• Title/Summary/Keyword: 단구간 푸리에 변환

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Spatial - Frequency Analysis of time-varying Coherence using ERP signals for attentional visual stimulus (시각 자극의 집중에 따른 시간 변화에 대한 뇌 유발전위의 공간 - 주파수간 상관 변화 분석)

  • Lee, ByuckJin;Yoo, Sun-Kook
    • Science of Emotion and Sensibility
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    • v.16 no.4
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    • pp.527-534
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    • 2013
  • In this study, we analyzed spatial-frequency relationship related brain function for change of the time during attentional visual stimulus through the analysis of Coherence. With experimentation about ERP(Event Related Potential)data, it revealed that change of the phase synchronization between different scalp locations at ${\theta}$, ${\alpha}$ band. ERP between left and right frontal lobes, between the frontal and central lobes showed the phase synchronization at the P100, N200, ERP between the frontal and occipital lobes showed the phase synchronization at the P300 related information of visual stimulus. Compared to STFT using the window of a fixed length, CWT is able to multi-resolution analysis with the adjustment of parameters of mother wavelet. Thus, coherence results with CWT was found to be effective for analysis of time-varying spatial-frequency relationship in ERP. The phase synchronization for inattentional visual stimulus was not observed.

Speech Recognition Model Based on CNN using Spectrogram (스펙트로그램을 이용한 CNN 음성인식 모델)

  • Won-Seog Jeong;Haeng-Woo Lee
    • The Journal of the Korea institute of electronic communication sciences
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    • v.19 no.4
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    • pp.685-692
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    • 2024
  • In this paper, we propose a new CNN model to improve the recognition performance of command voice signals. This method obtains a spectrogram image after performing a short-time Fourier transform (STFT) of the input signal and improves command recognition performance through supervised learning using a CNN model. After Fourier transforming the input signal for each short-time section, a spectrogram image is obtained and multi-classification learning is performed using a CNN deep learning model. This effectively classifies commands by converting the time domain voice signal to the frequency domain to express the characteristics well and performing deep learning training using the spectrogram image for the conversion parameters. To verify the performance of the speech recognition system proposed in this study, a simulation program using Tensorflow and Keras libraries was created and a simulation experiment was performed. As a result of the experiment, it was confirmed that an accuracy of 92.5% could be obtained using the proposed deep learning algorithm.

Fault Diagnosis for Rotating Machinery with Clearance using HHT (HHT를 이용한 간극이 있는 회전체의 고장진단)

  • Lee, Seung-Mock;Choi, Yeon-Sun
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2007.11a
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    • pp.895-902
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    • 2007
  • Rotating machinery has two typical faults with clearance, one is partial rub and the other is looseness. Due to these faults, non-linear and non-stationary signals are occurred. Therefore, time-frequency analysis is necessary for exact fault diagnosis of rotating machinery. In this paper newly developed time-frequency analysis method, HHT(Hilbert-Huang Transform) is applied to fault diagnosis and compared with other method of FFT, SFFT and CWT. The results show that HHT can represent better resolution than any other method. Consequently, the faults of rotating machinery are diagnosed efficiently by using HHT.

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Design of the Noise Suppressor Using Wavelet Transform (웨이블릿 변환을 이용한 잡음제거기 설계)

  • 원호진;김종학;이인성
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.7
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    • pp.37-46
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    • 2001
  • This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.

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Applications of the improved Hilbert-Huang transform method to the detection of thermo-acoustic instabilities (열음향학적 불안정성 검출에 대한 개선된 힐버트-후앙 변환의 적용)

  • Cha, Ji-Hyeong;Kim, Young-Seok;Ko, Sang-Ho
    • Proceedings of the Korean Society of Propulsion Engineers Conference
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    • 2012.05a
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    • pp.555-561
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    • 2012
  • The Hilbert Huang Transform (HHT) technigue with Empirical Mode Decomposition (EMD) is one of the time-frequency domain analysis methods and it has several advantages such that analyzing non-stationary and nonlinear signal is possible. However, there are shortcomings in detecting near-range of frequencies and added noise signals. In this paper, to analyze characteristics of each method, HHT and Short-Time Fourier Transform (STFT) effective in dealing with stationary signals are compared. And with thermoacoustic instabilities signals from a Rijke tube test, HHT and the improved HHT with Ensemble Empirical Mode Decomposition (EEMD) are compared. The results show that the improved HHT is more appropriate than the original HHT due to the relative insensitivity to noise. Therefore it will result in more accurate analysis.

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Optimizing Wavelet in Noise Canceler by Deep Learning Based on DWT (DWT 기반 딥러닝 잡음소거기에서 웨이블릿 최적화)

  • Won-Seog Jeong;Haeng-Woo Lee
    • 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.

Audio signal clustering and separation using a stacked autoencoder (복층 자기부호화기를 이용한 음향 신호 군집화 및 분리)

  • Jang, Gil-Jin
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.4
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    • pp.303-309
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    • 2016
  • This paper proposes a novel approach to the problem of audio signal clustering using a stacked autoencoder. The proposed stacked autoencoder learns an efficient representation for the input signal, enables clustering constituent signals with similar characteristics, and therefore the original sources can be separated based on the clustering results. STFT (Short-Time Fourier Transform) is performed to extract time-frequency spectrum, and rectangular windows at all the possible locations are used as input values to the autoencoder. The outputs at the middle, encoding layer, are used to cluster the rectangular windows and the original sources are separated by the Wiener filters derived from the clustering results. Source separation experiments were carried out in comparison to the conventional NMF (Non-negative Matrix Factorization), and the estimated sources by the proposed method well represent the characteristics of the orignal sources as shown in the time-frequency representation.