• Journal of the Korean Society for Precision Engineering
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• v.12 no.5
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• pp.126-135
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• 1995

An exhaust muffler is one of the simple and effective means to meet the demand for a quiet vehicle, and it deserves a close attention to effectively reduce the engine noise. The transfer function technique is one of the tools that have been used to analyze the noise characteristics of the exhaust muffler. In this paper we obtained a transfer function using the forward-going and backward-going components of sound pressure in the exhaust muffler, which is compatible wiht lattice filter algorithm. This form of transfer function is obtained for the basic elements of a muffler, such as uniform tube, open termin- ation, closed termination, anechoic termination, expansion, contraction, extended-tube resonator, hole, Helmholtz resonator, and concentric hole-cavity resonator. The results are combined to produce the transfer function of various types of mufflers. With this transfer function we calculate the transmission and insertion losses of mufflers, and examined the effects of various design parameters. Comparisons were made between the calculation and experimental results, which showed a good agreement, and we conclude that the transfer function of lattice form can be used to analyze the noise characteristics of the exhaust mufflers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.9C
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• pp.761-770
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• 2010

In this paper, we propose a new Frame Rate Up-Conversion (FRUC) scheme to increase the frame rate from a lower number into a higher one and enhance the decoded video quality at the decoder. The proposed algorithm utilizes the preliminary frames of forward and backward direction using bilateral prediction. In the process of the preliminary frames, an additional interpolation is performed for the occlusion area because if the calculated value of the block with reference frame if larger than the predetermine thresholdn the block is selected as the occlusion area. In order to interpolate the occlusion area, we perform re-search to obtain the osiomal block considerhe osiomnumber of available ne block consblock. The experimental results show that performance of the proposed algorithm has better PSNR and visual quality than the conventional methods.

• Journal of Information Processing Systems
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• v.18 no.4
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• pp.549-561
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• 2022

In the Khmer writing system, the Khmer script is the official letter of Cambodia, written from left to right without a space separator; it is complicated and requires more analysis studies. Without clear standard guidelines, a space separator in the Khmer language is used inconsistently and informally to separate words in sentences. Therefore, a segmented method should be discussed with the combination of the future Khmer natural language processing (NLP) to define the appropriate rule for Khmer sentences. The critical process in NLP with the capability of extensive data language analysis necessitates applying in this scenario. One of the essential components in Khmer language processing is how to split the word into a series of sentences and count the words used in the sentences. Currently, Microsoft Word cannot count Khmer words correctly. So, this study presents a systematic library to segment Khmer phrases using the bi-directional maximal matching (BiMM) method to address these problematic constraints. In the BiMM algorithm, the paper focuses on the Bidirectional implementation of forward maximal matching (FMM) and backward maximal matching (BMM) to improve word segmentation accuracy. A digital or prefix tree of data structure algorithm, also known as a trie, enhances the segmentation accuracy procedure by finding the children of each word parent node. The accuracy of BiMM is higher than using FMM or BMM independently; moreover, the proposed approach improves dictionary structures and reduces the number of errors. The result of this study can reduce the error by 8.57% compared to FMM and BFF algorithms with 94,807 Khmer words.

• Journal of Convergence for Information Technology
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• v.12 no.2
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• pp.23-29
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• 2022

Recently, due to the increase in the use of Internet of Things (IoT) devices, the amount of data transmitted and processed to cloud computing servers has increased rapidly. As a result, network problems (delay, server overload and security threats) are emerging. In particular, edge computing with lower computational capabilities than cloud computing requires a lightweight authentication algorithm that can easily authenticate numerous IoT devices.In this paper, we proposed a key-agreement protocol of a lightweight algorithm that guarantees anonymity and forward and backward secrecy between IoT and edge devices. and the proposed algorithm is stable in MITM and replay attacks for edge device and IoT. As a result of comparing and analyzing the proposed key-agreement protocol with previous studies, it was shown that a lightweight protocol that can be efficiently used in IoT and edge devices.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.4
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

• The Journal of Korea Robotics Society
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• v.15 no.4
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• pp.309-315
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• 2020

This study shows a control strategy that acquires both precision and manipulation sensitivity of remote center motion with manual traction for a surgical assistant robot. Remote center motion is an essential function of a laparoscopic surgical robot. The robot has to keep the position of the insertion port in a three-dimensional space, and general laparoscopic surgery needs 4-DoF (degree-of-freedom) motions such as pan, tilt, spin, and forward/backward. The proposed robot consists of a 6-axis collaborative robot and a 2-DoF end-effector. A 6-axis collaborative robot performs the cone-shaped trajectory with pan and tilt motion of an end-effector maintaining the position of remote center. An end-effector deals with the remaining 2-DoF movement. The most intuitive way a surgeon manipulates a robot is through direct teaching. Since the accuracy of maintaining the remote center position is important, direct teaching is implemented based on position control in this study. A force/torque sensor which is attached to between robot and end-effector estimates the surgeon's intention and generates the command of motion. The predefined remote center position and the pan and tilt angles generated from direct teaching are input as a command for position control. The command generation algorithm determines the direct teaching sensitivity. Required torque for direct teaching and accuracy of remote center motion are analyzed by experiments of panning and tilting motion.

• Journal of Korea Society of Industrial Information Systems
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• v.5 no.3
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• pp.117-124
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• 2000

A systolic array is a parallel processing system which consists of processing elements of basic computation capabilities, connected with regular and local communication lines. It has been known that a systolic array is on of effective systems to solve complicated communication problems occurred between densely connected neurons on ANN(Artificial Neural Network). In this paper, a systolic array simulator for the back-propagation ANN, which automatically constructs the proper systolic array for a given number of neurons of the ANN, is designed and constructed. With animation techniques of the simulators, it is easy for users to be able to examine the execution of the back-propagation algorithm on the designed systolic array step by step. Moreover the simulator can perform forward and backward operations of the back-propagation algorithm either in sequence or in parallel on the designed systolic array. Parallel execution can be performed by feeding continuous input patterns and by executing bidirectional propagations on all of processing elements of a systolic array at the same time.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.15 no.10
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• pp.3815-3833
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• 2021

MILP-based automatic search is the most common method in analyzing the security of cryptographic algorithms. However, this method brings many issues such as low efficiency due to the large size of the model, and the difficulty in finding the contradiction of the impossible differential distinguisher. To analyze the security of ESF algorithm, this paper introduces a simplified MILP-based search model of the differential distinguisher by reducing constrains of XOR and S-box operations, and variables by combining cyclic shift with its adjacent operations. Also, a new method to find contradictions of the impossible differential distinguisher is proposed by introducing temporary variables, which can avoid wrong and miss selection of contradictions. Based on a 9-round impossible differential distinguisher, 15-round attack of ESF can be achieved by extending forward and backward 3-round in single-key setting. Compared with existing results, the exact lower bound of differential active S-boxes in single-key setting for 10-round ESF are improved. Also, 2108 9-round impossible differential distinguishers in single-key setting and 14 12-round impossible differential distinguishers in related-key setting are obtained. Especially, the round of the discovered impossible differential distinguisher in related-key setting is the highest, and compared with the previous results, this attack achieves the highest round number in single-key setting.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.9A
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• pp.1290-1298
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• 2000

Turbo Code decoder is an iterate decoding technology, which extracts extrinsic information from the bit to be decoded by calculating both forward and backward metrics, and uses the information to the next decoding step Turbo Code shows excellent performance, approaching Shannon Limit at the view of BER, when the size of Interleaver is big and iterate decoding is run enough. But it has the problems which are increased complexity and delay and difficulty of real-time processing due to Interleaver and iterate decoding. In this paper, it is analyzed that MAP(maximum a posteriori) algorithm which is used as one of Turbo Code decoding, and the factor which determines its performance. MAP algorithm proceeds iterate decoding by determining soft decision value through the environment and transition probability between all adjacent bits and received symbols. Therefore, to improve the performance of MAP algorithm, the trust between adjacent received symbols must be ensured. However, MAP algorithm itself, can not do any action for ensuring so the conclusion is that it is needed more algorithm, so to decrease iterate decoding. Consequently, MAP algorithm and Turbo Code performance are analyzed in the nongaussian channel applying Robust equalization technique in order to input more trusted information into MAP algorithm for the received symbols.

• Journal of Broadcast Engineering
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• v.17 no.1
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• pp.108-121
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• 2012

As Scalable Video Coding (SVC) is a video compression standard extended from H.264/AVC, it is a way to provide scalability in terms of temporal, spatial and quality. Although the compression efficiency of SVC is increased due to the scalability in many aspect, it is essential to reduce the complexity in order to efficiently use because the complexity is relatively increased. To reduce the complexity of SVC in the paper, we propose fast mode decision algorithm to reduce the complexity of encoding process using direction information of B-picture by efficiently performing inter-layer prediction. The proposed algorithm is a fast mode decision algorithm that makes different from detection mode number of forward and backward, bi-direction in the way using best mode of base-layer up-sampled after simply SKIP mode detection using the direction information of best mode of base-layer up-sampled. The experimental results show that the proposed algorithm approach can achieve the maximum computational time saving about 53% with almost no loss of rate distortion (RD) performance in the enhancement layer.