• Title/Summary/Keyword: Signal Processing Method

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An Analysis of Partial Discharge signal Using Wavelet Transforms (웨이블렛 변환을 이용한 부분 방전 신호 분석)

  • 박재준;장진강;임윤석;심종탁;김재환
    • Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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    • 1999.05a
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    • pp.169-172
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    • 1999
  • Recently, the wavelet transform has been a new and powerful tool for signal processing. It is more suitable specially for the feature extraction and detection of non-stationary signals than traditional methods such as, the Fourier Transform(FT), the Fast Fourier Transform(FFT) and the Least Square Method etc. because of the characteristic of the multi-scale analysis and time-frequency domain localization. The wavelet transform has been developed for the analysis of PD pulse signal to raise in the progress of insulation degradation. In this paper, the wavelet transform was applied to one foundational method for feature extraction. For the obtain experimental data, a computer-aided partial discharge measurement system with a single acoustic sensor was used. If we are applying to the neural network method the accumulated data through the extracted feature, it is expected that we can detect the PD pulse signal in the insulation materials on the on-line.

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Speech Signal Processing for Analysis of Chaos Pattern (카오스 패턴 발견을 위한 음성 데이터의 처리 기법)

  • Kim, Tae-Sik
    • Speech Sciences
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    • v.8 no.3
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    • pp.149-157
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    • 2001
  • Based on the chaos theory, a new method of presentation of speech signal has been presented in this paper. This new method can be used for pattern matching such as speaker recognition. The expressions of attractors are represented very well by the logistic maps that show the chaos phenomena. In the speaker recognition field, a speaker's vocal habit could be a very important matching parameter. The attractor configuration using change value of speech signal can be utilized to analyze the influence of voice undulations at a point on the vocal loudness scale to the next point. The attractors arranged by the method could be used in research fields of speech recognition because the attractors also contain unique information for each speaker.

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Parameter estimation of weak space-based ADS-B signals using genetic algorithm

  • Tao, Feng;Jun, Liang
    • ETRI Journal
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    • v.43 no.2
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    • pp.324-331
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    • 2021
  • Space-based automatic dependent surveillance-broadcast (ADS-B) is an important emerging augmentation of existing ground-based ADS-B systems. In this paper, the problem of space-based ultra-long-range reception processing of ADS-B signals is described. We first introduce a header detection method for accurately determining the pulse position of a weak ADS-B signal. We designed a signal encoding method, shaping method, and fitness function. We then employed a genetic algorithm to perform high-precision frequency and phase estimations of the detected weak signal. The advantage of this algorithm is that it can simultaneously estimate the frequency and phase, meaning a direct coherent demodulation can be implemented. To address the computational complexity of the genetic algorithm, we improved the ratio algorithm for frequency estimation and raised the accuracy beyond that of the original ratio algorithm with only a slight increase in the computational complexity using relatively few sampling points.

Acoustic Signal Processing for ADCP using Zoom FFT Method to increase Frequency Resolution (주파수 해상도 증가를 위해 Zoom FFT 기법을 사용한 ADCP 음향신호처리)

  • Han, Jin-Hyun;Shim, Tae-Bo
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.47 no.5
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    • pp.229-234
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    • 2010
  • This paper proposed the acoustic signal processing techniques, which are applicable even in the shallow river, and will enhance the frequency resolution of the ADCP (Acoustic Doppler Current profiler). ADCP is a device that measures the velocity of a moving fluid. ADCP, in general, can be operated at ~300 Khz of center frequency due to no depth limit in the sea. However, it can hardly be used due to water depth of 30cm or shallower during the dry season in the river. Therefore, existing signal processing methods are not suitable to use in the shallow river. We are proposing an alternative acoustic signal processing method using Zoom FFT. Simulation results show that errors are reduced ${\pm}62\;cm/s$ in theory, and ${\pm}93\;cm/s$ in the experiment. The existing algorithm could not estimate the current speed at the shallow river below 30 cm, but proposed algorithm estimated the current speed that was faster than 20 cm/s at the shallow river below 30 cm.

A Method on the Improvement of the Signal Processing Calculation Structure of the Remote Measurement Level Meter (원격 측정 레벨계의 신호처리 연산 구조 개선 방법)

  • Park, Dongkun;Lee, Kijun
    • Korean Journal of Remote Sensing
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    • v.35 no.3
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    • pp.389-400
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    • 2019
  • Level meters are non-invasively capable of measuring the level of the medium, and a growing variety of level meters are being used in the industry in connection with safety and maintenance. The level meter can be measured according to various kinds of medium such as solid medium such as coal, flour, rice and liquid medium such as water and petroleum. In order to reduce the error depending on the medium, the measurement using the Doppler Effect can compensate the measurement error, However, the number of signal processing steps is increased, the operation speed is further increased, the hardware complexity increases, and a high cost structure is required. In this paper, we propose a method to improve the signal processing operation structure of the remote measurement level meter to reduce the amount of computation and the resource usage of the required FPGA.

A High Precision Line Detection Based on Local Area CCT Method (국소영역 내의 CCT법을 이용한 고정밀 직선 검출)

  • Jung, Nam-Chae
    • Journal of the Institute of Convergence Signal Processing
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    • v.14 no.2
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    • pp.82-89
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    • 2013
  • A detection method of high precision digital line within image is proposed in this paper. If we set the size of image to $N{\times}N$, in fact it is difficult to use the resulting values that the amount of computation is $O(N^4)$. Multiple algorithms are examined to reduced the amount of computation to $O(N^3)$, while suppressing the degradation of precision. How to detect line from the image processing, after stretching treatment of line segments extracted by Hough transform in the local area of an image is a great way to be able to detect several long or short line at high speed, but this method is slightly less precision in the detection of tilted line segments. In this paper, a line detection method improving the precision detection of tilted line segment is applied to the local area, thereby this method does not reduce the processing speed, while it is high precision method for detecting line segments. The experimental results confirm that the proposed method can detect a high precision line in a shorter period of time, compared with the existing methods.

Minimum Statistics-Based Noise Power Estimation for Parametric Image Restoration

  • Yoo, Yoonjong;Shin, Jeongho;Paik, Joonki
    • IEIE Transactions on Smart Processing and Computing
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    • v.3 no.2
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    • pp.41-51
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    • 2014
  • This paper describes a method to estimate the noise power using the minimum statistics approach, which was originally proposed for audio processing. The proposed minimum statistics-based method separates a noisy image into multiple frequency bands using the three-level discrete wavelet transform. By assuming that the output of the high-pass filter contains both signal detail and noise, the proposed algorithm extracts the region of pure noise from the high frequency band using an appropriate threshold. The region of pure noise, which is free from the signal detail part and the DC component, is well suited for minimum statistics condition, where the noise power can be extracted easily. The proposed algorithm reduces the computational load significantly through the use of a simple processing architecture without iteration with an estimation accuracy greater than 90% for strong noise at 0 to 40dB SNR of the input image. Furthermore, the well restored image can be obtained using the estimated noise power information in parametric image restoration algorithms, such as the classical parametric Wiener or ForWaRD image restoration filters. The experimental results show that the proposed algorithm can estimate the noise power accurately, and is particularly suitable for fast, low-cost image restoration or enhancement applications.

Implementation of EEG Artifact Removal Process Based on Bispectrum Analysis (바이스펙트럼 분석 기반의 뇌파 Artifact 제거 프로세스 구현)

  • Park, Junmo
    • Journal of the Institute of Convergence Signal Processing
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    • v.20 no.2
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    • pp.63-69
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    • 2019
  • In this study, bispectrum analysis method introduced to reduce variability of SEF(spectral edge frequency) and MF(median frequency), which are the anesthetic depth indexes extracted by EEG spectral analysis. Bispectrum analysis is an analytical method that can confirm the nonlinearity of EEG. Signal measurement and analysis in the surgical environment should take into consideration various external artifact factors. Bispectrum analysis can confirm the presence of externally introduced artifacts, thereby effectively eliminating artifacts that affect the EEG signal. By applying bispectrum parameters, real-time variability of the anesthetic depth parameters SEF, MF could be reduced. Elimination of variability makes it possible to use SEF, MF as a real-time index during surgery.

Implementation of Digital Signal Processing Board Suitable for a Semi-active Laser Tracking to Detect a Laser Pulse Repetition Frequency and Optimization of a Target Coordinates (반능동형 레이저 유도 추적에 적합한 레이저 펄스 반복 주파수 검출을 위한 디지털 신호처리 보드 구현 및 표적 좌표 최적화)

  • Lee, Young-Ju;Kim, Yong-Pyung
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.64 no.4
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    • pp.573-577
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    • 2015
  • In this paper, we propose a signal processing board suitable for a semi-active laser tracking to detect an optical signal generated from the laser target designator by applying an analog trigger signal, the quadrant photodetector and a high speed ADC(analog-digital converter) sampling technique. We improved the stability by applying the averaging method to minimize the measurement error of a gaussian pulse. To evaluate the performances of the proposed methods, we implemented a prototype board and performed experiments. As a result, we implemented a frequency counter with an error 14.9ns in 50ms. PRF error code has a stability of less than 1.5% compared to the NATO standard. Applying the three point averaging method to ADC sampling, the stability of 28% in X-axis and 22% in Y-axis than one point sampling was improved.

DCT and DWT Based Robust Audio Watermarking Scheme for Copyright Protection

  • Deb, Kaushik;Rahman, Md. Ashikur;Sultana, Kazi Zakia;Sarker, Md. Iqbal Hasan;Chong, Ui-Pil
    • Journal of the Institute of Convergence Signal Processing
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    • v.15 no.1
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    • pp.1-8
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    • 2014
  • Digital watermarking techniques are attracting attention as a proper solution to protect copyright for multimedia data. This paper proposes a new audio watermarking method based on Discrete Cosine Transformation (DCT) and Discrete Wavelet Transformation (DWT) for copyright protection. In our proposed watermarking method, the original audio is transformed into DCT domain and divided into two parts. Synchronization code is applied on the signal in first part and 2 levels DWT domain is applied on the signal in second part. The absolute value of DWT coefficient is divided into arbitrary number of segments and calculates the energy of each segment and middle peak. Watermarks are then embedded into each middle peak. Watermarks are extracted by performing the inverse operation of watermark embedding process. Experimental results show that the hidden watermark data is robust to re-sampling, low-pass filtering, re-quantization, MP3 compression, cropping, echo addition, delay, and pitch shifting, amplitude change. Performance analysis of the proposed scheme shows low error probability rates.