• Proceedings of the KSME Conference
• /
• 2003.04a
• /
• pp.304-309
• /
• 2003

In this study, natural flaws in welding parts are classified using the signal pattern classification method. The storage digital oscilloscope including FFT function and enveloped waveform generator is used and the signal pattern recognition procedure is made up the digital signal processing, feature extraction, feature selection and classifier design. It is composed with and discussed using the distance classifier that is based on euclidean distance the empirical Bayesian classifier. Feature extraction is performed using the class-mean scatter criteria. The signal pattern classification method is applied to the signal pattern recognition of natural flaws.

• Transactions of the Korean Society of Machine Tool Engineers
• /
• v.16 no.6
• /
• pp.62-70
• /
• 2007

In this study, the natural flaws in welding parts are classified using the signal pattern classification method. The storage digital oscilloscope including FFT function and enveloped waveform generator is used and the signal pattern recognition procedure is made up the digital signal processing, feature extraction, feature selection and classifier design. It is composed with and discussed using the distance classifier that is based on euclidean distance the empirical Bayesian classifier. Feature extraction is performed using the class-mean scatter criteria. The signal pattern classification method is applied to the signal pattern recognition of natural flaws.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
• /
• v.2
• /
• pp.399-403
• /
• 2006

Software-based GNSS receivers have the great advantage in flexibility compared with conventional receivers. But it has some problems to processing IF level Signal RAW data, need long time to process long term data and TTFF is long because the process is too slow. So this time, we concentrated on the signal acquisition, and examined the speedup technique. Using this technique, the acquisition was speedup dramatically, and signal-to-noise ratio was improved.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.10 no.11
• /
• pp.3121-3129
• /
• 2009

Conventional marine radar systems utilize pulse radar which is capable of high-power transmissions and is effective for remote detection purposes. A pulse radar is most commonly used on medium or large vessels due to its expensive installation and maintenance costs. I propose the use of a Frequency Modulated Continuous Wave (FMCW) radar system operated at low-power and high-resolution instead of the conventional pulse-radar based system. The transmitted and received signals of the FMCW radar system were theoretically analyzed and radar signal processing design and simulation experiments were performed to detect the range and speed. Intermediate Frequency (IF) signal mixed with virtual transmit and receive signals were generated to perform FMCW radar signal processing simulations where the IF signal underwent noise reduction through a lowpass filter. The maximum frequency was derived through the sample interval of the FFT size instead of using A/D converter. This maximum frequency was used to get the frequency range and frequency speed which were in turn used to calculate the range and speed. The virtual beat frequency generated using MATLAB is utilized to analyze the beat frequency used in the actual FMCW radar system signal processing. The differences in the range and speed of the beat frequency signals are processed and analyzed.

• Journal of the Institute of Convergence Signal Processing
• /
• v.14 no.1
• /
• pp.39-44
• /
• 2013

In this paper digital phase-shifter for multi-arm spiral antennas was designed by using Hilbert transform. All frequency components in input signal are phase-shifted for 90 degree by Hilbert transform, and the transform is implemented by FIT and IFIT. Digital phase-shifter generates two signals with phase difference of 90 degree by using Hilbert transform from input signals sampled by analog-digital converter(ADC), and then the input signal is phase-shifted for a given phase by using two signals. Hilbert transform based on digital phase-shifter is designed by Xilinx System generator, and the effects of input noise, FIT point, sampling period, initial phase of input signal, and shifted phase are simulated and its results are compared with Matlab results.

• IEMEK Journal of Embedded Systems and Applications
• /
• v.12 no.5
• /
• pp.267-275
• /
• 2017

Time of difference of arrival (TDOA) method using passive sonar sensor array has normally been used to estimate the location of a concealed moving target in underwater environment. Particle filter has been introduced for effective target estimation for non-Gaussian and nonlinear systems. In this paper, we propose a GPU-based acceleration of target position estimation using particle filter and propose efficient embedded system and software architecture. For the TDOA measurement from the passive sonar sensor, we use the generalized cross correlation phase transform (GCC-PHAT) method to obtain the correlation coefficient of the signal using FFT and we try to accelerate the calculation of GCC-PHAT based TDOA measurements using FFT with GPU CUDA. We also propose parallelization method of the target position estimation algorithm using the GPU CUDA to update the state of each particle for the target position estimation using the measured values. The target estimation algorithm was verified using Matlab and implemented using GPU CUDA. Then, we realized the proposed signal processing acceleration system using NVIDIA Jetson TX1 as the target board to analyze in terms of the execution time. The execution time of the algorithm is reduced by 55% to the CPU standalone-operation on the target board. Experiment results show that the proposed architecture is a feasible solution in terms of high-performance and area-efficient architecture.

• The Journal of the Acoustical Society of Korea
• /
• v.32 no.6
• /
• pp.509-517
• /
• 2013

In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.

• Journal of the Institute of Convergence Signal Processing
• /
• v.10 no.4
• /
• pp.214-219
• /
• 2009

This paper proposes a watermarking algorithm considering the frequency characteristics of Haegeum sounds for copyright protection of digital Haegeum sound contents. The harmonics of Haegeum sounds commonly have large magnitude values in 1500Hz~2000Hz and 2800Hz~3500Hz so that those bands are selected to embed a watermark. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and embeds the watermark bits generated by PN (pseudo noise) sequence into the harmonics in the selected bands. Furthermore, the proposed method is robust to lowpass filter, bandpass filter, cropping, noise addition, MP3 compression attacks and the maximum BER (bit error rate) is 1.41% after lowpass filter attack. To measure the quality of the watermarked sound, subjective listening test, MUSHRA (multiple stimuli with hidden reference and anchor), was conducted. The mean value of MUSHRA listening test is bigger than 98 and 96.67 for every Haegeum sounds and Korean classical music with Haeguem, respectively.

• The Transactions of the Korean Institute of Electrical Engineers D
• /
• v.50 no.12
• /
• pp.590-596
• /
• 2001

The pre-emphasis is an essential process for speech signal processing. Widely used two methods are the typical method using a fixed value near unity and te optimal method using the autocorrelation ratio of the signal. This study proposes a new pre-emphasis method using the short-term energy difference of speech signal, which can effectively compensate the glottal source characteristics and lip radiation characteristics. Using the proposed pre-emphasis, speech analysis, such as spectrum estimation, formant detection, is performed and the results are compared with those of the conventional two pre-emphasis methods. The speech analysis with 5 single vowels showed that the proposed method enhanced the spectral shapes and gave nearly constant formant frequencies and could escape the overlapping of adjacent two formants. comparison with FFT spectra had verified the above results and showed the accuracy of the proposed method. The computational complexity of the proposed method reduced to about 50% of the optimal method.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
• /
• v.19 no.2
• /
• pp.101-106
• /
• 2008

The speech signal has been put into a form suitable for storage and analysis by computer, several different operation can be performed. Filtering, sampling and quantization are the basic operation in digiting a speech signal. The waveform can be displayed, measured and even edited, and spectra can be computed using methods such as the Fast Fourier Transform (FFT), Linear predictive Coding (LPC), Cepstrum and filtering. The digitized signal also can be used to generate spectrograms. The spectrograph provide major advantages to the study of speech. So, author introduces the basic techniques for the acoustic recording, digital signal processing and the principles of spectrum and spectrogram.