• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.11-14
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• 2000

본 논문에서는 재귀적 지연추정기를 갖는 적응여파기를 이용하여 음성신호에 내재한 배경잡음을 제거하는 단일입력 적응잡음제거기를 TMS320C30 EVM 보드에서 실시간 구현하였다. 이를 위하여 샘플시간마다 지연정보를 구할 수 있는 재귀적 평균 절대차 함수를 사용하고, 정규화 된 최소평균자승(NLMS: Normalized Least Mean Square) 알고리듬을 사용하는 단일입력 잡음제거 시스템을 시뮬레이션한 (1)의 내용을 EVM 보드에 구현하였다. 그리고, (1)과 동일한 방법으로 백색 가우시안 잡음에 의해 왜곡된 음성에 대하여 SNR(Signal-to-Noise Ratio)에 따른 잡음제거 실험을 하였으며, EVM 보드에서의 실험결과를 (1)의 시뮬레이션 결과와 비교/검토하였다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.15-23
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• 2001

This paper presents a voice conversion technique that modifies the utterance of a source speaker as if it were spoken by a target speaker. Feature parameter conversion methods to perform the transformation of vocal tract and prosodic characteristics between the source and target speakers are described. The transformation of vocal tract characteristics is achieved by modifying the LPC cepstral coefficients using Linear Multivariate Regression (LMR). Prosodic transformation is done by changing the average pitch period between speakers, and it is applied to the residual signal using the LP-PSOLA scheme. Experimental results show that transformed speech by LMR and LP-PSOLA synthesis method contains much characteristics of the target speaker.

• Journal of the Korea Society for Simulation
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• v.26 no.3
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• pp.65-72
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• 2017

We propose a method to generate the object-target identifier mapping information for system performance and effectiveness analysis of submarine engage system and verify the validity of the proposed method through experiments. In the submarine model of the engage simulator, the signal processing algorithm of the actual sonar system is installed. In the target information obtained through the sonar or signal processing process, the actual object information is not known, and the simulator does not provide such information. Therefore, in this study, we generated identifier mapping information for simulation post-analysis by using bearing, range, and speed of the target obtaind from sonar signal processing and the object collected.

• Journal of Korean Ophthalmic Optics Society
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• v.19 no.4
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• pp.499-504
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• 2014

Purpose: This study is to evaluate the accuracy of dry eye screening test by comparing coincidences between three types of questionnaires and objective tests. Methods: 90 adults were classified into the normal and dry eye groups using evaluations of three types of questionnaires (SPEED, OSDI, TERTC-DEQ) and objective tests (NIBUT, Schirmer test). Coincidences-between the results of objective tests and questionnaires were compared and analyzed for dry eye diagnosis. Results: Coincidences between NIBUT test and each questionnaire were 83.3% by SPEED, 77.8% by OSDI and 72.3% by TERTC-DEQ, respectively. Concordance by SPEED was highest among three types of questionnaire. Coincidences between the Schirmer test and each questionnaire were 57.8% by SPEED, 58.9% by OSDI and 73.3% by TERTC-DEQ, respectively, thus coincidence by TERTC was higher than by others. But coincidences by Schirmer test were generally lower than those by NIBUT test. Conclusions: SPEED questionnaire showed the highest concordance with objective test as compared with others. Therefore, SPEED is expected to be used usefully to diagnose dry eyes.

• Spatial Information Research
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• v.8 no.1
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• pp.117-130
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• 2000

This paper presents a GIS-based method of spatial analysis that can be applied to the line determination in traffic infrastructure project . It is based on the spatial analysis of the sensitivity or suitability of areas relating to the integration of new infrastructure routes, carefully considering the requirement of intervention optimizing . It makes the comparison of alternatives and a final selection of a"best route " easier, offers new possibilities and simplifies and accelerates the whole process of line determination. The developed method for route-alignment is tried out by an example project for road planing in Germany. It point out that during the line determination all aspects can be equivalently and adequately evaluated. So this method is well suited as a new tool within a line determination procedure and can bring enormous advantages in the planning process of traffic infrastructure projects. projects.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.7
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• pp.1401-1410
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• 2017

Cryptographic synchronization which synchronizes internal state of cryptographic algorithm and ciphertext stream between an encryptor and a decryptor affects the quality of secure communication. If there happens a synchronization loss between a transmitter and a receiver in a secure communication, the output of the receiver is unintelligible until resynchronization is made. Especially, in the secure communication on a wireless channel with high BER, synchronization performance can dominate its quality. In this paper, we proposed a novel and noise robust synchronization signal generation method as well as its detection algorithm. We generated a synchronization signal in the form of a masking structure based on the maximal length sequence, and developed a detection algorithm using a correlation property of the maximal length sequence. Experimental results have demonstrated that the proposed synchronization signal outperforms the conventional concatenated type synchronization signal in a noisy environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1337-1344
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• 2004

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

• Journal of the Korea Society of Computer and Information
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• v.26 no.4
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• pp.85-91
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• 2021

Some sections of the highway are closed due to disasters and accidents. In this situation, it analyzes what kind of situation occurs due to functional failure in the highway network. The domestic highway network can be expressed as a graph. Blocking some sections of the highway can turn into a national disaster. In this paper, we analyze the robustness of the domestic highway network. The robustness of expressways refers to the degree to which the traffic conditions of the domestic expressway network deteriorate due to the blockage of some sections. The greater the robustness, the smaller the effect of some blocking appears. This study is used to evaluate the congestion level of one section of the transportation network, and a value obtained by dividing the section traffic volume (V) by the section traffic volume (C) is used. This study analyzes the robustness of highways by using the actual traffic volume data of the departure and arrival points of domestic highways, and analyzes the changes in traffic volume due to partial blockage through experimental calculations. Although this analysis cannot reflect the exact reality of domestic highways, it is judged to be sufficient for the purpose of confirming the basic robustness of the overall network.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.3
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• pp.304-315
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• 2001

In this paper, we peformed the real-time implementation of AMR(Adaptive Multi-Rate) speech coding algorithm which is adopted for IMT-2000 service using TMS320C6201, i.e., a Texas Instrument´s fixed-point DSP. With the ANSI C source code released from ETSI, optimization is performed to make it run in real-time with memory as small as possible using the C compiler and assembly language. Implemented AMR speech codec has the size of 32.06 kWords program memory, 9.75 kWords data RAM memory, and 19.89 kWords data ROM memory. And, The time required for processing one frame of 20 ms length speech data is about 4.38 ms, and it is short enough for real-time operation. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8B
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• pp.1460-1468
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• 2000

The goal of this research is the real-time implementation of MPEG-1 Layer III audio decoder using the fixed-point digital signal processor of TMS320C6201 The main job for this work is twofold: one is to convert floating-point operation in the decoder into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real-time with memory size as small as possible. We, especially, devote much time to the descaling module in the decoder for conversion of floating-point operation into fixed-point operation with high accuracy. The inverse modified cosine transform(IMDCT) and synthesis polyphase filter bank modules are optimized in order to reduce the amount of computation and memory size. After the optimization process, in this paper, the implemented decoder uses about 26% of maximum computation capacity of TMS320C6201. The program memory, data ROM, data RAM used in the decoder are about 6.77kwords, 3.13 kwords and 9.94 kwords, respectively. Comparing the PCM output of fixed-point computation with that of floating-point computation, we achieve the signal-to-noise ratio of more than 60 dB. A real-time operation is demonstrated on the PC using the sound I/O and host communication functions in the EVM board.