• Journal of Astronomy and Space Sciences
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• v.26 no.4
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• pp.567-588
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• 2009

Korea-Japan Joint VLBI Correlator (KJJVC) is being developed by collaborating KASI (Korea Astronomy and Space Science Institute), Korea, and NAOJ(National Observatory of Japan), Japan. In early 2010, KJJVC will work in normal operation. In this study, we developed the software correlator which is based on VCS (VLBI Correlation Subsystem) hardware specification as the core component of KJJVC. The main specification of software correlator is 8 Gbps, 8192 output channels, and 262,144-points FFT (Fast Fourier Transform) function same as VCS. And the functional algorithm which is same as specification of VCS and arithmetic register are adopted in this software correlator. To verify the performance of developed software correlator, the correlation experiments were carried out using the spectral line and continuum sources which were observed by VERA (VLBI Exploration of Radio Astrometry), NAOJ. And the experimental results were compared to the output of Mitaka FX correlator by referring spectrum shape, phase rate, and fringe detection and so on. Through the experimental results, we confirmed that the correlation results of software correlator are the same as Mitaka FX correlator and verified the effectiveness of it. In future, we expect that the developed software correlator will be the possible software correlator of KVN (Korean VLBI Network) with KJJVC by introducing the correlation post-processing and modifying the user interface as like GUI (Graphic User Interface).

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.6
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• pp.884-889
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• 2010

Recently, a music genre classification has been studied. However, experts use different criteria to classify each of these classifications is difficult to derive accurate results. In addition, when the emergence of a new genre of music genre is a newly re-defined. Music as a genre rather than to separate search should be classified as emotional words. In this paper, the feelings of people on the basis of brightness and darkness tries to categorize music. The proposed classification system by applying VCM(Variance Considered Machines) is the contrast of the music. In this paper, we are using three kinds of musical characteristics. Based on surveys made throughout the learning, based on musical attributes(beat, timbre, note) was used to study in the VCM. VCM is classified by the trained compared with the results of the survey were analyzed. Note extraction using the MATLAB, sampled at regular intervals to share music via the FFT frequency analysis by the sector average is defined as representing the element extracted note by quantifying the height of the entire distribution was identified. Cumulative frequency distribution in the entire frequency rage, using the difference in Timbre and were quantified. VCM applied to these three characteristics with the experimental results by comparing the survey results to see the contrast of the music with a probability of 95.4% confirmed that the two separate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8A
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• pp.889-897
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• 2004

In this Paper, we propose the algorithm to provide stable communication in OFDM system under the highly interfered environment by the same/different systems which use same bandwidth or other jamming signal, i.e., radar signal. The proposed Adaptive Subcarrier Assignment(ASA) method first estimates the received power of each subcarrier in the block or fin or OFDM receiver. Then we estimate the threshold level which is the average power of the transmitted OFDM signal with AWGN. The highly interfered subcarriers, which are greater powers than the specified threshold level, are rejected in the next transmission and the only non-interfered subcarriers are selected as the next transmission. This algorithm provides stable communication in any OFDM systems without changing the physical layer under the highly interfered communication environment. We estimated the status of the subcarriers based on the bandwidth and power of the jamming signal and showed the performance of the proposed algorithm by the simulation.

• Journal of Korea Multimedia Society
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• v.14 no.11
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• pp.1431-1437
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• 2011

This paper proposes a novel watermarking algorithm that protects digital copyrights and is robust to attacks. Watermarks are embedded in the subband including the significant part of the signal such as a pitch. Generally, the subband containing the pitch has the biggest energy. In order to find this subband, wavelet packet transform is used to decompose the subbands and their energy are calculated. The signal of the selected subbands is transformed in frequency domain using FFT. The watermarks are embedded using QIM for samples higher than a certain threshold. The blind detection uses the Euclidean distance. The proposed method shows less than 5% BER in the audio watermark benchmarking.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.822-829
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• 2009

This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

• Journal of the Korean Institute of Landscape Architecture
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• v.24 no.4
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• pp.74-84
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• 1997

최근. 급격한 도시화 현상으로 인한 불투성 지표의 증가는 도시의 생태적 측면에 변화를 일으키고 있으며, 도시의 온난화 현상. 도시의 사막화 현상 등이 그 결과로 나타나고 있다. 특히, 분투수성 지표가 많은 도시에서는 그 수문학적 기능이 투수성 지표에 의존하고 있으며, 도시의 투수성 지표에는 오픈 스페이스와 주변의 생산녹지 등이 있다 이러한 관점에서 투수성 지표에 관한 연구가 필요함에도 불구하고 많지 않았고, 대부분 포인트별 연구였다. 본 연구는 도시 내의 수문기능 환경변화개선에 관한 기초자료로 동경의 대표적 도시공원인 코카네이 (소금정) 공원과 동경 근교의 치바현 (천엽현)위치한 치바(천엽)대학 부속 카시와 (상) 농장을 대상지로 선정 토양환경기능의 간접적 지표가 되는 토양경도를 나카야마식(산중식) 토양경도 계륵 사용하여 5120 Cm의 라인위에 10 Cm간격으로 512개씩 코카네이(소금정) 공원에서 9라인. 카시와(상) 농장에서 7라인을 공간별로 측정하고. 그 측정값을 다중비교검정과 Bartlett's 검정에 의하여 검정후에, 랜덤 데이타분석에 적합한 FFT(Fast Fourier Transform)를 사용하여 분석하였다 본 연구의 측정값을 다중비교검정과 Bartlett's 검정한 결과, 유의타가 없었고, 각 데이터를 분석한 결과. 공원의 9라인과 농장의 7라인은 분석 그래프의 유형에 의해 각각 2가지로 분류되었고, 전체적으로는 3가지로 분류할 수 있었다 특히 공원과 농장의 분석 그래프에서는 수평 방향으로의 변화에 대한 수직방향의 변화가 비슷한 스펙트럼이 공통적으로 나타났고. 각각 최대 값은 다르나 제1주기 부분에서는 답압과 같은 인위적인 요인에 의한 논은 수직방향의 스펙트럼 변화가 관찰되었다. 또한, 강우가 녹지의 경도변화에 미치는 가를 관찰하기 위하여 강우전과 강우후에 같은 라인(Line1과 Line2)을 측정하여 분석한 결과, 측정값의 평균에서는 차이를 보였으나. 주기의 변화는 거의 없었다. 전체적으로 분석 그래프에서는 공간별로 스펙트럼의 수평방향에 대한 수직방향의 변화량의 차이는 보였으나 비슷한 주기를 나타냈고, 가작의 유형은 공원의 녹지공간(lawn-area)과 나지공간(bare-area)으로, 농장을 녹지공간(lawn-area)과 경작공간(field-area)으로 분류할 수있었다. 종합적으로 녹지를 지질학적 구조의 고유한 특징이 아닌. 답압 등과 같은 인위적인 요인에 따른 속성들에 의하여 나타나는 스펙트럼의 분석을 통하여 녹지공간(lawn-area). 나지공간(bare-area)그리고, 경작공간(field-area)으로 분류할 수 있었다..

• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.158-162
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• 2009

Sound effects coming with appropriate tactile stimuli can strengthen its reality. For example, gunfire in games and movies, if it is accompanied by vibrating effects, can enhance the impressiveness. On a similar principle, adding the vibration information to existing sound data file and playing sound while generating vibration effects through haptic interfaces can augment the sound experience. In this paper, we propose a method to generate vibration information by analyzing the sound. The vibration information consists of vibration patterns and the timing within a sound file. Adding the vibration information is labor-intensive if it is done manually. We propose a sound detection algorithm to search the moments when specific sounds occur in a sound file and a method to create vibration effects at those moments. The sound detection algorithm compares the frequency characteristic of specific sounds and finds the moments which have similar frequency characteristic within a sound file. The detection ratio of the algorithm was 98% for five different kinds of gunfire. We also develop a GUI based vibrating pattern editor to easily perform the sound search and vibration generation.

• KIPS Transactions on Software and Data Engineering
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• v.7 no.3
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• pp.113-120
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• 2018

In this study, we proposed a monitoring system for identifying and handling faulty sensing stream data on manufacturing equipments where low-cost sensors can be safely used. Low cost sensors will lessen the cost of implementing distributed monitoring system, but suffer from sensor noises and inaccurate sensed data. Therefore, a distributed monitoring system with low cost sensors should identify faulty signal data as either of sensor fault or machine fault, and filter out faulty signals from sensing fault. To this end, we adopted a fourier transform based diagnostic approach mixed with a weighed moving averaging method, in order to identify faulty signals. We measured how effective our approach is and found out our approach can filter out one-third faulty signals from our experimental environment. In addition, we attached wireless communication modules to reduce sensor and network installation cost. To handle massive sensor data efficiently, we employed unstructured data format with NoSQL based database.

• Journal of KIISE:Software and Applications
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• v.34 no.3
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• pp.226-234
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• 2007

In this paper, we develop a speech recognition system using a throat microphone. The use of this kind of microphone minimizes the impact of environmental noise. However, because of the absence of high frequencies and the partially loss of formant frequencies, previous systems developed with those devices have shown a lower recognition rate than systems which use standard microphone signals. This problem has led to researchers using throat microphone signals as supplementary data sources supporting standard microphone signals. In this paper, we present a high performance ASR system which we developed using only a throat microphone by taking advantage of Korean Phonological Feature Theory and a detailed throat signal analysis. Analyzing the spectrum and the result of FFT of the throat microphone signal, we find that the conventional MFCC feature vector that uses a critical pass filter does not characterize the throat microphone signals well. We also describe the conditions of the feature extraction algorithm which make it best suited for throat microphone signal analysis. The conditions involve (1) a sensitive band-pass filter and (2) use of feature vector which is suitable for voice/non-voice classification. We experimentally show that the ZCPA algorithm designed to meet these conditions improves the recognizer's performance by approximately 16%. And we find that an additional noise-canceling algorithm such as RAST A results in 2% more performance improvement.