• 한국광학회지
• /
• 제10권5호
• /
• pp.413-418
• /
• 1999

광섬유 회로를 이용한 마이크로파의 신호처리 네트워크의 주파수 특성을 연구하였다. 단일 모드 광섬유 지연선로와 Mach-Zehnder 및 Fabry-Perot 간섭계를 이용한 3종류의 마이크로파 주파수 계수를 제거 필터를 구성하여 이론적 모델과 측정결과를 비교 하였다. 마이크로파 필터의 주파수 전달함수 계산은 해석이 간단한 Mason의 그래프 분석 방법을 이용하였으며 실험결과와 잘 일치하였다. 깊이 36㏈의 우수한 특성을 갖는 광섬유 마이크로파 주파수 제거 필터를 구현하였다.

• Structural Engineering and Mechanics
• /
• 제49권3호
• /
• pp.411-425
• /
• 2014

This paper presents the processing of nonlinear features associated with a damage event by quadratic time-frequency distributions for damage identification in a frame structure. A time-frequency distribution is a function which distributes the total energy of a signal at a particular time and frequency point. As the occurrence of damage often gives rise to non-stationary, nonlinear structural behavior, simultaneous representation of the dynamic response in the time-frequency plane offers valuable insight for damage detection. The applicability of the bilinear time-frequency distributions of the Cohen class is examined for the damage assessment of a frame structure from the simulated acceleration data. It is shown that the changes in instantaneous energy of the dynamic response could be a good damage indicator. Presence and location of damage can be identified using Choi-Williams distribution when damping is ignored. However, in the presence of damping the Page distribution is more effective and offers better readability for structural damage detection.

• The Journal of the Acoustical Society of Korea
• /
• 제20권4E호
• /
• pp.24-30
• /
• 2001

This paper describes a fast harmonic estimation, referred to as Delta Adjustment (DA), using a low resolution pitch. The presented DA method is based on modification of the Generalized Dual Excitation (GDE) technique[1] which was proposed to improve speech enhancement performance. We introduce the GDE technique and modify it to be suitable for low bit rate harmonic coding that uses only an integer pitch estimate. Unlike the GDE, the DA matches a frequency-warped version of the original spectrum that conforms to a fixed pitch at all harmonic bands. In addition, complexity and performance of the presented method are described in comparison with those of the conventional Fractional Pitch (FP) based harmonic estimation. Experimental results showed that the DA algorithm significantly reduces the complexity of the FP method while maintaining the performance.

• 전자공학회논문지
• /
• 제50권1호
• /
• pp.97-101
• /
• 2013

There is a growing need for optical communication systems that convert large volumes of data to optical signals and that accommodate and transmit the signals across long distances. Digital optical communication consists of a master unit (MU) and a slave unit (SU). The MU transmits data to SU using digital optical signals. However, digital optical units that are commercially available or are under development transmit data using two's complement representation. At low input levels, a large number of SSOs (simultaneous switching outputs) are required because of the high rate of bit switching in two's complement, which thereby increases the power noise. This problem reduces the overall system capability because a DSP (digital signal processor) chip (FPGA, CPLD, etc.) cannot be used efficiently and power noise increases. This paper proposes a change from two's complement to a more efficient method that produces less SSO noise and can be applied to existing digital optical units.

• 제어로봇시스템학회:학술대회논문집
• /
• 제어로봇시스템학회 2002년도 ICCAS
• /
• pp.96.2-96
• /
• 2002

In this paper, a new method for the mixed image separation is presented using the independent component analysis, the innovation process, and the expectation-maximization. In general, the independent component analysis (ICA) is one of the widely used statistical signal processing scheme that represents the information from observations as a set of random variables in the form of linear combinations of another statistically independent component variables. In various useful applications, ICA provides a more meaningful representation of the data than the principal component analysis through the transformation of the data to be quasi-orthogonal to each other, which can be utilized in linear p...

• Journal of Communications and Networks
• /
• 제9권1호
• /
• pp.67-74
• /
• 2007

The performance of dual diversity receivers operating over correlated Ricean fading channels is analyzed. Using a previously derived rapidly converging infinite series representation for the bivariate Ricean probability density function, analytical expressions for the statistics of dual-branch selection combining, maximal-ratio combining, and equal-gain combining output signal-to-noise ratio (SNR) are derived. These expressions are employed to obtain novel analytical formulae for the average output SNR, amount of fading, average bit error probability, and outage probability. The proposed mathematical analysis is used to study various novel performance evaluation results with parameters of interest the fading severity, average input SNRs, and the correlation coefficient. The series convergence rate is also examined verifying the fast convergence of the analytical expressions. The accuracy of most of the theoretical performance evaluation results are validated by means of computer simulations.

• 제어로봇시스템학회:학술대회논문집
• /
• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
• /
• pp.469-469
• /
• 2000

The use of orthogonal functions with the aim of adapting the system and signal representation to the specific properties of the systems and signals has a long history. A least-squares identification method is studied that estimates a finite number of expansion coefficients in the series expansion of a transfer function, where the expansion is in terms of recently introduced generalized orthogonal functions. It is shown that there exist orthogonal functions that are generated by stable linear dynamical systems.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 1988년도 전기.전자공학 학술대회 논문집
• /
• pp.624-627
• /
• 1988

This paper describes the implementation of recognition of speaker - dependent Korean spoken continuous digits. The recognition system can be divided into two parts, acoustic - phonetic processor and lexical decoder. Acoustic - phonetic processor calculates the feature vectors from input speech signal and the performs frame labelling and phone labelling. Frame labelling is performed by Bayesian classification method and phone labelling is performed using labelled frame and posteriori probability. The lexical decoder accepts segments (phones) from acoustic - phonetic processor and decodes its lexical structure through phone network which is constructed from phonetic representation of ten digits. The experiment carried out with two sets of 4continuous digits, each set is composed of 35 patterns. An evaluation of the system yielded a pattern accuracy of about 80 percent resulting from a word accuracy of about 95 percent.

• 대한전자공학회논문지
• /
• 제27권4호
• /
• pp.636-644
• /
• 1990

Hidden Markov models(HMM's) have been known to be useful representation for speech signal and are used in a wide variety of speech systems. For speech recognition applications, it is desirable to incorporate durational information of states in model which correspond to phonetic duration of speech segments. In this paper we propose duration-dependent HMM's that include durational information of states appropriately for the left-to-right model. Reestimation formulae for the parameters of the proposed model are derived and their convergence is verified. Finally, the performance of the proposed models is verified by applying to an isolated word, speaker independent speech recognition system.

• Smart Structures and Systems
• /
• 제9권5호
• /
• pp.441-459
• /
• 2012

Identification of vibration parameters from the analysis of the dynamic response of a structure plays a key role in current health monitoring systems. This study evaluates the capabilities of the recently developed Synchrosqueezed Wavelet Transform (SWT) to extract instant frequencies and damping values from the simulated noise-contaminated response of a structure. Two approaches to estimate the modal damping ratio from the results of the SWT are presented. The results obtained are compared to other signal processing methods based on Continuous Wavelet (CWT) and Hilbert-Huang (HHT) transforms. It was found that the time-frequency representation obtained via SWT is sharped than the obtained using just the CWT and it allows a more robust extraction of the individual modal responses than using the HHT. However, the identification of damping ratios is more stable when the CWT coefficients are employed.