• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.987-994
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• 2020

In this paper, we studied the effect of kernel size of CNN layer on performance in acoustic noise attenuators. This system uses a deep learning algorithm using a neural network adaptive prediction filter instead of using the existing adaptive filter. Speech is estimated from a single input speech signal containing noise using a 100-neuron, 16-filter CNN filter and an error back propagation algorithm. This is to use the quasi-periodic property in the voiced sound section of the voice signal. In this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed to verify the performance of the noise attenuator for the kernel size. As a result of the simulation, when the kernel size is about 16, the MSE and MAE values are the smallest, and when the size is smaller or larger than 16, the MSE and MAE values increase. It can be seen that in the case of an speech signal, the features can be best captured when the kernel size is about 16.

• Journal of the Institute of Convergence Signal Processing
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• v.21 no.3
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• pp.121-126
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• 2020

Sound-based machine fault diagnosis is the automatic detection of abnormal sound in the acoustic emission signals of the machines. Conventional methods of using mathematical models were difficult to diagnose machine failure due to the complexity of the industry machinery system and the existence of nonlinear factors such as noises. Therefore, we want to solve the problem of machine fault diagnosis as a deep learning-based image classification problem. In the paper, we propose a CNN-based automatic machine fault diagnosis method using Spectrogram images. The proposed method uses STFT to effectively extract feature vectors from frequencies generated by machine defects, and the feature vectors detected by STFT were converted into spectrogram images and classified by CNN by machine status. The results show that the proposed method can be effectively used not only to detect defects but also to various automatic diagnosis system based on sound.

• 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.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.1
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• pp.71-76
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• 2023

This paper studied a method of attenuating nonlinear noise using a Wiener filter of a neural network structure in an acoustic noise attenuator. This system improves nonlinear noise attenuation performance with a deep learning algorithm using a neural network Wiener filter instead of using a conventional adaptive filter. A voice is estimated from a single input voice signal containing nonlinear noise using a 128-neuron, 8-neuron hidden layer and an error back propagation algorithm. In this study, a simulation program using the Keras library was written and a simulation was performed to verify the attenuation performance for nonlinear noise. As a result of the simulation, it can be seen that the noise attenuation performance of this system is significantly improved when the FNN filter is used instead of the Wiener filter even when nonlinear noise is included. This is because the complex structure of the FNN filter expresses any type of nonlinear characteristics well.

• Proceedings of the Korea Water Resources Association Conference
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• 2022.05a
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• pp.267-267
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• 2022

댐 운영에 있어서 필요한 수문자료는 강수량, 수위, 유량, 저수량 자료 등이 있다. 이중 저수량은 주로 댐수위-저수용량 곡선식을 이용하여 계산한다. 댐수위-저수용량 곡선식은 댐 부근에서 계측 되는 한 개의 수위자료를 이용하여 저수용량을 산정하며, 이는 큰 저수지 면적과 저수지 수면이 일정하지 않다는 것을 고려할 때 큰 오차가 발생할 수 있다. 본 연구에서는 음향 도플러 유속계 ADCP(Acoustic Doppler Current Profiler) 이용하여 보성강댐 저수지 수면경사를 실측하고, 동시에 실시간 이동측위시스템인 RTK-GPS(Real Time Kinematic)를 이용하여 이를 검증하였다. ADCP는 유수의 흐름을 방해하지 않으면서 수중에 발사된 음파의 도플러 효과를 이용하여 유속, 유량 및 측량이 가능한 장비이며, RTK-GPS의 경우 정밀한 위치정보를 가지고 있는 기준국의 위상에 대한 보정치를 실시간으로 이용하여 오차가 ±0.03m 이하인 것으로 알려졌다. 보성강댐의 하류에서 ADCP와 RTK-GPS를 장착한 보트를 저수지 종방향으로 처음부터 끝까지 이동하여 약 7.5km 종단측량을 실시하였고 저수지 지형적 특성을 고려하여 약 700m마다 횡단측량을 실시하여 종방향뿐만 아니라 횡방향 수면차도 조사하였다. 그 결과 보성강댐의 상류로 갈수록 수면경사가 전체적으로 상승하는 경향을 보였지만 일부구간에서 수위가 하강하는 경우도 발생하였다. 이는 미약하지만 저수지 내에 흐름이 발생하고 이 흐름에 따른 통제가 변화되는 것과 중간에 유입되는 지류의 영향 등으로 구간별로 수면경사 차이가 발생하는 것으로 추정된다. 횡방향 수면차는 지류가 유입되는 일부구간에서 다소 차이를 보였지만 큰 영향을 없는 것으로 판단된다. 보성강댐 저수지 수면을 종방향 및 횡방향으로 실측한 결과 구간별로 차이를 보였으며 최대 EL. 126.60m, 최소 EL. 126.33m 나타났다. 댐 상류 부근의 수면높이 EL. 126.50m와 비교하면 +0.10m, -0.17m 차이를 보였으며 이는 저수량 산정에 큰 오차를 발생시킨다. 효과적인 댐 운영을 위해서는 유입량 및 유출량을 정확하게 산정하는 것도 필요하지만 저수량을 정확하게 파악하는 것 역시 필요하다. 저수량을 정확하게 산정하려면 수킬로미터가 넘는 저수지 크기를 고려하여 수면경사를 실시간으로 계측하는 등의 노력이 필요한 것으로 판단된다.

• Journal of Korean Library and Information Science Society
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• v.54 no.3
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• pp.129-163
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• 2023

The purpose of this study is to identify reading competency and its components according to the concept of reading competency in order to strengthen the reading competency of children and adolescents with disabilities, develop diagnostic questions, and provide basic data for the development of a reading competency diagnostic tool for children and adolescents with disabilities, Research methods include literature research, brainstorming, delphi survey, and preliminary research. As a result of the study, the components of the reading competency diagnostic tool are broadly divided into 2 areas (affective domain, environmental domain), 4 categories (reading motivation, reading attitude, human environment, and physical environment), and a total of 13 components in each of the 4 categories (Reading interest, reading value, reading recognition, reading expectations, reading habits, reading efficacy, reading immersion, reading anxiety (avoidance), home/family, school/teacher, peers, reading environment, media environment) and the corresponding questions. was developed. Based on these results, a direction for developing a reading competency diagnostic tool for children and adolescents with disabilities was presented.

• Smart Media Journal
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• v.12 no.9
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• pp.81-88
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• 2023

Recently, communication through online is increasing due to the spread of non-face-to-face services due to COVID-19. In non-face-to-face situations, the other person's opinions and emotions are recognized through modalities such as text, speech, and images. Currently, research on multimodal emotion recognition that combines various modalities is actively underway. Among them, emotion recognition using speech data is attracting attention as a means of understanding emotions through sound and language information, but most of the time, emotions are recognized using a single speech feature value. However, because a variety of emotions exist in a complex manner in a conversation, a method for recognizing multiple emotions is needed. Therefore, in this paper, we propose a multi-emotion regression model that extracts feature vectors after preprocessing speech data to recognize complex, inherent emotions and takes into account the passage of time.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.6
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• pp.1103-1108
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• 2023

In this paper, we propose a new algorithm for attenuating the background noises in acoustic signal. This algorithm improves the noise attenuation performance by using the FNN(: Full-connected Neural Network) deep learning algorithm instead of the existing adaptive filter after wavelet transform. After wavelet transforming the input signal for each short-time period, noise is removed from a single input audio signal containing noise by using a 1024-1024-512-neuron FNN deep learning model. This transforms the time-domain voice signal into the time-frequency domain so that the noise characteristics are well expressed, and effectively predicts voice in a noisy environment through supervised learning using the conversion parameter of the pure voice signal for the conversion parameter. In order to verify the performance of the noise reduction system proposed in this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed. As a result of the experiment, the proposed deep learning algorithm improved Mean Square Error (MSE) by 30% compared to the case of using the existing adaptive filter and by 20% compared to the case of using the STFT(: Short-Time Fourier Transform) transform effect was obtained.

• The Journal of the Korea institute of electronic communication sciences
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• v.19 no.1
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• pp.113-118
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• 2024

In this paper, we propose an optimal wavelet in a system for canceling background noise of acoustic signals. This system performed Discrete Wavelet Transform(DWT) instead of the existing Short Time Fourier Transform(STFT) and then improved noise cancellation performance through a deep learning process. DWT functions as a multi-resolution band-pass filter and obtains transformation parameters by time-shifting the parent wavelet at each level and using several wavelets whose sizes are scaled. Here, the noise cancellation performance of several wavelets was tested to select the most suitable mother wavelet for analyzing the speech. In this study, to verify the performance of the noise cancellation system for various wavelets, a simulation program using Tensorflow and Keras libraries was created and simulation experiments were performed for the four most commonly used wavelets. As a result of the experiment, the case of using Haar or Daubechies wavelets showed the best noise cancellation performance, and the mean square error(MSE) was significantly improved compared to the case of using other wavelets.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.37 no.2
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• pp.141-147
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• 2024

The measurement of strain under an electric field has been widely employed to comprehend the fundamental principles of electro-mechanical responses in ferroelectric, piezoelectric, and electrostrictive materials. In particular, understanding the strain properties of piezoelectric materials in response to electrical stimulation is crucial for researching and developing components such as piezoelectric actuators, acoustic devices, and ultrasonic generators. This tutorial paper introduces the components and operational principles of the linear variable differential transducer (LVDT), a widely used displacement measurement device in various industries. Additionally, we present the configuration of an experimental setup using LVDT to measure the strain characteristics of ferroelectric, piezoelectric, or electrostrictive materials under the application of an electric field. This paper includes simple measurement results and analyses obtained through the LVDT experimental setup, providing valuable information on research methods for the electro-mechanical interactions of various materials.