• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.416-418
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• 2006

In this paper, a new acoustic echo cancellation approach based on the DUET algorithm and scaling factor estimation is proposed to solve the scaling ambiguity in case of blind separation based acoustic echo cancellation in a noisy environment. In hands-free full-duplex communication system. acoustic noises picked up by the microphone are mixed with echo signal. For this reason, the echo cancellation system may provide poor performance. For that purpose, a degenerate unmixing estimation technique, adjusted in the time-frequency domain, is employed to separate undesired echo signals and noises. Also, since scaling and permutation ambiguities have not been solved in the blind source separation algorithm, kurtosis for the desired signal selection and a scaling factor estimation algorithm are utilized in this rarer for the separation of an echo signal. Simulation results demonstrate that the proposed approach yields better echo cancellation and noise reduction performances, compared with conventional methods.

• Journal of the Korea Society of Computer and Information
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• v.21 no.3
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• pp.25-32
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• 2016

Mobile Ad Hoc Networks(MANETs) consists of mobile nodes which communicate with each other without any centralized infrastructure. Message broadcasting by flooding for route discovery in MANET can result in high redundant retransmission, contention and collision of broadcasting packet, known as the broadcast storm problem collectively. The cross-layer design is adopted in this paper, which lets routing layer share the received signal strength information at MAC layer. Also this paper proposes a new probabilistic approach that dynamically adjusts the rebroadcasting probability of a node for routing request packets (RREQs) according to the received signal strength. The simulation results show that the proposed approach demonstrates better performance than blind flooding, fixed probabilistic broadcasting approaches.

• Journal of Power Electronics
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• v.10 no.1
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• pp.1-8
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• 2010

A new scheme for sliding-mode control (SMC) of DC-DC converters with a constant switching frequency is proposed. The scheme is based on the averaging model and the output signal of the controller is $d^+$ or $d^-$ instead of the on or off signal of a direct sliding-mode (SM) controller or the continuous signal d = $u_{eq}$ of an indirect SM controller. Two approaches using the new scheme are also proposed and the design procedures for a buck converter are given in detail. The first approach called constant $d^+$ and $d^-$ SMC is simple, cost effective and dynamically fast. In order to improve the dynamic characteristics of the reaching phase and to alleviate chattering, the second approach called reaching law SMC is also presented. Analyses and simulation results demonstrate the feasibility of the proposed scheme.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.189-191
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• 2005

In the scalable extension of H.264/AVC, the codec is based on a layered approach to enable spatial scalability. In each layer, the basic concepts of motion compensated prediction and intra prediction are employed as in standard H.264/AVC. Additionally inter-layer prediction algorithm between successive spatial layers is applied to remove redundancy. In the inter-layer prediction, as the prediction we can use the signal that is the upsampled signal of the lower resolution layer. In this case, coding efficiency can be variable as the kinds of interpolation filter. In this paper, we investigate the approach to select the interpolation filter for residual signal in order to optimal prediction.

• JSTS:Journal of Semiconductor Technology and Science
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• v.13 no.6
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• pp.608-616
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• 2013

This paper presents an adaptive sampling method for electrocardiogram (ECG) signal detection. First, by employing the strings matching process with compression dictionary, we recognize each segment of ECG with different characteristics. Then, based on the non-uniform sampling strategy, the sampling rate is determined adaptively. As the results of simulation indicated, our approach reconstructed the ECG signal at an optimized sampling rate with the guarantee of ECG integrity. Compared with the existing adaptive sampling technique, our approach acquires an ECG signal at a 30% lower sampling rate. Finally, the experiment exhibits its superiority in terms of energy efficiency and memory capacity performance.

• Journal of the Korean Society for Nondestructive Testing
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• v.32 no.4
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• pp.355-361
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• 2012

This paper presents an experimental study on the detection and location of nonlinear scattering source due to the presence of fatigue crack in a laboratory specimen. The proposed technique is based on a combination of nonlinear elastic wave spectroscopy(NEWS) and time reversal(TR) focusing approach. In order to focus on the nonlinear scattering position due to the fatigue crack, we employed only one transmitting transducer and one receiving transducer, taking advantage of long duration of reception signal that includes multiple linear scattering such as mode conversion and boundary reflections. NEWS technique was then used as a pre-treatment of TR for spatial focusing of reemitted second harmonic signal. The robustness of this approach was demonstrated on a cracked specimen and the nonlinear TR focusing behavior is observed on the crack interface from which the second harmonic signal was originated.

• Journal of Institute of Control, Robotics and Systems
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• v.4 no.6
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• pp.765-771
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• 1998

An artificial neural network(NN) technique is developed for hardware redundant sensor validation. Since the measurement space is a continuous space with many operating regions, it is difficult to train a NN to correctly detect failure in an accurate measurement system. A conventional backpropagation NN is modified to include an additional preprocessing layer that extracts classification features from scalar measurements. This feature extraction means transform the measurement space to parity space. The NN is independent of the state variable being measured, the instrument range, and the signal tolerance. This NN resembles the parity space approach to signal validation, except that analytical parity equations are unneeded and the NN pattern recognition capability is utilized for decision making.

• 한국신재생에너지학회:학술대회논문집
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• 2008.10a
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• pp.94-97
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• 2008

In this paper, A Digital Voltage Mode Controller is designed for the Photovoltaic power converter applications. The designed Digital Voltage Mode Controller is derived analytically from the continuous time small signal model of the boost converter. Due to the small signal model based derivations of the control law, the designed control method can be applicable to K-factor Approach method and bilinear transformation. In order to show the usefulness of a designed controller, and the simulation results are verified.

• ETRI Journal
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• v.33 no.2
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• pp.176-183
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• 2011

This paper presents a new analytical approach and experimental verification for the improvement of noise performance in phased-array receivers. For analysis purposes, a multi-channel array system is converted into an equivalent single-channel system, such that the two presents the identical signal and noise powers at the output, respectively. We define an effective gain, noise figure, and signal-to-noise ratio in the equivalent system. Through the proposed approach, the noise performance of the array receiver is analyzed in a general and straightforward manner and then compared to that of each individual array channel. In addition, the phase noise of the array system is analyzed in a rigorous manner, showing its effective reduction by a factor of the array size. The predicted improvement of the noise performance is experimentally confirmed with a CMOS integrated phased-array receiver.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.1133-1136
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• 2005

The MPEG/audio standard results from three years of co-work by an international committe of high-fidelity audio compression experts in the Moving Picture Experts Group (MPEG/audio). The MPEG standard is rigid only where necessary to ensure interoperability. In this paper, a new approach of training and classification of acoustic signal is addressed. This is some what a fields of application aspects rather than technonical problems such as MPEG/codec, MIDI. In preprocessing, acoustic signal is transformmed using DWT so as to extract a feature parameters of sound such as loudness, pitch, bandwidth and harmonicity. these accoustic parameters are exploited to the input vector of neural perceptron. Experimental results showed that proposed approach can be used for tunning the dissonance chord.