• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.2
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• pp.253-260
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• 2009

This paper, mathematically analyzes the performance of newly proposed TR-UWB system which the frequency components of a UWB pulse were processed so that the system could be implemented with ADCs of a few MHz sampling rate, and presents the comparison with an existing frequency-domain based TR-UWB system. The comparison is mainly based on the SNR ratio which depends on the mean and the variance of the frequency components. We also shows that the simulation results to support the theoretical analysis where the comparison is made under the IEEE 802.15.3a channel model as well as AWGN channel.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.22 no.1
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• pp.40-46
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• 2011

Background and Objectives : Vocal fold bowing is an organic voice disorder that is associated with an abnormal structure of the vocal folds whereas vocal fold polyp is a functional voice disorder caused by an abnormal use of the vocal folds. Both types of vocal folds share a common property in that they make one's voice breathy or strained. The purpose of this study is to compare voice from two types of vocal folds and to offer information of clinical importance. Materials and Method: Vocal fold bowing and vocal fold polyp groups consisted of 7 male subjects, respectively. All subjects recorded /a/ in the state of measuring MPT (maximum phonation time), repeating 3 times, by a voice recorder (48 kHz sampling rate; 24 bit quantization). They answered the questions of K-VHI. Time domain parameters (such as perturbation parameters including HNR, Jitter, etc.) were calculated for the whole duration of /a/ and those of the frequency domain were measured in initial 40 ms and stable 40 ms of /a/, respectively. Mann-Whitney V-test was used for the time domain parameters and K-VHI survey, and Wilcoxon signed rank test was applied to the frequency domain parameters (H1, H2, H1-H2). Results: For K-VHI survey and the time domain analysis, there was no significant difference between bowing and polyp group. For frequency domain analysis, H1 and H2 showed a significantly different result between two groups. Vocal fold bowing group has longer duration and lower intensity than that of vocal fold polyp group in the 'aspirated interval', which could be observable prior to ordinary vowel oscillation. Conclusion: Both groups seem to show breathy voice. This could be referred on the basis of the value of H1-H2. The K-VHI survey says that subjects with vocal fold bowing feel more uncomfortable than subjects with vocal fold polyp.

• Journal of Positioning, Navigation, and Timing
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• v.9 no.4
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• pp.357-366
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• 2020

This paper addressed a relative navigation method for autonomous rendezvous and docking of cube-satellites using single frequency Differential GPS (DGPS) under the intermittent communication between satellites. Since the ionospheric error of GPS measurement is variable depending on the visible satellites, a few meters error of relative navigation is occurred in the Low-Earth Orbit (LEO) environment. Therefore, it is essential to remove the ionospheric error to perform relative navigation. Besides, an intermittent communication period for receiving GPS measurements of the target satellite is limited for getting information every sampling time. To solve this problem, a method combining range domain DGPS and orbit propagation is proposed in this paper. The proposed method improves the performance of DGPS by using Hatch filter and solves an intermittent communication problem by estimating the relative position and velocity using Hill-Clohessy-Wiltshire Equation. Through the simulation, it is verified that the suggested algorithm provides the relative position error within RMS 0.5 m and the relative velocity error within RMS 3 cm/s. Furthermore, it has the advantage that it is suitable for real-time implementation using single-frequency GPS measurements and is computationally efficient.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.10C
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• pp.951-962
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• 2002

A DMT transceiver recovers the sampling time from reserved sub-carriers, the pilots. Since the pilots are available after the FFT, the symbol synchronization must be done before sample synchronization. In DMT VDSL system, symbol synchronization is handled separately from sample synchronization, although the two processes are intimately related. The DMT symbol itself contains sufficient information, the cyclic extension, for symbol synchronization. Using only the sign bit of received signal, the Maximum Likelihood Estimation solution is derived. The Tx windowing in the transmitter of DMT VDSL system results in the blurring of MLE peaks. We propose the weighted summing MLE method using the sign bit which produces the clearly sharp top of MLE peaks. The stability of symbol synchronization is improved significantly by averaging over a few symbols. This paper presents the study of the original MLE and the weighted summing MLE using sign bit. A clock difference between transmitter and receiver destroys the oahogonality of the carriers. Therefore, a receiver using asynchronous sampling must perform timing correction in the discrete-time domain. We introduce an efficient digital sample synchronization method which is based on temporal and frequency domain digital signal processing.

• Journal of IKEEE
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• v.13 no.1
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• pp.41-48
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• 2009

This paper describes an implementation of the analog front end (AFE) incorporated with the image signal processing (ISP) unit in the SoC, dominating the performance of the CCD image sensor system. New schemes are exploited in the high-frequency sampling to reduce the sampling uncertainty apparently as the frequency increases, in the structure for the wide-range variable gain amplifier (VGA) capable of $0{\sim}36\;dB$ exponential gain control to meet the needed bandwidth and accuracy by adopting a new parasitic insensitive capacitor array. Moreover, the double cancellation of the black-level noise was efficiently achieved both in the analog and the digital domain. The proposed topology fabricated in a $0.35-{\mu}m$ CMOS process was proved in a full CCD camera system of 10-bit accuracy, dissipating 80 mA at 15 MHz with a 3.3 V supply voltage.

• Journal of Korean Vacuum Science & Technology
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• v.3 no.2
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• pp.107-111
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• 1999

We developed the experimental function (ERF) which can be used for the numerical curve fitting analysis in photoelectron spectroscopy (PES). We selected the core-levels of Ag 3d5/2 and Au 4f7/2 to obtain the ERF from the measured core-level spectra. For the numerical fourier transformation we applied the fast transform (FFT) algorithm. we considered optical (Wiener) filtering with the FFT due to noise and used Hann window function to remedy the information leakage in frequency domain due to discrete and finite sampling of measurement. The comparison of the curve fitting results using the ERF obtained in this work and the mathematical response function with a gaussian in the conventional approach shows clearly the improvement of the curve fitting analysis.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2329-2338
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• 1997

A conventional digital receiver sampled a baseband signal and processed it digitally for demodulation. But now we can sample at sufficiently high speed a wideband signal to take enough discrete data values due to the advent of economic high-speed ADC. With this technical background, a wideband frequency-division-multiplexed signal can be undersampled and channelized in digital domain by DFT analysis filter using the theory of polyphase. In this paper, we propose a new digital receiver which can digitally process the multichannel received signal by sampling at IF band, develop a mathematical theory and algorithm, and analyze the performance by using C-language simulaation. The proposed receiver can demodulate analog and digital FM signals.

• Proceedings of the Optical Society of Korea Conference
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• 2009.10a
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• pp.331-332
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• 2009

We present an analysis on the quality factors of the Fourier hologram generated from multiple orthographic view images of three-dimensional object. In the analysis, we analyze both the maximum size of the reconstructed object and its spatial resolution. For the maximum size of the reconstruction, we found that the main factor is the orthographic projection angle interval. Too large projection angle interval causes overlapping in the reconstruction space domain. For the spatial resolution, there are three factors, i.e. the capturing lens array pitch which determines the spatial sampling rate of the original three-dimensional objects, the maximum orthographic projection angle, and the spatial frequency bandwidth of the object. The dominant factor is determined by the relationship between those three factors.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.29 no.10 s.241
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• pp.1430-1437
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• 2005

It is very important to achieve a high reproducibility in the ultrasonic measurement of bone mineral density. In this study, we examined number of sampling waveform, control of temperature, diameter of region of interest as factors to improve reproducibility. We decided the optimal number of waveforms to be converted to frequency domain as period of 1. We have minimized the effects of variable temperature and constrained generation of micro bubble by keeping temperature within a range of $32\pm0.5^{\circ}C$ with a precise temperature controlling algorithm. We also found the optimal diameter of region of interest to be 13mm. In this paper, we demonstrated the improved reproducibility by controlling various factors affecting the ultrasonic measurement of bone mineral density.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.