http://chineseinput.net/에서 pinyin(병음)방식으로 중국어를 변환할 수 있습니다.
변환된 중국어를 복사하여 사용하시면 됩니다.
Multi-channel Audio System Based on Human Psychoacoustic Model
Sung, Koeng-Mo 이화여자대학교 음악연구소 2004 하계 국제학술회의 자료집 Vol.2004 No.-
3-channel microphone array systems are used in multi-channel audio applications, such as home theatre systems. With microphone array signals, the loudspeakers' positions can be calculated based on the spatial information of an auditory space. Using newly proposed virtual loudspeaker positioning algorithms, we seek to reconcile disagreements in loudspeaker positioning recommendations. We also use the position information to align the levels of multiple loudspeakers for total binaural energy. Directional impulse responses are then measured by 5-channel microphone array systems, and the artificial sound field is generated with these directional impulse responses.
Reduced Complexity Self-Tuning Adaptive Algorithms in Application to Channel Estimation
Seongwook Song,Koeng-Mo Sung Institute of Electrical and Electronics Engineers 2007 IEEE transactions on communications Vol.55 No.8
<P>In this letter, reduced complexity self-tuning algorithms are proposed using simplified parameter updating procedures. Convergence analysis based on the independence assumption and the ordinary differential equation (ODE) method shows that the tuning parameter of the proposed algorithm attains the same limit as the conventional self-tuning adaptive algorithm. Simulations are carried out for channel estimation to support the analysis and performance of the proposed algorithms.</P>
Prony based Multipath Channel Parameter Estimation not Requiring the Number of Received Rays
Lim, Jun-Seok,Sung, Koeng-Mo The Acoustical Society of Korea 1996 韓國音響學會誌 Vol.15 No.e1
This paper presents an algorithm for multipath channel parameter estimation by an improved Prony method. This algorithm applies a modified regularized spectral estimation to the conventional SVD Prony method. This method requires no a priori information on the number of multipath. The performance of the proposed algorithm is almost the same as that of the SVD based multipath channel parameter estimation algorithm.
Extraction of Prosodic Information from Spoken Word for Synthesizing Korean Speech
Kim, Jin Young,Sung, Koeng-Mo 서울대학교 어학연구소 1991 語學硏究 Vol.27 No.2
This paper describes a study on prosodic information extracted from spoken Korean words, as a preliminary study to synthesize natural Korean speech. We investigated prosodic features-duration, amplitude and pitch-of some 500 words, of which selection was based on appearing frequency in Korean elementary school textbooks. Some results on word-based prosodic information are deduced.
허성욱,성굉모,Heo, Seong-Wook,Sung, Koeng-Mo 한국음향학회 1997 韓國音響學會誌 Vol.16 No.3
The most commonly used method to generate sonar transmitting beam is extracting digital samples out of memory, which are to excite transducers of the phased array respectively. As several types of signals have been used in sonar to enhance the performance of sonar in various environments, a large amount of memory is required to store them. In this paper, we adopt recursive algorithm to synthesize every different time-delayed signal for transmitting beams with small amount of memory and simple arithmetic operations. The error due to recursive calculation is also analyzed. 일반적으로 소나에서의 송신빔형성시에는 운용할 빔의 특성에 따라 array의 각 소자에 인가할 신호를 ROM에 저장한 후 송신빔을 형성하는 방법을 사용해 왔다. 이 경우 인가 신호의 길이가 길어지거나 다양한 형태의 신호를 사용하는 경우 많은 메모리가 필요하게 된다. 본 논문에서는 귀납알고리듬(recursive algorithm)을 이용하여 적은 메모리량으로 인가신호를 합성하여 송신빔을 형성하는 방안을 제시하였다. 이 신호합성방법을 사용하면 각 변환자에 인가할 신호를 신호제원에 부합하도록 계산된 초기값과 사인테이블의 값으로부터 간단히 합성할 수 있다. 본 논문에서는 운용하고자 하는 빔을 형성하기 위한 신호를 합성하는 경우의 필요 메모리와 계산량을 정량화하여 보였으며 합성한 신호와 실제 신호와의 오차를 분석하여 이의 타당성을 보였다.
Adaptive Moving Jammer Cancellation Algorithm with the Robustness to the Array Aperture
Song, Joon-il,Lim, Jun-Seok,Sung, Koeng-Mo The Acoustical Society of Korea 2004 韓國音響學會誌 Vol.23 No.e2
In moving jammer environments, the performance of conventional adaptive beamformer is severely degraded and the robust adaptive beamformer requires additional sensors to obtain desired performances. Therefore, it is necessary to develop efficient algorithm without any additional requirement of the number of sensors, etc. In this paper, we introduce a fast adaptive algorithm with variable forgetting factor, which does not have any additional requirements. From the computer simulations, we obtain the better performances than those of other techniques for the arrays with various aperture lengths.
두세진,성굉모,Doo, Se-Jin,Sung, Koeng-Mo The Acoustical Society of Korea 1994 韓國音響學會誌 Vol.13 No.e1
저주파 대역에서의 스피커 왜곡을 무향실 없이 측정할 수 있는 방법을 제안한다. 이 방법은 제 n고조파의 경우 밀폐형 스피커 내부에 비해 외부에서 40log n dB 증폭된다는 사실에 기초를 두고 있다. 인클로저 내부의 정재파에 의한 영향을 상쇄함으로써 측정가능한 주파수 대역을 넓혔으며 측정오차의 원인을 분석하였다. A method for measuring the loudspeaker distortions at low frequencies without an anechoic is proposed. This method is based on the fact that the n-th harmonic distortion outside the enclosure is boosted by 40log n dB compared to that inside the enclosure. The applicable frequency range is extended by cancelling the effect of standing wave inside the enclosure. Causes of measurement error are also analyzed.