• Transactions on Control, Automation and Systems Engineering
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• v.4 no.4
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• pp.305-310
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• 2002

The measured rate of the tracking sensor becomes biased under some operational situation. For a highly maneuverable aircraft in 3D space, the target dynamics changes from time to time, and the Kalman filter using position measurement only can not be used effectively to reject the rate measurement bias error. To cope with this problem, we present a new algorithm which incorporate FIR-type filter and FIR-type fixed-lag smoother, and demonstrate that it has the optimal performance in terms of both estimation accuracy and response time through an application example to the anti-aircraft gun fire control system(AAGFCS).

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.60 no.1
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• pp.19-26
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• 2011

In this paper, we designed Interpolation FIR(Finite Impulse Response) filter and 1-bit SDM(Sigma- Delta Modulator) for small digital audio speaker, which has low power consumption and high output characteristics. In order to achieve high linearity and low distortion performance of the systems, we adopt Type I Chevychev FIR filter which has equiripple characteristics in the pass band and proposed high efficient FIR filter structure. SDM is the most efficient modulation technique among the noise shaping techniques. In this paper, we implemented SDM using CIFB(Cascade of Intergrators, Feed-Back) which is generally used in DAC of small digital audio speakers. The proposed SDM structure can achieve high SNR, high-efficiency characteristics and low power consumption in mobile devices. Also considering manufacture of SoC(System on Chip), we performed simulation with Matlab and Verilog HDL to obtain optimal number of operational bits and verified a good experimental results.

• International journal of advanced smart convergence
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• v.4 no.2
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• pp.138-144
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• 2015

Digital filters are extensively used in the world of communication. In order to design a digital finite impulse response (FIR) filter that satisfies all the required conditions is challenging. In this paper, design techniques of digital low pass FIR filters using Rectangular window method, Hamming window, Hanning window, and Optimal Parks McClellan method are presented. The stability, number of components required and filter coefficients are demonstrated for different design techniques. It is demonstrated that filter design using hamming window is comparatively better than rectangular and hanning window though the components required for all of the windowing technique are same, hamming shows higher stability. The stability is shown with the help of magnitude and phase spectrum of each window. Simulation is carried out using MATLAB and comparisons are made entirely based on the output of the simulation.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.2
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• pp.249-256
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• 1986

In this paper, we propose the MDM algorithm by which one can desing an FIR filter that is maximally flat and requires no multiplication. We use the modified MAXFLAT subroutive of Kaiser to achieve the maximally-flat characteristics. The filter coefficients are encoded in MDM-code and the optimal stepsize is determined the steepest- descent method. Simulation results shows that the FIR filter designed is almost maximally-flat in passband, but has about -30dB ripples in stopband due to MDM quantization error.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.686-689
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• 2002

This paper proposes a weighted least-squares(WLS) method for designing variable one-dimensional (1-D) FIR digital filters with simultaneously variable magnitude and variable non-integer phase-delay responses. First, the coefficients of a variable FIR filter are represented as the two-dimensional (2-D) polynomials of a pair of spectral parameters: one is for tuning the magnitude response, and the other is for varying its non-integer phase-delay response. Then the optimal coefficients of the 2-D polynomials are found by minimizing the total weighted squared error of the variable frequency response. Finally, we show that the resulting variable FIR filter can be implemented in a parallel form, which is suitable for high-speed signal processing.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.1
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• pp.105-113
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• 2000

This paper proposes the optimal design method of PSF(pulse shaping filter) with fixed-point coefficients, often used in digital communication system. Though RCF (Raised Cosine Filter) and Root-Squared RCF have less attenuation in stopband and are designed with floating point coefficients, those are selected by the reason that the design is simple. In this paper, 1 introduce the optimal design method for fixed point PSF including Root Squared type by using mixed integer linear programming. Through some design examples, it is shown that the proposed method better performs in ISI and requires less complexity. The complexity of the proposed filter is reduced to 20% as compared to conventional RCF and Root Squared RCF. For IS-95, that is the standard of CDMA system, the proposed filter reduces ISI up to 75% compared to the standard transmission filter.

• ETRI Journal
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• v.33 no.4
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• pp.641-644
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• 2011

In this letter, we propose a novel bitrate control, using both Kalman and FIR filters, based on a Hamiltonian analysis with respect to the amount of bits from each macroblock, in an encoding of a general video codec such as H.264/AVC. Since the proposed bitrate control is based on the simple computation of an optimal control method based on the Hamiltonian analysis, it is not necessary to use additional computation, such as a DCT or quantization, to estimate the bits for bitrate control. As a result, the proposed algorithm can be applied to single-pass encoding and can provide sufficient encoding speed with respect to various applications, even those requiring real-time control.

• The Journal of the Acoustical Society of Korea
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• v.11 no.2
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• pp.59-67
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• 1992

Recently, the problem which actively control the unwanted noise propagated from the technical structure by the generated secondary sound has become considerable topic from the environmental preservation point of view. In most of these studies, active noise control deals with a plane wave propagation at low frequency using adaptive filtering techniques. On the other hand, in real acoustic systems are mostly short due to the limitation of geometric configuration. In this case, the acoustic properties such as reflections and resonances inside the acoustic system should be considered. In this paper, the acoustic modeling method for short length duct was introduced using the transfer matrix method, and the active noise control problem was investigated with \implementation of FIR filter for the transfer function of control system derived from this modeling method. The identification methods for the acoustic model of actual control system was proposed by numerical computation technique based on the estimation of optimal FIR filter coefficients. The acceptable attenuation on the real acoustic system and stability of the controller are predicted in this computational simulation.

• Journal of the Korea Institute of Military Science and Technology
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• v.3 no.2
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• pp.121-130
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• 2000

A practical method is proposed to improve hit probability of the digital gun fire control system, when the measured rate of the tracking sensor becomes biased under some operational situation. For ground moving target it is shown that the well-known Kalman filter which uses position measurement only can be optimally used to eliminate the rate bias error. On the other hand, for 3D moving aircraft we present a new algorithm which incorporate FIR-type filter, which uses position and rate measurement at the same time, and the fixed-lag smoother using position measurement only, and show that it has the optimal performance in terms of both estimation accuracy and response time.

• Journal of Information Processing Systems
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• v.15 no.2
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• pp.374-385
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• 2019

This paper presents an optimal implementation of a Daubechies-based pipelined discrete wavelet packet transform (DWPT) processor using finite impulse response (FIR) filter banks. The feed-forward pipelined (FFP) architecture is exploited for implementation of the DWPT on the field-programmable gate array (FPGA). The proposed DWPT is based on an efficient transpose form structure, thereby reducing its computational complexity by half of the system. Moreover, the efficiency of the design is further improved by using a canonical-signed digit-based binary expression (CSDBE) and advanced functional sharing (AFS) methods. In this work, the AFS technique is proposed to optimize the convolution of FIR filter banks for DWPT decomposition, which reduces the hardware resource utilization by not requiring any embedded digital signal processing (DSP) blocks. The proposed AFS and CSDBE-based DWPT system is embedded on the Virtex-7 FPGA board for testing. The proposed design is implemented as an intellectual property (IP) logic core that can easily be integrated into DSP systems for sub-band analysis. The achieved results conclude that the proposed method is very efficient in improving hardware resource utilization while maintaining accuracy of the result of DWPT.