화기술)

  • Lee, Ki-Seung
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.2
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    • pp.161-168
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    • 2021
  • In this paper, a Silent Speech Interface (SSI) technology was proposed in which Doppler frequency shifts of the reflected signal were used to synthesize the speech signals when 40kHz ultrasonic signal was incident to speaker's mouth region. In SSI, the mapping rules from the features derived from non-speech signals to those from audible speech signals was constructed, the speech signals are synthesized from non-speech signals using the constructed mapping rules. The mapping rules were built by minimizing the overall errors between the estimated and true speech parameters in the conventional SSI methods. In the present study, the mapping rules were constructed so that the distribution of the estimated parameters is similar to that of the true parameters by using Generative Adversarial Networks (GAN). The experimental result using 60 Korean words showed that, both objectively and subjectively, the performance of the proposed method was superior to that of the conventional neural networks-based methods.

Blind Noise Separation Method of Convolutive Mixed Signals (컨볼루션 혼합신호의 암묵 잡음분리방법)

  • Lee, Haeng-Woo
    • The Journal of the Korea institute of electronic communication sciences
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    • v.17 no.3
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    • pp.409-416
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    • 2022
  • This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

Linear prediction analysis-based method for detecting snapping shrimp noise (선형 예측 분석 기반의 딱총 새우 잡음 검출 기법)

  • Jinuk Park;Jungpyo Hong
    • The Journal of the Acoustical Society of Korea
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    • v.42 no.3
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    • pp.262-269
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    • 2023
  • In this paper, we propose a Linear Prediction (LP) analysis-based feature for detecting Snapping Shrimp (SS) Noise (SSN) in underwater acoustic data. SS is a species that creates high amplitude signals in shallow, warm waters, and its frequent and loud sound is a major source of noise. The proposed feature takes advantage of the characteristic of SSN, which is sudden and rapidly disappearing, by using LP analysis to detect the exact noise interval and reduce the effects of SSN. The error between the predicted and measured value is large and results in effective SSN detection. To further improve performance, a constant false alarm rate detector is incorporated into the proposed feature. Our evaluation shows that the proposed methods outperform the state-of-the-art MultiLayer-Wavelet Packet Decomposition (ML-WPD) in terms of receiver operating characteristic curve and Area Under the Curve (AUC), with the LP analysis-based feature achieving a higher AUC by 0.12 on average and lower computational complexity.

Application of ray-based blind deconvolution to long-range acoustic communication in deep water (음선 기반 블라인드 디컨볼루션의 장거리 심해 환경으로의 적용)

  • Kim, Donghyeon;Park, Heejin;Kim, J.S.;Hahn, Joo Young
    • The Journal of the Acoustical Society of Korea
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    • v.41 no.2
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    • pp.242-253
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    • 2022
  • When the source waveform is unknown, the Green's function can be estimated by Ray-based Blind Deconvolution (RBD) based on the simple array signal processing. In previous papers, RBD was successfully demonstrated using simulation and experiments in shallow water environment. In this paper, we investigate the applicability of RBD for a long-range communication (e.g., 30 km, 60 km, and 90 km) in a deep water environment (1,000 m ~), using experimental data conducted in the east of Pohang, South Korea, in October 2018. Data results are presented to demonstrate Green's function estimation of a communication signal (2.2 kHz ~ 2.9 kHz) using a 16-element, 42-m long vertical array. The results show that the Green's function estimated from RBD is comparable to that of matched filter result. Additional communication performance at a maximum range of 90 km will be also presented.

Development of Offshore Construction ROV System applying Pneumatic Gripper (공압 gripper를 적용한 해양 건설 ROV 시스템 개발)

  • Park, Jihyun;Hwang, Yoseop
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.26 no.11
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    • pp.1697-1705
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    • 2022
  • The safety of marine construction workers and marine pollution problems are occurring due to large-scale offshore construction. In particular, underwater construction work in the sea has a higher risk than other work, so it is necessary to apply an unmanned alternative system that considers the safety of the workers. In this paper, the ROV system for offshore construction has been developed for underwater unmanned work. A monitoring system was developed for position control through the control of underwater propellants, pneumatic gripper, and monitoring of underwater work. As a result of the performance evaluation, the underwater movement speed of the ROV was evaluated to be 0.89 m/s, and it was confirmed that the maximum load of the pneumatic gripper was 80 kg. In addition, the network bandwidth required for underwater ROV control and underwater video streaming was evaluated to be more than 300Mbps, wired communication at 92.7 ~ 95.0Mbit/s at 205m, and wireless communication at 78.3 ~ 84.8Mbit/s.

Dialect classification based on the speed and the pause of speech utterances (발화 속도와 휴지 구간 길이를 사용한 방언 분류)

  • Jonghwan Na;Bowon Lee
    • Phonetics and Speech Sciences
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    • v.15 no.2
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    • pp.43-51
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    • 2023
  • In this paper, we propose an approach for dialect classification based on the speed and pause of speech utterances as well as the age and gender of the speakers. Dialect classification is one of the important techniques for speech analysis. For example, an accurate dialect classification model can potentially improve the performance of speaker or speech recognition. According to previous studies, research based on deep learning using Mel-Frequency Cepstral Coefficients (MFCC) features has been the dominant approach. We focus on the acoustic differences between regions and conduct dialect classification based on the extracted features derived from the differences. In this paper, we propose an approach of extracting underexplored additional features, namely the speed and the pauses of speech utterances along with the metadata including the age and the gender of the speakers. Experimental results show that our proposed approach results in higher accuracy, especially with the speech rate feature, compared to the method only using the MFCC features. The accuracy improved from 91.02% to 97.02% compared to the previous method that only used MFCC features, by incorporating all the proposed features in this paper.

Frequency Domain Double-Talk Detector Based on Gaussian Mixture Model (주파수 영역에서의 Gaussian Mixture Model 기반의 동시통화 검출 연구)

  • Lee, Kyu-Ho;Chang, Joon-Hyuk
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.4
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    • pp.401-407
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    • 2009
  • In this paper, we propose a novel method for the cross-correlation based double-talk detection (DTD), which employing the Gaussian Mixture Model (GMM) in the frequency domain. The proposed algorithm transforms the cross correlation coefficient used in the time domain into 16 channels in the frequency domain using the discrete fourier transform (DFT). The channels are then selected into seven feature vectors for GMM and we identify three different regions such as far-end, double-talk and near-end speech using the likelihood comparison based on those feature vectors. The presented DTD algorithm detects efficiently the double-talk regions without Voice Activity Detector which has been used in conventional cross correlation based double-talk detection. The performance of the proposed algorithm is evaluated under various conditions and yields better results compared with the conventional schemes. especially, show the robustness against detection errors resulting from the background noises or echo path change which one of the key issues in practical DTD.

Covariance-based source localization performance improvement for underwater ultra-short baseline systems (공분산 기반 수중 ultra-short baseline 시스템의 위치 추정 성능 개선 기법)

  • Sangman Han;Minhyuk Cha;Haklim Ko;Hojun Lee
    • The Journal of the Acoustical Society of Korea
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    • v.43 no.1
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    • pp.89-94
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    • 2024
  • Since Ultra-Short BaseLine (USBL) uses an array with narrow sensor spacing, precise synchronization is required to improve source localization performances. However, in the underwater environment, synchronization errors occur due to relatively strong noise and underwater acoustic channels such as multipath and Doppler, which deteriorates the source localization performances. This paper proposes a covariance-based synchronization compensation method to improve the source localization performances of the underwater USBL systems. The proposed method arranges the received signals through cross-correlation and calculates the covariance of the arranged signals. The synchronization error is related to the phase difference in the covariance. Thus, the phase difference is estimated as the covariance and compensated. Computer simulations demonstrate that the proposed method has better source localization performances than the conventional cross-correlation method.