• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2035-2038
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• 2002

The necessity of the filter coefficients compensation for the LTJ adaptive filter was explained generally and easily by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.438-445
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• 2002

In this paper, the necessity of the filter coefficients compensation for the lattice transversal joint (LTJ) adaptive filter was explained in general and with ease by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed using the property that speech signal is stationary during a short time period and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%.

• Speech Sciences
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• v.11 no.2
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• pp.227-235
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• 2004

In this paper, we proposed a new practical implementation method of the lattice transversal joint (LTJ) adaptive filter using speech codec's information. And it was applied to the adaptive echo cancellation problem to verify the efficiency of the proposed method. Realtime implementation of the LTJ adaptive filter is very difficult due to high computational complexity for the filter coefficients compensation. However, in case of using speech codec, complexity can be reduced since linear predictive coding (LPC) coefficients are updated each frame or sub-frame instead of every sample. Furthermore, LPC coefficients can be acquired from speech decoder and transformed to the reflection coefficients. Therefore, the computational complexity for updates of the reflection coefficients can be reduced. The effectiveness of the proposed LTJ adaptive filter was verified by the experiments about convergence and tracking performance of the adaptive echo canceller.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.1
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• pp.67-79
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• 1996

This paper presents a composite luminance signal processing system for NTSC, PAL and SECAM standards. Eaxh of the three standards employs its own specifications of subcarmier bandwidth and luminance signal waveform. The proposed system, compatible to the specifications of the three standard and B/W TV, implements variable freqneucy characteristics by controlling filter coefficients. The major features of the system are a luminance/chroma separation unit and an aperture compensation unit. The luminance/chroma separation unit employes a notch filter selection a trap freqneyc to atenuate unwanted color signals in luminance signal bands. The aperture compensation unit comprises two subunits, to provide clear color definition for each of the three standards: a primary compensation circuit and a variable compensation circuits. The proposed system yields a 40 dB gain from the chroma/luminance separation and a 10 dB gain from the aperture compensation unit.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.1
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• pp.87-96
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• 2002

This paper proposes an efficient decimation filter structure in programmable down converter for software radio. The decimation filter consists of the cascaded integrator-comb(CIC) filter, a compensation filter for CIC, cascaded comb and modified halfband filters, and programmable FIR filter. Since the compensation filter is used in CIC, the passband drooping is compensated and stopband attenuation is improved. Therefor the more decimation can be implemented in CIC filter. The compensation filter in CIC reduced the computational complexity of other decimation filters and the coefficients of PFIR, thereby achieving a significant hardware reduction over existing approaches. We can reduce the multiply operator by 20% in hardware and operation by 50% as compared with PDC of Harris.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.63-71
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• 2000

A new type of second-order digital peaking filters for professional-quality digital audio system is proposed whose frequency response can be elaborately controlled throughout the composite structure of a standard band-pass filter and a 0-dB bypass gain. The proposed method for designing the peaking filter uses the Q-compensation technique to prevent the Q-distortion caused by the variation of the gain factor and is reduced into a compact form which is proper to the real-time implementation. Methods are examined for computing its coefficients, which are exact and very straightforward to compute with small amount of the system resources.

• Journal of Navigation and Port Research
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• v.40 no.5
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• pp.241-247
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• 2016

The tracking filter plays a key role in the accurate estimation and prediction of maneuvering a vessel's position and velocity when attempting to enhance safety by avoiding collision. Therefore, in order to achieve accurate estimation and prediction, many oceangoing vessels are equipped with the Automatic Radar Plotting Aid (ARPA) system. However, the accuracy of prediction depends on the tracking filter's ability to reduce noise and maintain a stable transient response. The purpose of this paper is to derive the optimal values of the gain parameters used in tracking a High Dynamic Warship. The algorithm employs a ${\alpha}-{\beta}-{\gamma}$ filter to provide accurate estimates and updates of the state variables, that is, positions, velocity and acceleration of the high dynamic warship based on previously observed values. In this study, the filtering coefficients ${\alpha}$, ${\beta}$ and ${\gamma}$ are determined from set values of the damping parameter, ${\xi}$. Optimization of the damping parameter, ${\xi}$, is achieved experimentally by plotting the residual error against different values of the damping parameter to determine the least value of the damping parameter that results in the optimum smoothing coefficients leading to a reduction in the noise corruption effect. Further investigation of the performance of the filter indicates that optimal smoothing coefficients depend on the initial and average velocity of the target.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.7
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• pp.637-648
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• 2007

This paper presents a sensor fusion method based on indirect Kalman filter(IKF) for error compensation of low-cost inertial sensors and its application to the determination of attitude and position of small flying robots. First, the analysis of the measurement error characteristics to zero input is performed, focusing on the bias due to the temperature variation, to derive a simple nonlinear bias model of low-cost inertial sensors. Moreover, from the experimental results that the coefficients of this bias model possess non-deterministic (stochastic) uncertainties, the bias of low-cost inertial sensors is characterized as consisting of both deterministic and stochastic bias terms. Then, IKF is derived to improve long term stability dominated by the stochastic bias error, fusing low-cost inertial sensor measurements compensated by the deterministic bias model with non-inertial sensor measurement. In addition, in case of using intermittent non-inertial sensor measurements due to the unreliable data link, the upper and lower bounds of the state estimation error covariance matrix of discrete-time IKF are analyzed by solving stochastic algebraic Riccati equation and it is shown that they are dependant on the throughput of the data link and sampling period. To evaluate the performance of proposed method, experimental results of IKF for the attitude determination of a small flying robot are presented in comparison with that of extended Kaman filter which compensates only deterministic bias error model.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.26-27
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• 2014

This paper presents an adaptive interpolation filtering scheme for the High Efficiency Video Coding (HEVC) standard. In regards to interpolation for motion estimation and compensation, the conventional HEVC employs 8-tap and 4-tap filters for luma and chroma samples, respectively. Coefficients in such filters are determined by discrete cosine transform (DCT). In the proposed scheme, boundary strength values are stored after the execution of the deblocking filter. For each block, the sum of boundary strength values is calculated to indicate whether its region is complex or simple. Consequently, based on the region classification, 12-tap and 8-tap interpolation filters are used for complex and simple regions, respectively. This process is applied to luma sample interpolation only. Simulation results show 1.8% average BD-rate reduction compared to the conventional method.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.7
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• pp.1003-1009
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• 1995

In This Paper, Linearization problem is discussed to reduce distortion of a nonlinear system based on Schetzen's pth-orfer inverse theorem. We propose a nonlinear adaptive prefiltering algorithm which can reduse nonlinear distortion up to pth order by tandemly connecting a pth-order Volterra filter before the nonlinear system under the consideration and by adjusting the filter coefficients adaptively. The feasibility of applying the proposed algorithm to a nonlinear system is conformed via computer simulation by observing significant reduction of total nonlinear distortion for the case of random input and sinusoidal input excitation.