• Journal of the Korea Institute of Military Science and Technology
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• v.22 no.3
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• pp.332-340
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• 2019

Electronic warfare systems are needed to be advantageous in the modern war. Many radar threat signals with various frequency spectrums and complicated techniques exist. For detecting the threats, a receiver with wide and narrow-band digital processing is needed. To process a wide-band searching mode, a polyphase filter bank has become the architecture of choice to efficiently detect threats. A polyphase N-path filter aligns the re-sampled time series in each path, and a discrete Fourier transform aligns phase and separates the sub-channel baseband aliases. Multiple threats and CW are detected or rejected when the signals are received in different sub-channels. And also, to process a narrow-band precision mode, a direct down converter is needed to reduce aliasing by using a decimation filter. These digital logics are designed in a FPGA. This paper shows how to design and develop a wide and narrow-band digital receiver that is capable to detect the threats.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.65 no.1
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• pp.179-193
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• 2016

In power quality monitoring, voltage fluctuation is one of the power quality problems, which cause light flickers. To determine the flicker severity, the flicker meter concept was developed in an IEC 61000-4-15 standard. Generally, voltage fluctuations are described as an amplitude modulation(AM). The flicker meter of IEC 61000-4-15 as an international standard for flicker measurement recommends square demodulation method to detect flicker signals from voltage fluctuation signals. This paper suggests a new effective method using filter bank to detect and estimate flicker signals, which do not need square demodulation. For the accurate detection of flicker signals, the filter bank is designed with a full consideration of the spectrum characteristics of voltage fluctuation signals described as AM. The frequency and magnitude of the detected flicker signals are estimated using recursive method. Computer simulations were performed on synthesized signals to prove validity of the proposed method.

• JOURNAL OF ELECTRICAL WORLD
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• s.264
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• pp.84-89
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• 1998

지금까지 수전설비의 역률조정에는 진상콘덴서가 사용되었고 콘덴서뱅크의 투입-해별에 따라 Step 모양으로 제어하여, 역률 개선효과는 0.98$\~$0.97정도가 일반적이었다.그러나 자에너지(에너지 절약$\cdot$이용합리화)에 대한 관심이 높아지는 가운데, 부하기기에 대해서는 파워일렉트로닉스 기술을 이용한 인버터화 등에 의하여 자에너지화가 도모되고 있다. 또한 코제너레이션의 설치에 의한 설비 전체로서의 에너지의 유효활동도 이루어져, 역률조정에서도 변동이 적고 가능한 한 고역률을 유지할 것을 요망하고 있다. 또 자에너지화에 크게 기여하고 있는 인버터 등에서 발생하는 전기공해라고도 할 수 있는 고조파전류가 많이 발생하고, 이 고조파전류의 유입으로 진상콘덴서의 과열$\cdot$이음발생 등의 이상을 일으키는 경우가 최근에 많이 발생하고 있다. 이와 같은 배경에서 액티브필터 제어기술을 적용하여 전력용콘덴서를 사용하지 않는 무효전력보상장치인 액티브 콘덴서(active condenser)를 제품화하여 시장에 내놓았다. 액티브 콘덴서는-무효전류 연속제어에 의하여 전원역률을 1.0으로 유지 -고속응답성 -대용량 IGBT(Insulted Gate Bipolar transistor)소자의 채용에 의한 소형화 -고조파의 영향을 받지 않음 등 지금까지의 진상콘덴서에는 없는 우수한 장점이 있어, 자에너지와 고조파문제에 유효한 수단이 되고 있다.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.6
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• pp.1-8
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• 1998

This paper presents an asynchronous DS/CDMA multiuser detector, which is a two stage, symbol-by-symbol scheme consisting of conventional detectors followed by linear decorrelating detectors. The conventional detector first makes temporal decisions and the detected symbols are delayed by one symbol period to be used for the selection of decorrelating bases in the subsequent decorrelaing detection stage. It also employs a bank of early-late correlators in place of a bank of single correlators taking the small offset of chip timing asynchronism into account. The proposed detector requires only the coarse knowledge of relative time delays of interfering users and is suitable for digital implementation. To verify the detector performance, the analytical BER performance will be given and compared with the simulation results for BPSK DS/CDMA signals in AWGN channel. While the performance of the proposed detector will be analyzed for time-limited signal, the simulation is carried out for both the time-limited and band-limited signals. As can be seen in the simulation results, the proposed scheme shows good results.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.2
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• pp.205-212
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• 2016

In this paper, we implement the development platform of multichannel digital hearing aid algorithm using Simulink provided by Matlab. The digital hearing aids are considered medical devices designed to compensate for hearing loss, they need to be correctly selected, to help a person who has difficulty in hearing. The development platform that implemented in this paper, includes WOLA filterbank for analysis/synthesis of input signal, Wide dynamic range compression for hearing loss compensation and adaptive filter for feedback cancellation. Using the development platform, algorithm parameters for each block can be set depending on the hearing aid user. Thus it is possible to test the algorithm before the machine language. As a result, the time for algorithm development can be saved and performance and computational complexity can be optimized.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.46 no.1
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• pp.128-135
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• 2009

Techniques for tracking the same region of physical space with the temporal sequences of images by matching the contours of constant phase show robust and stable performance in relative to the tracking techniques using or assuming the constant intensity. Using this property, we describe an algorithm for obtaining the robust motion parameters caused by the global camera motion. First, we obtain the optical flow based on the phase of spacially filtered sequential images on the region in a direction orthogonal to orientation of each component of gabor filter bank. And then, we apply the least squares method to the optical flow to determine the affine motion parameters. We demonstrate hat proposed method can be applied to the vision based pointing device which estimate its motion using the image including the display device which cause lighting condition varieties and noise.

• Journal of the Korea Society of Computer and Information
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• v.12 no.3
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• pp.105-113
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• 2007

In this paper, an image segmentation method which is based on edge salience map and region merging is presented. The edge salience map is calculated by combining a texture edge map with a color edge map. The texture edge map is computed over multiple spatial orientations and frequencies by using Gabor filter. A color edge is computed over the H component of the HSI color model. Then the Watershed transformation technique is applied to the edge salience map to and homogeneous regions where the dissimilarity of color and texture distribution is relatively low. The Watershed transformation tends to over-segment images. To merge the over-segmented regions, first of all, morphological operation is applied to the edge salience map to enhance a contrast of it and also to find mark regions. Then the region characteristics, a Gabor texture vector and a mean color, in the segmented regions is defined and regions that have the similar characteristics, are merged. Experimental results have demonstrated the superiority in segmentation results for various images.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.40 no.5
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• pp.333-340
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• 2003

Doppler processing is the heart of pulsed Doppler radar. It gives a clutter elimination and coherent integration. With the improvement of digital signal processors (DPSs), the implementation using them is more widely used in radar systems. Generally, so as for Doppler processor to process the input data in real time, a parallel processing concept using multiple DSPs should be used. This paper implements a programmable Doppler processor, which consists of MTI filter, DFB and square-law detector, using 8 ADSP21060s. Formulating the distribution time of the input data, the transfer time of the output data and the time required to compute each algorithm, it estimates total processing time and the number of required DSP. Finally, using the TSG that provides radar control pulses and simulated target signals, performances of the implemented Doppler processor are evaluated.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.5 s.311
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• pp.79-87
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• 2006

In this study, we evaluated the performance of a companding strategy as a preprocessing for speech enhancement and noise reduction. The proposed algorithm is based on two tone suppression that is human's hearing characteristics. This algorithm enhances spectral peak of speech signal and reduces background noise, however it has tradeoff characteristics between speech distortion and noise reduction due to limited channel number and nonlinear block. Therefore, we designed two different companding structures that have relative characteristics of noise reduction and speech distortion and found suitable companding structures by difference of individual speech perception ability in noise environment. Thus we proposed speech perception enhancement of cochlear implant user in noise environment with low SNR. The performance of the proposed algorithm was evaluated through 5 normal hearing listeners using noise band simulation. Improvement of speech perception was observed for all subjects and each subject preferred the different type of companding structure.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.77-85
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• 2001

This paper proposes a method of time-scale modification of polyphonic audio signals based on a sinusoidal model. The signals are modeled with sinusoidal component and noise component. A multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in time-scale modification a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. For extracting sinusoidal components and calculating their parameters matching pursuit algorithm is applied to each analysis frame of subband signal. In accordance with spectrum analysis a psychoacoustic model implementing the effect of frequency masking is incorporated with matching pursuit to provide a resonable stop condition of iteration and reduce the number of sinusoids. The noise component obtained by subtracting the synthesized signal with sinusoidal components from the original signal is modeled by line-segment model of short time spectrum envelope. For various polyphonic audio signals the result of simulation shows suggested sinusoidal modeling can synthesize original signal without loss of perceptual quality and do more robust and high quality time-scale modification for large scale factor because of representing transients without any perceptual loss.