• 한국인터넷방송통신학회논문지
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• 제17권6호
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• pp.69-78
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• 2017

5G 스마트폰의 샤논과 신호처리의 푸리에가 표본화정리(최고 주파수의 2배분1 즉, $\frac{1}{2f_n}=T$)에서 만난다. 본 논문에서는 초기 샤논 정리가 Point-to-Point에서 샤논 용량을 구했지만 5G는 Multi point MIMO로 기술이 발전했음을 Relay 채널에서 보인다. 푸리에 변환은 고정매개변수로 신호처리를 했는데, 멀티미디어 시대에 2N-1 다변수인 푸리에-Jacket 변환을 제안해서 성능을 분석했다. 이 연구에서 저자는 시간 계산 측면에서 프리 코딩 / 디코딩 복잡성을 줄이기위한 Jacket 기반의 빠른 방법을 제안함으로써 신호 처리의 복잡성 문제를 해결한다. 재킷 변환은 신호 처리 및 코딩 이론에서 응용 프로그램을 찾는 것으로 나타냈다. 재킷 변환은 속성 $AA^{\dot{+}}=nl_n$이 있는 필드 F에 대해 $n{\times}n$ 행렬 $A=(a_{jk})$로 정의되며, 여기서 $A^{\dot{+}}$는 A의 원소 역행렬의 전치 행렬, 즉 $A^{\dot{+}}=(a^{-1}_{kj})$이며, 이는 변환을 일반화하고 중심 가중 변환, 특히 재킷 변환 특성을 이용하여, 저자는 전송 기반의 중계 기반 DF 협동 무선 네트워크에서 분산 다중 입력 다중 출력 채널의 프리 코딩 및 디코딩에 적용하여 새로운 고유치 분해 (EVD : eigenvalue decomposition) 방법을 제안한다. 단일 심볼 디코딩 가능한 시공간 블록 코드를 사용한다. 본 논문은은 제안 된 Jacket 기반 EVD 방법이 기존의 EVD 방법에 비해 계산 시간이 현저히 단축되었다. 계산 시간 단축과 관련된 성능은 수학적 분석 및 수치결과를 통해 정량적으로 평가했다.

• The Journal of the Acoustical Society of Korea
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• 제17권2E호
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 춘계학술대회논문집
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• pp.774-779
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• 2004

This paper addresses the way in which we can find where impact noise sources are. Specifically, we have an interest in the case that the signal is embedded in noise. We propose a signal processing method that can identify impulsive sources’location. The method is robust with respect to noise; spatially distributed noise. This has been achieved by a beamforming method with regard to cepstrum domain is used. It is noteworthy that the cepstrum has the ability to detect periodic pulse signal in noise. Numerical simulation and experiments are performed to verify the method. Results show that the proposed technique is quite powerful for localizing the faults in noisy environments. The method also required less microphones than conventional beamforming method.

• 한국소음진동공학회논문집
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• 제14권9호
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• pp.877-883
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• 2004

This paper addresses the way in which we can find where impulsive noise sources are. Specifically, we have an interest in the case that the signal is embedded in noise. We propose a signal processing method that can identify impulsive sources' location. The method is robust with respect to spatially distributed noise. This has been achieved by the modified beamforming method with regard to cepstrum domain is used. It is noteworthy that the cepstrum has the ability to detect periodic pulse signal in noise. Numerical simulation and experiments are performed to verify the method. Results show that the proposed technique is quite powerful for localizing the faults in noisy environments. The method also required less microphones than conventional beamforming method.

• 융합신호처리학회논문지
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• 제16권1호
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• pp.1-8
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• 2015

In many image processing tasks, connected component labeling (CCL) is performed to extract regions of interest. CCL was usually done in a sequential fashion when image resolution was relatively low and there are small number of input channels. As image resolution gets higher up to HD or Full HD and as the number of input channels increases, sequential CCL is too time-consuming to be used in real time applications. To cope with this situation, parallel CCL framework was introduced where multiple cores are utilized simultaneously. Several parallel CCL methods have been proposed in the literature. Among them are NSZ label equivalence (NSZ-LE) method[1], modified 8 directional label selection (M8DLS) method[2], and HYBRID1 method[3]. Soh [3] showed that HYBRID1 outperforms NSZ-LE and M8DLS, and argued that HYBRID1 is by far the best. In this paper we propose an improved hybrid parallel CCL algorithm termed as HYBRID2 that hybridizes M8DLS with label backtracking (LB) and show that it runs around 20% faster than HYBRID1 for various kinds of images.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 학술대회지
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• pp.897-899
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• 2004

There is scientific work containing the recurrence method of synthesis the new class of orthogonal circle m-invariant signals: designed effective algorithms of fast-direct computing m-convolution in time domain: engineer methods of design economic scheme of decoders for optimal receiving in aggregate of suggested signal.

• 한국인쇄학회지
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• 제22권1호
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• pp.9-18
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• 2004

Recently as color image processing to be become independent have been desired at the light source in an image processing and it have been enlarged. An image processing of the light source which is become independent means an image processing which uses a spectral reflectance information. We improved it in the spectral reflectance estimation method which uses existing 3-band image in this research that the improvement of an identity color population generation method which uses the hue angle and the processing speed improvement and introduces a labelling method. The precision of a spectral reflectance estimation appeared to the ${\Delta}E^*_{ab}$ of an average 2.7 comparing with the measurement price. The practical use possibility came to be fast and appeared a processing speed compared with existing method.

• 한국광학회:학술대회논문집
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• 한국광학회 1998년도 제15회 광학 및 양자전자 학술발표회 논문집
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• pp.210-211
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• 1998

• 한국정밀공학회지
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• 제29권10호
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• pp.1096-1100
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• 2012

In the phase-shift measurement method, the distance light travels can be obtained based on the phase difference between the reference signal and the measured signal. When the object having various colors is measured, the intensity of the measured signal much varies even at the same distance, and it causes different phase delay due to wide dynamic range input to a signal processing circuit. In this work, an measured intensity control method is proposed to solve this phase delay problem.

• 한국마린엔지니어링학회:학술대회논문집
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• 한국마린엔지니어링학회 2002년도 춘계학술대회논문집
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• pp.219-223
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• 2002

Voltage and current signals during impulse tests on transformer are treated as non-stationary signals. A new method incorporating signal-processing method such as Wavelets and courier transform is proposed for failure identification. It is now possible to distinguish failure during impulse tests. The method is experimentally validated on a transformer winding. The wavelet transforms enables the detection of the time of occurrence of switching or failure events. After establishing the time of occurrence, the original waveform is split into two or more sections. The wavelet transform has ability to analysis the failure signal on time domain as well as frequency domain. Therefore, the wavelet transform is superior than courier transform to analysis the failure signal. In this paper, the fact was proved by real data which was achieved.