• Journal of KIISE:Computer Systems and Theory
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• v.27 no.7
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• pp.619-628
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• 2000

Abstract Perspective views provide a powerful depth cue and thus aid the interpretation of complicated images. The main drawback of current perspective volume rendering is the long execution time. In this paper, we present an efficient perspective volume rendering algorithm based on coherency between rays. Two sets of templates are built for the rays cast from horizontal and vertical scanlines in the intermediate image which is parallel to one of volume faces. Each sample along a ray is calculated by interpolating neighboring voxels with the pre-computed weights in the templates. We also solve the problem of uneven sampling rate due to perspective ray divergence by building more templates for the regions far away from a viewpoint. Since our algorithm operates in object-order, it can avoid redundant access to each voxel and exploit spatial data coherency by using run-length encoded volume. Experimental results show that the use of templates and the object-order processing with run-length encoded volume provide speedups, compared to the other approaches. Additionally, the image quality of our algorithm improves by solving uneven sampling rate due to perspective ray di vergence.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• Proceedings of the KIPE Conference
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• 2017.11a
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• pp.191-192
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• 2017

본 6-스텝 운전은 하드웨어의 수정 없이 운전 영역을 확장할 수 있다는 장점을 가진다. 하지만, 샘플링 주파수 대비 기본파 주파수가 크면 저주파수 전류 맥동 문제가 발생한다. 이는 시스템의 효율을 낮추고 안정성 문제를 일으킨다. 따라서, 본 논문에서는 가변 시간 제어 방법을 제안한다. 선형화과정을 통해 정확한 전달 함수를 구하고, 이를 이용하여 추가적인 약 자속 제어기나 전류 지령 생성기 없이 6-스텝 운전을 구현 할 수 있다.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04b
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• pp.97-99
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• 2001

이동객체란 시간에 따라 객체의 공간속성이 연속적으로 변하는 객체를 말한다. 기존의 데이터베이스를 이용하여 이를 관리할 경우, 단순히 객체의 현재 위치정보만을 저장하므로 과거 및 미래의 위치정보를 제공하지 못하는 단점을 가진다. 따라서 이 논문에서 제안한 이동 객체 관리 및 위치 예측기는 과거 이력질의 뿐만 아니라 샘플링되지 않은 객체의 위치를 보간법을 사용하여 계산한다. 그리고 객체의 행위정보를 저장하고 시간에 대한 함수를 사용하여 가까운 미래에 대한 이동객체의 향후위치를 사용자에게 제공하는 하는 특징을 가진다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1996.06a
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• pp.13-17
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• 1996

다수의 신호를 동시에 사용하는 경우에 나타나는 광대역 신호나 센서의 수와 형성하고자 하는 빔의 수가 많은 경우를 다루기 위해서는 주파수 영역 기법이 필요하다. FFT를 사용하는 낮은 샘플링 주파수에서도 시간 영역에서의 보간 방법보다 빠르게 정확한 시간 지연을 줄 수 있어 코히어런트 신호처리가 가능하다. 또한 특정 센서가 불량인 경우 보정이 상대적으로 용이하다는 장점을 가진다. 여러 가지 주파수 영역 빔형성 기법 중 계산량과 저장 용량면으로 효율적인 방법은 CZT와 FIR interpolation 방법이다. 또한, 공액 빔을 형성할 경우에는 Goertzel의 알고리듬을 이용하는 방법도 효율적이다.

• Journal of Internet of Things and Convergence
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• v.3 no.1
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• pp.21-24
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• 2017

Modern computers have to deal with multiple processes. The operating system uses scheduling to handle many processes with a small number of CPUs. Types of scheduling include FCFS, SJF, and RR. Of these, RR shall determine the maximum allocation time. In this paper, we analyzed the GLM algorithm for specific samples to find the optimal maximum allocation time. This analysis method allows us to specify the maximum allocation time according to the desired conditions.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.6
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• pp.114-122
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• 2011

We propose an automatic localization technique based on Uniform Arc Length Sampling (UALS) of 360 degree range sensor data. The proposed method samples 3D points from dense a point-cloud which is acquired by the sensor, registers the sampled points to a digital surface model(DSM) in real-time, and determines the location of an Unmanned Ground Vehicle(UGV). To reduce the sampling and registration time of a sequence of dense range data, 3D range points are sampled uniformly in terms of ground sample distance. Using the proposed method, we can reduce the number of 3D points while maintaining their uniformity over range data. We compare the registration speed and accuracy of the proposed method with a conventional sample method. Through several experiments by changing the number of sampling points, we analyze the speed and accuracy of the proposed method.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.29 no.2
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• pp.75-82
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• 2006

본 연구에서 조합 최적화(Combinatorial Optimization) 이론에 바탕을 두고 있는 네스티드 분할(Nested Partition, 이하 NP) 방법을 이용한 최적화 기탄 요소선택 방법(Feature Selection)을 제안한다. 이 새로운 방법은 좋은 요소 부분집합을 찾는 휴리스틱 탐색 절차를 채용하고 있으며 데이터의 인스턴스(Instances 또는 Records)의 무작위 추출(Random Sampling)을 이용하여 이 요소선택 방법의 처리시간 관점에서의 성능을 항상 시키고자 한다. 이 새로운 접근 방법은 처리시간 향상을 위해 2단계 샘플링 방법을 채용하여 근접 최적해로의 수렴(Convergence)을 보장하는 샘플 사이즈를 결정한다. 이는 앨고리듬이 유한한 시간내에 끝이날 때 최종 요소 부분집합 해의 질(Qualtiy)에 관한 정확한 설명을 할 수 있는 이론적인 배경을 제시한다. 중요 결과를 예시하기 위해서 다양한 형태의 다섯 개의 데이터 셋을 이용하였으며 다섯 번의 반복 실험을 통한 실험 결과가 제시되며, 이 새로운 접근 방법이 기존의 단순 네스티드 분할 방법 기반의 요소선택 방법보다 처리시간 관점에서 더욱 효율적임을 보여준다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.4
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• pp.49-57
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• 2012

This paper presents a fast multiple-image-based deblurring method that decreases the computation loads in the image deblurring, enhancing the sharpness of the textures or edges of the restored images. First, two blurred images with some blurring artifacts and one noisy image including severe noises are consecutively captured under a relatively long and short exposures, respectively. To improve the processing speeds, the captured multiple images are downsampled at the ratio of two, and then a way of estimating the point spread function(PSF) based on the image or edge patches extracted from the whole images, is introduced. The method enables to effectively reduce the computation time taken in the PSF prediction. Next, the texture-enhanced image deblurring method of supplementing the ability of the texture representation degraded by the downsampling of the input images, is developed and then applied. Finally, to get the same image size as the original input images, an upsampling method of utilizing the sharp edges of the captured noisy image is applied. By using the proposed method, the processing times taken in the image deblurring, which is the main obstacle of its application to the digital cameras, can be shortened, while recovering the fine details of the textures or edge components.

• The Journal of the Acoustical Society of Korea
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• v.36 no.5
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• pp.345-352
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• 2017

We propose a methods which is reducing direction estimation error of sound source in the reverberant and noisy environments. The proposed algorithm divides speech signal into voice and unvoice using VAD. We estimate the direction of source when current frame is voiced. TDOA (Time-Difference of Arrival) between microphone array using the GCC-PHAT (Generalized Cross Correlation with Phase Transform) method will be estimated in that frame. Then, we compare the peak value of cross-correlation of two signals applied to estimated time-delay with other time-delay in time-table in order to improve the accuracy of source location. If the angle of current frame is far different from before and after frame in successive voiced frame, the angle of current frame is replaced with mean value of the estimated angle in before and after frames.