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궤적 민감도와 LMI를 이용한 강인 차량 전복 방지 제어기 설계
임성진(Seong-jin Yim),박영진(Young-jin Park),박윤식(Youn-sik Park) 한국자동차공학회 2006 한국자동차공학회 춘 추계 학술대회 논문집 Vol.- No.-
This paper presents a method to design a robust controller for rollover prevention with trajectory sensitivity and linear matrix inequalities(LMIs). Active suspension is used to prevent rollover through controlling the height and roll stiffness of vehicles But, this approach has drawback in loss of directional stability. To overcome this drawback, active yaw moment controller is presented. To cope with parameter uncertainties in vehicle systems, a design method for robust controller based on trajectory sensitivity minimization and LMI is presented. Through the simulation, the newly proposed method is shown to be effective in preventing rollover and coping with parameter uncertainties.
실내 환경에서 쾌적성 향상을 위한 Vector Base Amplitude Panning 기반의 소음제어
권병호(Kwon, Byoung-Ho),박영진(Park, Young-Jin),박윤식(Park, Youn-Sik) 한국소음진동공학회 2011 한국소음진동공학회 논문집 Vol.21 No.6
A variety of noise control methods have been developed as an interest on noise issues increases. Among them, noise control methods using masking effect, a phenomenon to reduce the ability to notice the unwanted sound by proper sound, to implement a pleasant sound environment have been studied under the name of soundscape. We proposed a novel vector base amplitude panning(VBAP) based noise control method to apply to the building environment. The proposed method could improve the amenity inside the building to reproduce the sounds with excellent masking effect on the incoming path of noise using the control speakers, considering the direction of noise source. The directional masking sounds can be generated by using VBPA technique. To verify the performance of the proposed method, we carried out the subjective test for the degree of amenity according to direction of the masking sound. Subjective test results showed that it is possible to improve the amenity inside the building by controlling the direction of masking sound considering the human's auditory characteristic.
Comparison of Head-related Transfer Function Models Based on Principal Components Analysis
황성목(Hwang, Sung-Mok),박영진(Park, Young-Jin),박윤식(Park, Youn-Sik) 한국소음진동공학회 2008 한국소음진동공학회 논문집 Vol.18 No.6
This study deals with modeling of head-related transfer functions(HRTFs) using principal components analysis(PCA) in the time and frequency domains. Four PCA models based on head-related impulse responses(HRIRs), complex-valued HRTFs, augmented HRTFs, and log-magnitudes of HRTFs are investigated. The objective of this study is to compare modeling performances of the PCA models in the least-squares sense and to show the theoretical relationship between the PCA models. In terms of the number of principal components needed for modeling, the PCA model based on HRIR or augmented HRTFs showed more efficient modeling performance than the PCA model based on complex-valued HRTFs. The PCA model based on HRIRs in the time domain and that based on augmented HRTFs in the frequency domain are shown to be theoretically equivalent. Modeling performance of the PCA model based on log-magnitudes of HRTFs cannot be compared with that of other PCA models because the PCA model deals with log-scaled magnitude components only, whereas the other PCA models consider both magnitude and phase components in linear scale.
음성 신호를 이용한 시간지연 추정에 미치는 영향들에 관한 연구
권병호(Kwon, Byoung-Ho),박영진(Park, Young-Jin),박윤식(Park, Youn-Sik) 한국소음진동공학회 2008 한국소음진동공학회 논문집 Vol.18 No.8
Since it needs the light computational load and small database, sound source localization method using time delay of arrival(TDOA method) is applied at many research fields such as a robot auditory system, teleconferencing and so on. Researches for time delay estimation, which is the most important thing of TDOA method, had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Silverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time. Also, we define the performance index(PI) to show a similar tendency to R/D ratio, and propose the method to improve the performance of time delay estimation with PI.
공간좌표로 사상된 GCC 함수를 이용한 음원 위치 추정 방법
권병호(Kwon, Byoung-Ho),박영진(Park, Young-Jin),박윤식(Park, Youn-Sik) 한국소음진동공학회 2009 한국소음진동공학회 논문집 Vol.19 No.4
Sound source localization method based on the time delay of arrival(TDOA) is applied to many research fields such as a robot auditory system, teleconferencing and so on. When multi-microphones are utilized to localize the source in 3 dimensional space, the conventional localization methods based on TDOA decide the actual source position using the TDOAs from all microphone arrays and the detection measure, which represents the errors between the actual source position and the estimated ones. Performance of these methods usually depends on the number of microphones because it determines the resolution of an estimated position. In this paper, we proposed the localization method using spatially mapped GCC functions. The proposed method does not use just TDOA for localization such as previous ones but it uses spatially mapped GCC functions which is the cross correlation function mapped by an appropriate mapping function over the spatial coordinate. A number of the spatially mapped GCC functions are summed to a single function over the global coordinate and then the actual source position is determined based on the summed GCC function. Performance of the proposed method for the noise effect and estimation resolution is verified with the real environmental experiment. The mean value of estimation error of the proposed method is much smaller than the one based on the conventional ones and the percentage of correct estimation is improved by 30% when the error bound is ${\pm}20^{\circ}$.