• Journal of the Korean Geotechnical Society
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• v.34 no.3
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• pp.13-27
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• 2018

To study how stable the rubble mound breakwaters are, one can look to the research of wave induced seepage flow through the pores of the rubble mound. Seepage flow is generally generated by the difference between the water level around the breakwater during a typhoon. The existing stability analysis method of the rubble mound is the static analysis which simply considers the force equilibrium taking into account the horizontal force acting on the concrete block induced by a wave (calculated by Goda equation) and the vertical force induced by the weight inclusive of the concrete block, quarry run, filter, and armor layer above the slipping plane. However, this static method does not consider the wave-induced seepage flow in the rubble mound. Such seepage may decrease the stability of the rubble mound. The stability of a rubble mound breakwater under the action of seepage was studied based on the results of CFD software (OpenFOAM) and Limit Equilibrium Method (GeoStudio). The numerical analysis result showed that the seepage flow decreased the stability of the rubble mound breakwaters. The results of the numerical analyses also revealed the stability of the rubble mound was varied with time. Especially, the most critical state happened at the condition of overtopping the concrete block, acting strong uplift pressure raising along side and underneath the concrete block, and generating high pore pressure inside rubble mound due to seepage flow. Therefore, it may be necessary to conduct a dynamic analysis considering the effect of wave-induce seepage flow together with the static analysis.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.05a
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• pp.465-469
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• 2000

This paper describes a single-chip full-custom implementation of pipelined adaptive decision-feedback equalizer (PADFE) using a 0.25-${\mu}{\textrm}{m}$ CMOS technology for wide-band wireless digital communication systems. To enhance the throughput rate of ADFE, two pipeline stage are inserted into the critical path of the ADFE by using delayed least-mean-square (DLMS) algorithm Redundant binary (RB) arithmetic is applied to all the data processing of the PADFE including filter taps and coefficient update blocks. When compared with conventional methods based on two's complement arithmetic, the proposed approach reduces arithmetic complexity, as well as results in a very simple complex-valued filter structure, thus suitable for VLSI implementation. The design parameters including pipeline stage, filter tap, coefficient and internal bit-width and equalization performance such as bit error rate (BER) and convergence speed are analyzed by algorithm-level simulation using COSSAP. The singl-chip PADFE contains about 205,000 transistors on an area of about 1.96$\times$1.35-$\textrm{mm}^2$. Simulation results show that it can safely operate with 200-MHz clock frequency at 2.5-V supply, and its estimated power dissipation is about 890-mW.

• Journal of Broadcast Engineering
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• v.7 no.1
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• pp.45-54
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• 2002

This paper proposes a video compression algorithm using characteristics of wavelet coefficients. The proposed algorithm can provide lowed bit rate and faster running time while guaranteeing the reconstructed image qualify by the human virtual system. In this approach, each video sequence is decomposed into a pyramid structure of subimages with various resolution to use multiresolution capability of discrete wavelet transform. Then similarities between two neighboring frames are obtained from a low-frequency subband which Includes an important information of an image and motion informations are extracted from the similarity criteria. Four legion selection filters are designed according to the similarity criteria and compression processes are carried out by encoding the coefficients In preservation legions and replacement regions of high-frequency subbands. Region selection filters classify the high-frequency subbands Into preservation regions and replacement regions based on the similarity criteria and the coefficients In replacement regions are replaced by that of a reference frame or reduced to zero according to block-based similarities between a reference frame and successive frames. Encoding is carried out by quantizing and arithmetic encoding the wavelet coefficients in preservation regions and replacement regions separately. A reference frame is updated at the bottom point If the curve of similarity rates looks like concave pattern. Simulation results show that the proposed algorithm provides high compression ratio with proper Image quality. It also outperforms the previous Milton's algorithm in an Image quality, compression ratio and running time, leading to compression ratio less than 0.2bpp. PSNR of 32 dB and running tome of 10ms for a standard video image of size 352${\times}$240 pixels.

• The Journal of the Acoustical Society of Korea
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• v.39 no.5
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• pp.441-446
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• 2020

In this paper, we propose the α-Feature Map Scaling (α-FMS) method which extends the FMS method that was designed to enhance the discriminative power of feature maps of deep neural networks in Speaker Verification (SV) systems. The FMS derives a scale vector from a feature map and then adds or multiplies them to the features, or sequentially apply both operations. However, the FMS method not only uses an identical scale vector for both addition and multiplication, but also has a limitation that it can only add a value between zero and one in case of addition. In this study, to overcome these limitations, we propose α-FMS to add a trainable parameter α to the feature map element-wise, and then multiply a scale vector. We compare the performance of the two methods: the one where α is a scalar, and the other where it is a vector. Both α-FMS methods are applied after each residual block of the deep neural network. The proposed system using the α-FMS methods are trained using the RawNet2 and tested using the VoxCeleb1 evaluation set. The result demonstrates an equal error rate of 2.47 % and 2.31 % for the two α-FMS methods respectively.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.4
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• pp.97-104
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• 2010

In this paper, we provide an embedded type non-contact bio-radar heart and respiration rate monitoring system. We implemented the rate finding algorithm into the embedded system. The high-speed and reliable real-time signal processor is then tested. To avoid null-point data loss problem, we applied quadrature demodulation. Among several other combining techniques, we suggest arctangent demodulation for quadrature channel combining and DSP is used for real-time signal processing. We also suggest DC-offset compensation technique to preserve the wanted DC components of the IQ signals for accurate demodulation while keeping the dynamic range of the ADC lower. Using Texas Instrument C6711 series DSP and external 12Bit ADC, we implemented proper elliptic digital filter and autocorrelation detection algorithm for robust commercial hand held device.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.95-101
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• 2006

A face detection plays an important role in face recognition, video surveillance, and human computer interaction. In this paper, we propose a progressive threshold method to detect human faces in real time. The consecutive face images are acquired from camera and transformed into YCbCr color space images. The skin color of the input images are separated using a skin color filter in the YCbCr color space and some candidated face areas are decided by connected component analysis. The intensity equalization is performed to avoid the effect of many circumstances and an arbitrary threshold value is applied to get binary images. The eye area can be detected because the area is clearly distinguished from others in the binary image progressive threshold method searches for an optimal eye area by progressively increasing threshold from low values. After progressive thresholding, the eye area is normalized and verified by back propagation algorithm to finalize the face detection.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.46 no.8
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• pp.18-21
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• 2009

In the digital system, ROM has a large power-consumption and a speed-bottleneck. According to gradual growth of system speed, ROM is demanded to have low-power consumption and high-speed operation design. The ROM adapted in FFT or FIR filter needs method of sequential accessed addressing. We proposed a reduction method for the number of storage cells in this paper. The number of storage cells which is connected with bi-line is reduced by the proposed method so that the capacitance value of bit-time is reduced. In this case, delay time, and power consumption are reduced. Design result of ROM in this paper using the proposed method could reduce up to 86.3% of storage cell '1' compare with conventional method.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.27 no.4
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• pp.923-941
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• 2017

A side-channel attack is a method of attacking a cipher based on physical information of a cryptographic device. The masking method, which is a typical method overcoming this attack, is a method of calculating an arbitrary masking value at the round intermediate value through rounds. Thus, it is difficult to guess the intermediate value by the side-channel attack, but if the masking operation is applied to all rounds of the encryption algorithm, the encryption process may become overloaded. Therefore, it is practical to use a reduced-round masking technique that applies a masking technique to only a part of the cipher for lightweight equipment such as Internet of Things(IoT) and wearable devices. In this paper, we describe a Hamming weight filtering for SIMON family with reduced-round masking technique and it is shown that first round key recovery is possible through actual programming.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.17 no.5 s.108
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• pp.408-415
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• 2006

This paper suggests an improved system-level model of RF power amplifiers(PAs) including memory effects, and validates the suggested model by analyzing the power spectral density of the output signal with a predistortion linearizer. The original three-box(Wiener-Hammerstein) model uses input and output filters to capture RF frequency response of PAs. The adjacent spectral regrowth that occurs in three-box model can be perfectly removed by Hammerstein structure predistorter. However, the predistorter based on Hammerstein structure achieves limited performance in real PA applications due to other memory effects except RF frequency response. The spectrum of the output signal can be predicted accurately using the suggested model that changes a memoryless block in a three-box model with a memory polynomial. The proposed model accurately predicts the output spectrum density of PA with Hammerstein structure predistorter with less than 2 dB errors over ${\pm}30$ MHz adjacent channel ranges for IEEE 802.11 g WLAN signal.