• Journal of the Korea Computer Graphics Society
• /
• v.4 no.1
• /
• pp.21-29
• /
• 1998

This paper deals with character animation using motion capture. The captured motion requires the editing process to smooth the jerky motion by the sensor noise, or to combine several clip-motion libraries. For this purpose, we describe a simple technique for editing the captured motion using the Kalman filter technique. Our formulation allows the generated motion to satisfy the kinematic constraints of the human model. Furthermore, it provides us with a multi-level control mechanism of the motion resolution by changing the uncertainty of the measurement model and the seamless motion transition.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06e
• /
• pp.411-414
• /
• 1998

음성의 음질 향상(Speech Enhancement)을 위한 여러 가지 방법 중에서 주파수 차감법(Spectral Subtraction)은 계산량이 적기 때문에 현재 실시간으로 Speech Enhancement를 할 수 있는 가장 적절한 방법이다. 그러나, 이 방법은 원래의 입력음성에 없던 새로운 잡음을 만들어내는 큰 단점이 있는데, 이를 제거하기 위해 많은 연구가 되어오고 있다. 이러한 연구의 방향은 대부분 주변프레임 또는 주변의 주파수 성분과의 평균을 통해 피크값을 무디게 해 줌으로써 새로 생긴 튀는 잡음을 감소시키는 것이다. 이런 방법은 음성자체의 정보 또한 평균이 되어버리게 하는 새로운 단점을 낳는데, 이런 현상은 무성음구간에서 특히 심각해진다. 본 논문에서는 입력음성의 LPC 분석으로 백색필터(Whitening Filter)를 구성하여 이를 통과시킨 잔류신호(Residual)를 주파수 차감하여 얻은 새로운 잔류신호를 역 필터링하여(Synthesis Filter) 개선된 음성을 얻는 방법을 제안하였다. 제안된 알고리듬은, 주파수 차감시 포만트(Formant)의 정보가 더 유지 될 수 있기 때문에 잔류잡음을 줄일 수 있다. 청취 테스트 결과 제안한 방법이 기존의 방법보다 잔류잡음을 더 줄이는 사실을 확인할 수 있었다.

• Journal of the Korean Institute of Telematics and Electronics C
• /
• v.34C no.10
• /
• pp.1-7
• /
• 1997

This paper proposes a new real-time 2-D convolver filter architecture wihtout using any multiplier. To meet the massive amount of computations for real-time image processing, several commercial 2-D convolver chips have many multipliers occupying large VLSI area. Te proposed architecture using only one shift-and-accumulator can reduce the chip size by more than 70% of commercial 2-D convolver filter chips and can meet the real-time image processing srequirement, i.e., the standard of CCIR601. In addition, the proposed chip can be used for not only 2-D image processing but also 1-D signal processing and has bood scalability for higher speed applications. We have simulated the architecture by using VHDL models and have performed logic synthesis. We used the samsung SOG cell library (KG60K) and verified completely function and timing simulations. The implemented filter chip consists of only 3,893 gates, operates at 125 MHz and can meet the real-time image processing requirement, that is, 720*480 pixels per frame and 30 frames per second (10.4 mpixels/second).

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.25 no.8
• /
• pp.525-531
• /
• 2015

Driving sound is an effective element to promote the product identity of a vehicle by providing customers with attractive sound which reflects the concept of a vehicle. Recently, major automakers are focusing on the target sound setting so that the sound can represent the brand image as well as the unique concept of a vehicle. In this study, a new method of target setting for the driving sound will be introduced based on using formant filter and musical harmonics characteristics. In addition, a target sound suggested from this method will be realized and verified by using active noise control in vehicle.

• Proceedings of the KIEE Conference
• /
• 2007.10a
• /
• pp.107-108
• /
• 2007

In this paper, a cosine-modulated filter bank with a modified synthesis part is proposed for arbitrary scaling of images, whereby down/up-sampling kernels based on a compactly supported sampling function (CSSF) are utilized. Also, an optimized adaptive interpolation technique is incorporated into the filter bank structure to compensate for quality degradation arising in scaled images. Finally, simulation results verify that high quality images with arbitrary sizes can be obtained by applying the proposed approach.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1994.06a
• /
• pp.1021-1026
• /
• 1994

This paper proposes a filter structure suitable for speech synthesis applications. We first derive the lossy pole-zero model by employing the wave digital filter(WDF) adaptor formula, and by converting the fixed termination value - 1 into a loss factor $\mu$c$\in$(-1, 1). Then we discuss how to determine the reflection We employ the Durbin's method in estimating the numerator polynomial of the lossy pole-zero transfer function from the given speech sound, and then apply the step-down algorithm on the numerator to extract the reflection coefficients of the closed-termination tract. For determining the reflection coefficients of the other parts we employ a pre-calculated pole-estimator polynomial.

• Bulletin of the Korean Chemical Society
• /
• v.31 no.9
• /
• pp.2727-2730
• /
• 2010

• Bulletin of the Korean Chemical Society
• /
• v.32 no.3
• /
• pp.1033-1036
• /
• 2011

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.21 no.3
• /
• pp.629-635
• /
• 2017

In this paper, we proposed a nonlinear synthesis filter for noise reduction to reduce the effects of Gaussian noise and impulse noise. When the centralization of the local mask is judged to be Gaussian noise by the noise judgment, the weight value of the weight filter are applied differently according to the spatial weight filter and the pixel change by using the sample variance in the local mask. And if it is determined as the impulse noise, we proposed an algorithm that applies different weights of local histogram weight filter and standard median filter according to noise density of mask. In order to evaluate the performance of the proposed filter algorithm, we used PSNR(peak signal to noise ratio) and compared existing methods and proposed filter algorithm in the mixed noise environment with Gaussian noise, impulsive noise, and two noises mixed.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.3
• /
• pp.49-60
• /
• 1995

This paper is a study on the design and implementation of the Korean Text-to-Speech system which is used for a small and simple system. In this paper, a parameter synthesis method is chosen for speech syntheiss method, we use PARCOR(PARtial autoCORrelation) coefficient which is one of the LPC analysis. And we use phoneme for synthesis unit which is the basic unit for speech synthesis. We use PARCOR, pitch, amplitude as synthesis parameter of voice, we use residual signal, PARCOR coefficients as synthesis parameter of unvoice. In this paper, we could obtain the 60% intelligibility by using the residual signal as excitation signal of unvoiced sound. The result of synthesis experiment, synthesis of a word unit is available. The controlling of phoneme duration is necessary for synthesizing of a sentence unit. For setting up the synthesis system, PC 486, a 70[Hz]-4.5[KHz] band pass filter for speech input/output, amplifier, and TMS320C30 DSP board was used.