• Proceedings of the KIEE Conference
• /
• 2011.07a
• /
• pp.1788-1789
• /
• 2011

In order to develop a one-axis gimbaled acoustic source tracker for mobile robots, a pseudo-linear direction of arrival(DOA) estimator is proposed using a linear ultrasonic sensor array. Under the assumption that the sensor measurement errors are negligible, a linear measurement model is derived using the linear prediction relation of the received sinusoidal acoustic signals. Applying the Kalman filtering technique for this model, the linear recursive DOA estimator is designed. For its linear recursive filter structure, it is preferable for real-time implementation on a commercial DSP. Through the experiments, the effectiveness of the suggested method is demonstrated.

• Journal of the Korean Society for Precision Engineering
• /
• v.18 no.8
• /
• pp.79-86
• /
• 2001

A method for actively minimizing dynamic reaction forces in a flexible structure subject to persistent excitations is presented. One difficulty with the method, however, is that forces and moments do not converge as quickly as displacements in mathematical discretization of continuous systems, so a controller based on a truncated model of a continuous system can produce poor results. A technique using residual flexibility matrix is presented for correcting the truncated force representation. A controller designed for reaction force minimization, using the residual flexibility matrix, is applied to a model of a flexible structure, and the results are presented. Implications of various reaction force penalty combinations on the resulting control performance are also discussed.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06c
• /
• pp.245-248
• /
• 1998

In this paper, a new variable wavelet packet decomposition audio coder, based on the time varying characteristic of the audio signals, is proposed and presents a technique to incorporate psychoacoustic models into an adaptive wave let packet scheme. The proposed filterbank improves the defect of the polyphase filterbank that could not properly represent the critical band and the defect of QMF-tree filter that need high complexity to implement. The filterbank consists of varying number of subband from 4 to 26 bands and use Daubechies 6-order wave let. The codec yields excellent quality at total bit rates of about 128kbps for monophonic CD-quality signals with an sampling frequency of 44.1kHz and reduces complexity by 19% for various bit-rates and sources with encoding and decoding process.

• Journal of the Korean Mathematical Society
• /
• v.54 no.2
• /
• pp.613-626
• /
• 2017

For the overdetermined linear system, when both the data matrix and the observed data are contaminated by noise, Total Least Squares method is an appropriate approach. Since an ill-conditioned data matrix with noise causes a large perturbation in the solution, some kind of regularization technique is required to filter out such noise. In this paper, we consider a Dual regularized Total Least Squares problem. Unlike the Tikhonov regularization which constrains the size of the solution, a Dual regularized Total Least Squares problem considers two constraints; one constrains the size of the error in the data matrix, the other constrains the size of the error in the observed data. Our method derives two nonlinear equations to construct the iterative method. However, since the Jacobian matrix of two nonlinear equations is not guaranteed to be nonsingular, we adopt a trust-region based iteration method to obtain the solution.

• Proceedings of the KIEE Conference
• /
• 2005.05a
• /
• pp.119-121
• /
• 2005

The sliding-mode control technique could make a system unstable which external disturbance and uncertainty exists in. This paper suggests a robust sliding-mode control algorithm which can be applied to a linear system with parameter uncertainties. To reduce the chattering effect, the whole system is comprised of using a state variable in which the state's estimated value is added. The condition of estimated state results from state observer. The proposed control algorithm uses the optimal feedback controller following the dynamic system equation which consists of a state variable resulting from its own state variable, controller input, estimated state variable. Through comparison with the time optimal control algorithm using simulation, the suggested algorithm shows the improved stability and robustness while it manifests the fast tracking characteristics.

• 제어로봇시스템학회:학술대회논문집
• /
• 2005.06a
• /
• pp.2362-2367
• /
• 2005

This paper reports a technique for discriminating double talk and echo path change using the stochastic characteristics of power change for an adaptive noise canceller. The causes of rapid error increasing are double talk and echo path change. When the echo path is changed, the system corrects the impulse response in order to reduce the error. However, in the case of double talk, the system has to suspend the updating impulse response in order to maintain the quality of the voice signal. In the conventional system, it was difficult to discriminate between the two situations. In this research, the stochastic characteristics of the voice power change in the double talk period were experimentally verified to be different from the power change during echo path changing. Based on the results, a new double talk detection method is proposed.

• Journal of the Korean Institute of Telematics and Electronics B
• /
• v.32B no.1
• /
• pp.96-102
• /
• 1995

This paper presents the VHSIC Hardware Description Language(VHDL) implementation of the Fixed Point Covariance Lattice(FLAT) algorithm for an Linear Predictive Coding(LPC) analysis and its related algorithms, such as the forth order high pass Infinite Impulse Response(IIR) filter, covariance matrix calculation, and Spectral Smoothing Technique(SST) in the Vector Sum Exited Linear Predictive(VSELP) speech coder that has been Selected as the standard speech coder for the North America and Japanese digital cellular. Existing Digital Signal Processor(DSP) chips used in digital cellular phones are derived from general purpose DSP chips, and thus, these DSP chips may not be optimal and effective architectures are to be designed for the above mentioned algorithms. Then we implemented the VHDL code based on the C code, Finally, we verified that VHDL results are the same as C code results for real speech data. The implemented VHDL code can be used for performing logic synthesis and for designing an LPC Application Specific Integrated Circuit(ASOC) chip and DsP chips. We first developed the C language code to investigate the correctness of algorithms and to compare C code results with VHDL code results block by block.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.4E
• /
• pp.53-57
• /
• 1996

An extension of the well-known Fourier transform is developed in this paper. It is denoted as the generalized Fourier transform(GFT), since it encompasses the Fourier transform as its special case. The first idea of this extension can be found on [1]. In the definition of the N-point discrete GFT, it first construct a passband in time which functions as a window in the time domain. An appropriate interpretation of each variables are introduced during the definition of the GFT, followed by the formal derivation of the inverse GFT. This transform pair is similar to the windowing in the frequency domain such as the subband coding technique (or filter bank approach) and could be extended to the wavelet transform.

• 한국전산유체공학회:학술대회논문집
• /
• 2006.10a
• /
• pp.47-50
• /
• 2006

In the aeroacoustic application of computational fluid dynamics, the physical phenomena like the crackle in the unsteady compressible jets should be based on very time-accurate numerical solution. The accuracy of the present numerical scheme is extended to the fifth order, using the WENO filter to the sixth-order central difference computation. However, the computational capacity is very restricted by the environment of computational power, so therefore the quadrilateral adaptive grids technique is introduced for this high-order accuracy scheme. The first problem is the multi-dimensional interpolation between fine and coarse grids. Some general benchmark problems are solved to show the effectiveness of this method.

• Transactions of the Korean Society of Mechanical Engineers
• /
• v.19 no.8
• /
• pp.2030-2038
• /
• 1995

In this study, three turbulent flame propagation models are compared using experimentally measured data of a 4 valves/cylinder spark-ignition engine. First two conventional models are B.K model and GESIM combustion model. The burning rates calculated from the two models are compared with the burning rates calculated from measured pressure data using the one-zone heat release analysis. GESIM combustion model predicts burning rates closer to the data acquired from the experiment in wide operating ranges than B-K model does. The third model is refined based on GESIM combustion model by including the effect of flame stretch, turbulent length scale band pass filter and a variable that considers flame size and the area of flame contacting the cylinder wall surface. The refined combustion model predicts burning rates closer to experimental results than GESIM combustion model does. Also, the refined combustion model predicts flame radius close to the experimental result measured by using optical fiber technique.