• Title/Summary/Keyword: Signal Waveform

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A Practical standard Air Flow Generator System to Calibrate and Compare Performance of Two Different Respiratory Air Flow Measurement Modules (호흡기류 계측모듈의 교정과 성능 비교를 위한 실용적인 표준기류 생성 시스템)

  • Lee, In-Kwang;Park, Mi-Jung;Lee, Sang-Bong;Kim, Kyoung-Ok;Cha, Eun-Jong;Kim, Kyung-Ah
    • Journal of Biomedical Engineering Research
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    • v.36 no.4
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    • pp.115-122
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    • 2015
  • A standard air flow generator system was developed to generate air flows of various levels simultaneously applied to two different air flow transducer modules. Axes of two identical standard syringes for spirometer calibration were connected with each other and driven by a servo-motor. Linear displacement transducer was also connected to the syringe axis to accurately acquire the volume change signal. The user can select either sinusoidal or square waveform of volume change and manually input any volume as well as maximal flow rate levels ranging 0~3 l and 0~15 l/s, respectively. Various volume and flow levels were input to operate the system, then the volume signal was acquired followed by numerical differentiation to obtain the air flow signal. The measured volumes and maximal air flow rates were compared with the user input data. The relative errors between the user-input and the measured stroke volumes were all within 0.5%, demonstrating very accurate driving of the system. In case of the maximal flow rate, relatively large error was observed when the syringe was driven very fast within a very short time duration. However, except for these few data, most measured flow rates revealed relative errors of approximately 2%. When the measure and user-input stroke volume and maximal flow rate data were analyzed by linear regression analysis, respectively, the correlation coefficients were satisfactorily higher than 0.99 (p < 0.0001). These results demonstrate that the servo-motor controls the syringes with enough accuracy to generate standard air flows. Therefore, the present system would be very much practical for calibration process as well as performance evaluation and comparison of two different air flow transducer modules.

Characteristics of Impulse Radios for Mu1tipath Channels (다중 경로 채널에서 임펄스 라디오의 특징)

  • 이호준;한병칠
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.26 no.11B
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    • pp.1501-1509
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    • 2001
  • Recently, the use of wireless communication systems has been rapidly increasing, which results in a difficult problem in efficient control of limited frequency resources. As a way of solving this problem, the ultra wideband time hopping impulse radio system attracts much attention. The impulse radio system communicates pulse position modulated data using Gaussian monocycle pulses of very short duration less than 1 nsec. Thus the transmitted signal has very low power spectral density and ultra wide bandwidth from near D.C. to a few GHz. It is blown that it hardly interferes with the existing communication systems because of its very low power spectral density. The purpose of this paper is to characterize multipath propagation of the impulse radio signal and to evaluate the performance of the correlator-based receiver for the multipath environments. In this paper, we consider the deterministic two-path model and the statistical indoor multipath model of Saleh and Valenzuela. For the two-path model the output of the correlator with the ideal reference waveform varies according to the relative difference between the indirect path delay and the time interval of PPM, and to the indirect path gains. In addition, the characteristics of bit error rates is measured for the two models through computer simulation. The simulation results indicate that the performance of the impulse radio system depends both on the relative difference between the indirect path delay and the time interval of PPM, and on the indirect path gains. Furthermore, it is observed that the reference signal designed for the AWGN channel can not be applied to the multipath channels.

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PCA­based Waveform Classification of Rabbit Retinal Ganglion Cell Activity (주성분분석을 이용한 토끼 망막 신경절세포의 활동전위 파형 분류)

  • 진계환;조현숙;이태수;구용숙
    • Progress in Medical Physics
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    • v.14 no.4
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    • pp.211-217
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    • 2003
  • The Principal component analysis (PCA) is a well-known data analysis method that is useful in linear feature extraction and data compression. The PCA is a linear transformation that applies an orthogonal rotation to the original data, so as to maximize the retained variance. PCA is a classical technique for obtaining an optimal overall mapping of linearly dependent patterns of correlation between variables (e.g. neurons). PCA provides, in the mean-squared error sense, an optimal linear mapping of the signals which are spread across a group of variables. These signals are concentrated into the first few components, while the noise, i.e. variance which is uncorrelated across variables, is sequestered in the remaining components. PCA has been used extensively to resolve temporal patterns in neurophysiological recordings. Because the retinal signal is stochastic process, PCA can be used to identify the retinal spikes. With excised rabbit eye, retina was isolated. A piece of retina was attached with the ganglion cell side to the surface of the microelectrode array (MEA). The MEA consisted of glass plate with 60 substrate integrated and insulated golden connection lanes terminating in an 8${\times}$8 array (spacing 200 $\mu$m, electrode diameter 30 $\mu$m) in the center of the plate. The MEA 60 system was used for the recording of retinal ganglion cell activity. The action potentials of each channel were sorted by off­line analysis tool. Spikes were detected with a threshold criterion and sorted according to their principal component composition. The first (PC1) and second principal component values (PC2) were calculated using all the waveforms of the each channel and all n time points in the waveform, where several clusters could be separated clearly in two dimension. We verified that PCA-based waveform detection was effective as an initial approach for spike sorting method.

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A Digital Image Watermarking Using A Bottom-up Attention Module (상향식 주의 모듈을 사용한 디지털 워터마킹 기법)

  • Cheoi, Kyung-Joo
    • The KIPS Transactions:PartB
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    • v.15B no.4
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    • pp.293-300
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    • 2008
  • This paper takes a bottom-up attention module into consideration for digital image watermarking. A bottom-up attention module is employed to obtain the region of interest, and watermark information is embedded into the obtained region. Previous studies in digital image watermarking have been focused on the signal processing techniques, especially in waveform coding spreading watermarks over the entire target image. However, we notice that the third party's visual attention is usually concentrated on a few regions in an image but not on all of them. These regions are easy to be the target of attacks. If watermark information is inserted into these regions from the beginning, it can be detected with high correlation. Various kinds of images are tested, and the results showed good quality.

Algorithm Study for Diagnosis the Breast Cancer Using LMA and FDTD (LMA와 FDTD를 이용한 유방암 진단용 알고리즘 연구)

  • Seo, Min-Gyeong;Kim, Tae-Hong;Mun, Ji-Yeon;Jeon, Soon-Ik;Pack, Jeong-Ki
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.22 no.12
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    • pp.1124-1131
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    • 2011
  • In this paper, image reconstruction algorithm for breast cancer detection using MT(Microwave Tomography) was investigated. The breast cancer detection system under development uses 16 transmit/receive antennas. The signal waveform was a sinusoidal wave at 900 MHz. To solve the 2D inverse scattering problem, we used the 2D FDTD (Finite Difference Time Domain) method for forward calculation and LMA(Levenberg-Marquardt Algorithm) for optimization. The result of the image reconstruction using the numerical phantom by MRI(Magnetic Resonance Imaging) obtained from real patient of breast cancer showed that we can detect the position of the tumor accurately.

Music Genre Classification using Spikegram and Deep Neural Network (스파이크그램과 심층 신경망을 이용한 음악 장르 분류)

  • Jang, Woo-Jin;Yun, Ho-Won;Shin, Seong-Hyeon;Cho, Hyo-Jin;Jang, Won;Park, Hochong
    • Journal of Broadcast Engineering
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    • v.22 no.6
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    • pp.693-701
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    • 2017
  • In this paper, we propose a new method for music genre classification using spikegram and deep neural network. The human auditory system encodes the input sound in the time and frequency domain in order to maximize the amount of sound information delivered to the brain using minimum energy and resource. Spikegram is a method of analyzing waveform based on the encoding function of auditory system. In the proposed method, we analyze the signal using spikegram and extract a feature vector composed of key information for the genre classification, which is to be used as the input to the neural network. We measure the performance of music genre classification using the GTZAN dataset consisting of 10 music genres, and confirm that the proposed method provides good performance using a low-dimensional feature vector, compared to the current state-of-the-art methods.

Maximum Canopy Height Estimation Using ICESat GLAS Laser Altimetry

  • Park, Tae-Jin;Lee, Woo-Kyun;Lee, Jong-Yeol;Hayashi, Masato;Tang, Yanhong;Kwak, Doo-Ahn;Kwak, Han-Bin;Kim, Moon-Il;Cui, Guishan;Nam, Ki-Jun
    • Korean Journal of Remote Sensing
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    • v.28 no.3
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    • pp.307-318
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    • 2012
  • To understand forest structures, the Geoscience Laser Altimeter System (GLAS) instrument have been employed to measure and monitor forest canopy with feasibility of acquiring three dimensional canopy structure information. This study tried to examine the potential of GLAS dataset in measuring forest canopy structures, particularly maximum canopy height estimation. To estimate maximum canopy height using feasible GLAS dataset, we simply used difference between signal start and ground peak derived from Gaussian decomposition method. After estimation procedure, maximum canopy height was derived from airborne Light Detection and Ranging (LiDAR) data and it was applied to evaluate the accuracy of that of GLAS estimation. In addition, several influences, such as topographical and biophysical factors, were analyzed and discussed to explain error sources of direct maximum canopy height estimation using GLAS data. In the result of estimation using direct method, a root mean square error (RMSE) was estimated at 8.15 m. The estimation tended to be overestimated when comparing to derivations of airborne LiDAR. According to the result of error occurrences analysis, we need to consider these error sources, particularly terrain slope within GLAS footprint, and to apply statistical regression approach based on various parameters from a Gaussian decomposition for accurate and reliable maximum canopy height estimation.

Design of 10-Gb/s Adaptive Decision Feedback Equalizer with On-Chip Eye-Opening Monitoring (온 칩 아이 오프닝 모니터링을 탑재한 10Gb/s 적응형 Decision Feedback Equalizer 설계)

  • Seong, Chang-Kyung;Rhim, Jin-Soo;Choi, Woo-Young
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.48 no.1
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    • pp.31-38
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    • 2011
  • With the increasing demand for high-speed transmission systems, adaptive equalizers have been widely used in receivers to overcome the limited bandwidth of channels. In order to reduce the cost for testing high-speed receiver chips, on-chip eye-opening monitoring (EOM) technique which measures the eye-opening of data waveform inside the chip can be employed. In this paper, a 10-Gb/s adaptive 2-tap look-ahead decision feedback equalizer (DFE) with EOM function is proposed. The proposed EOM circuit can be applied to look-ahead DFEs while existing EOM techniques cannot. The magnitudes of the post-cursors are measured by monitoring the eye of received signal, and coefficients of DFE are calculated using them by proposed adaptation algorithm. The circuit designed in 90nm CMOS technology and the algorithm are verified with post-layout simulation. The DFE core occupies $110{\times}95{\mu}m^2$ and consumes 11mW in 1.2V supply voltage.

Development of Blood Pressure Estimation Methods Using The PPG and ECG Sensors (PPG 및 ECG 센서를 이용한 혈압추정 기법 개발)

  • Park, Hyun-Moon;Lee, Jung-Chul;Hwang, Tae-Ho
    • The Journal of the Korea institute of electronic communication sciences
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    • v.14 no.6
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    • pp.1257-1264
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    • 2019
  • The traditional cuff-based method for BP(Blood Pressure) measurement is not suitable for continuous real-time BP measurement techniques. For this reason, the previous studies estimated various blood pressures by fusion with the electrocardiography (ECG) and photoplethysmogram (PPG) sensor signals. However, conventional techniques based on PPG bio-sensing measurement face many challenging issues such as noisy supply fluctuation, small pulsation, and drifting non-pulsatile. This paper proposed a novel BP estimation methods using PPG and ECG sensors, which can be derived from the relationship between PPG and ECG using PTT(Pulse Transit Time) and PWV(Pulse Wave Velocity). Unlike conventional height ratio features, which are extracted on the basis of the peaks in the PPG and ECG waveform. The proposed method can be reliably obtained even if there are missing peaks among the sensed PPG signal. The increased reliability comes from periodical estimation of the peak-to-peak interval time using ECG and PPG. After 250,000 times trials of the blood pressure measurement, the proposed estimation technique was verified with the accuracy of ±28.5% error, compared to a commercialized BP device.

A Study on Multi-Pulse Speech Coding Method by Using V/S/TSIUVC (V/S/TSIUVC를 이용한 멀티펄스 음성부호화 방식에 관한 연구)

  • Lee See-Woo
    • Journal of Korea Multimedia Society
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    • v.7 no.9
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    • pp.1233-1239
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    • 2004
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new multi-pulse coding method by using V/S/TSIUVC switching, individual pitch pulses and TSIUVC approximation-synthesis method in order to restrict a distortion of speech quality. The TSIUVC is extracted by using the zero crossing rate and individual pitch pulse. And the TSIUVC extraction rate was 91% for female voice and 96.2% for male voice respectively. The important thing is that the frequency information of 0.347kHz below and 2.813kHz above can be made with high quality synthesis waveform within TSIUVC. I evaluate the MPC use V/UV and the FBD-MPC use V/S/TSIUVC. As a result, I knew that synthesis speech of the FBD-MPC was better in speech quality than synthesis speech of the MPC.

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