• Journal of Communications and Networks
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• v.12 no.4
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• pp.289-307
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• 2010

The problem of model selection arises in a number of contexts, such as subset selection in linear regression, estimation of structures in graphical models, and signal denoising. This paper studies non-asymptotic model selection for the general case of arbitrary (random or deterministic) design matrices and arbitrary nonzero entries of the signal. In this regard, it generalizes the notion of incoherence in the existing literature on model selection and introduces two fundamental measures of coherence-termed as the worst-case coherence and the average coherence-among the columns of a design matrix. It utilizes these two measures of coherence to provide an in-depth analysis of a simple, model-order agnostic one-step thresholding (OST) algorithm for model selection and proves that OST is feasible for exact as well as partial model selection as long as the design matrix obeys an easily verifiable property, which is termed as the coherence property. One of the key insights offered by the ensuing analysis in this regard is that OST can successfully carry out model selection even when methods based on convex optimization such as the lasso fail due to the rank deficiency of the submatrices of the design matrix. In addition, the paper establishes that if the design matrix has reasonably small worst-case and average coherence then OST performs near-optimally when either (i) the energy of any nonzero entry of the signal is close to the average signal energy per nonzero entry or (ii) the signal-to-noise ratio in the measurement system is not too high. Finally, two other key contributions of the paper are that (i) it provides bounds on the average coherence of Gaussian matrices and Gabor frames, and (ii) it extends the results on model selection using OST to low-complexity, model-order agnostic recovery of sparse signals with arbitrary nonzero entries. In particular, this part of the analysis in the paper implies that an Alltop Gabor frame together with OST can successfully carry out model selection and recovery of sparse signals irrespective of the phases of the nonzero entries even if the number of nonzero entries scales almost linearly with the number of rows of the Alltop Gabor frame.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.12
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• pp.1250-1260
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• 2014

In this paper, a novel resource scheduling, which is used for hub network based long distance airborne tactical communication, is proposed. Recently, some countries of the world has concentrated on developing data rate and networking performance of CDL, striving to keep pace with modern warfare, which is changed into NCW. And our government has also developed the next generation high capacity CDL. In hub network, a typical communication structure of CDL, hybrid FDMA/TDMA can be considered to exchange high rate data among multiple UAVs simultaneously, within limited bandwidth. However, due to different RTT and traffic size of UAV, idle time resource and unnecessary packet transmission delay can occur. And these losses can reduce entire efficiency of hub network in long distance communication. Therefore, in this paper, we propose RTT and data traffic size based UAV scheduling, which selects time/frequency resource of UAVs by using iterative sorting algorithm. The simulation results verified that the proposed scheme improves data rate and packet delay performance in low complexity.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.733-741
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• 2009

The design rule of digital communication systems is the reliable data transmission with high spectral efficiency and minimum allowable power. This paper suggests the method that saves the average power by implementing a multi-dimensional constellation in case of multi-carrier communication system. By using multi-dimensional constellations we can relocate constellation points in the form of a sphere. If we simply convert the two-dimensional QAM modulation into multi-dimensional QAM, constellation points of 2 N dimensional cube form are made up. Relocating outermost constellation points of 2 N dimensional cube form into low energy constellation points, the constellation of the 2 N-dimensional sphere form is made up which decreases power consumption. In this paper, the multi-dimensional constellations of 2 N-dimensional sphere form are designed from 16-QAM to 2,048-QAM, and power reductions are obtained by comparing constellations of 2-dimensional QAMs and multi-dimensional constellations of 2 N-dimensional sphere form. The result shows that the average power consumption of higher dimensional constellations increases, because the more a dimension elevates, the more the relocatable constellation points increase. But, the increment of the average power savings decreases as the a dimension elevates. The transmission of the data by using multi-dimensional constellations of the sphere form is effective to save the average power consumption with little hardware complexity.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.43 no.1 s.343
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• pp.27-36
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• 2006

Conventional delay-insensitive (DI) data encodings usually require 2N+1 wires for transferring N-bit. To reduce complexity and power dissipation of wires in designing a large scaled chip, an encoder and a decoder circuits, where N-bit data transfer can be peformed with only N+l wires, are proposed. These circuits are based on a quasi delay-insensitive (QDI) model and designed by using current-mode multiple valued logic (CMMVL). The effectiveness of the proposed data transfer mechanism is validated by comparisons with conventional data transfer mechanisms using dual-rail and 1-of-4 encodings through simulation at the 0.25 um CMOS technology. In general, simulation results with wire lengths of 4 mm or larger show that the CMMVL scheme significantly reduces delay-power product ($D{\ast}P$) values of the dual-rail encoding with data rate of 5 MHz or more and the 1-of-4 encoding with data rate of 18 MHz or more. In addition, simulation results using the buffer-inserted dual-rail and 1-of-4 encodings for high performance with the wire length of 10 mm and 32-bit data demonstrate that the proposed CMMVL scheme reduces the D*P values of the dual-rail encoding with data rate of 4 MHz or more and 1-of-4 encoding with data rate of 25 MHz or more by up to $57.7\%\;and\;17.9\%,$ respectively.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.46 no.5
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• pp.8-14
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• 2009

In the research field of speech recognition, pinpointing the endpoints of speech utterance even with the presence of background noise is of great importance. These noise present during recording introduce disturbances which complicates matters since what we just want is to get the stationary parameters corresponding to each speech section. One major cause of error in automatic recognition of isolated words is the inaccurate detection of the beginning and end boundaries of the test and reference templates, thus the necessity to find an effective method in removing the unnecessary regions of a speech signal. The conventional methods for speech endpoint detection are based on two linear time-domain measurements: the short-time energy, and short-time zero-crossing rate. They perform well for clean speech but their precision is not guaranteed if there is noise present, since the high energy and zero-crossing rate of the noise is mistaken as a part of the speech uttered. This paper proposes a novel approach in finding an apparent threshold between noise and speech based on Lyapunov Exponents (LEs). This proposed method adopts the nonlinear features to analyze the chaos characteristics of the speech signal instead of depending on the unreliable factor-energy. The excellent performance of this approach compared with the conventional methods lies in the fact that it detects the endpoints as a nonlinearity of speech signal, which we believe is an important characteristic and has been neglected by the conventional methods. The proposed method extracts the features based only on the time-domain waveform of the speech signal illustrating its low complexity. Simulations done showed the effective performance of the Proposed method in a noisy environment with an average recognition rate of up 92.85% for unspecified person.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.4
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• pp.71-81
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• 2010

In grid system, Task scheduling based on list scheduling models has showed low complexity and high efficiency in fully connected processor set environment. However, earlier schemes did not consider sufficiently the communication cost among tasks and the composition process of lightpath for communication in optical gird environment. In this thesis, we propose LSOG (Leveling Selection in Optical Grid) which sets task priority after forming a hierarchical directed acyclic graph (DAG) that is optimized in optical grid environment. To determine priorities of task assignment in the same level, proposed algorithm executes the task with biggest communication cost between itself and its predecessor. Then, it considers the shortest route for communication between tasks. This process improves communication cost in scheduling process through optimizing link resource usage in optical grid environment. We compared LSOG algorithm with conventional ELSA (Extended List Scheduling Algorithm) and SCP (Scheduled Critical Path) algorithm. We could see the enhancement in overall scheduling performance through increment in CCR value and smoothing network environment.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.4
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• pp.181-190
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• 2014

The underwater acoustic lens system is one of the systems getting high-resolution images on the seafloor by the beam forming method using acoustic lens. The beam forming using acoustic lenses reduces complexity and driving power. When receiving an incoming beam with the acoustic lens array, beam pattern analysis and arrangement problem of the array sensor must be addressed. Introducing SIR (Signal to Interference Ratio), the relationship among sensor interval, beam pattern and image quality would be analyzed. Generally if the sensor interval getting wider, the less effect of the side lobes makes SIR high. If the amplitude of a side lobe is high, SIR is generally getting low. The type of the apodization function changes the width, shape and amplitude of both main lobe and side lobes. Thus an appropriate apodization function can improve SIR. In this paper, SIR is stable at the sensor interval of 13mm with 0-10dB, which is not high relatively. By applying the Chebyshev function, the SIR becomes 80dB over the sensor interval of 37 mm or higher. The Hann and triangular functions demonstrate better SIR when the sensor interval becomes narrower.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1082-1091
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• 2000

In this paper, the efficient equalization method for OFDM(Orthogonal Frequency Division Multiflexing) System using the QAM(Quadrature Amplitude Modulation) in multipath fading channel is proposed in order to faster and more efficiently equalize the received signals that are sent over real channel. In generally, the one-tap linear equalizers have been used in the frequency-domain as the existing equalization method for OFDM system. In this technique, if characteristics of the channel are changed fast, the one-tap linear equalizers cannot compensate for the distortion due to time variant multipath channels. Therefore, in this paper, we use one-tap non-linear equalizers instead of using one-tap linear equalizers in the frequency-domain, and also use the linear equalizer in the time-domain to compensate the rapid performance reduction at the low SNR(Signal-to-Noise Ratio) that is the disadvantage of the non-linear equalizer. In the frequency-domain, when QAM signals, consisting of in-phase components and quadrature (out-phase) components, are sent over the complex channel, the only in-phase and quadrature components of signals distorted by the multipath fading are changed the same as signals distorted by the noise. So the cross components are canceled in the frequency-domain equalizer. The time-domain equalizer and the adaptive algorithm that has lower-error probability and fast convergence speed are applied to compensate for the error that is caused by canceling the cross components in the frequency-domain equalizer. In the time-domain, To compensate for the performance of frequency-domain equalizer the time-domain equalizes the distorted signals at a frame by using the Gold-code as a training sequence in the receiver after the Gold-codes are inserted into the guard signal in the transmitter. By using the proposed equalization method, we can achieve faster and more efficient equalization method that has the reduced computational complexity and improved performance.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.1
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• pp.135-141
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• 2018

In this paper, we describe design and implementation of optimal smart home control system. Recent developments in technologies such as sensors and communication have enabled the Internet of Things to control a wide range of objects, such as light bulbs, socket-outlet, or clothing. Many businesses rely on the launch of collaborative services between them. However, traditional IoT systems often support a single protocol, although data is transmitted across multiple protocols for end-to-end devices. In addition, depending on the manufacturer of the Internet of things, there is a dedicated application and it has a high degree of complexity in registering and controlling different IoT devices for the internet of things. ARIoT system, special marking points and edge extraction techniques are used to detect objects, but there are relatively low deviations depending on the sampling data. The proposed system implements an IoT gateway of object based on OneM2M to compensate for existing problems. It supports diverse protocols of end to end devices and supported them with a single application. In addition, devices were learned by using deep learning in the artificial intelligence field and improved object recognition of existing systems by inference and detection, reducing the deviation of recognition rates.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.2
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• pp.40-46
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• 2010

Multi-user MIMO (Multiple-Input Multiple-Output) systems require collaborative PF schedulers to improve the performance of the log sum of average transmission rates. While the performance of single cell based conventional PF schedulers has been evaluated over various channel conditions, scheduling algorithms by multiple base stations which select multiple users over a given time frame and their performance require further investigations. In this paper, we apply a collaborative PF scheduler to the distributed multi-user MIMO system, which assigns radio resources to multiple users by exchanging user channel information from base stations located in three adjacent sectors. We further evaluate its performance in terms of the log sum of average transmission rates. The performance is compared to that of the full-search collaborative PF scheduler which searches over all possible combinations of user groups, and that of a parallel PF scheduler that determines users without channel information exchange among base stations. We show the log sum of average transmission rates of the collaborative PF scheduler outperforms that of the parallel PF scheduler in low percentile region. In addition, the collaborative PF scheduler exhibits a negligible performance degradation when compared to the full-search collaborative PF scheduler while a significant reduction of the computational complexity is achievable at the same time.