• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.30-32
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be Improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.331-337
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• 2018

In this paper, we apply the NR (Noise Rating) to the evaluation of the residual noise of the window type ANC (Active Noise Control) and study the active noise controller for minimizing the NR value in rooms. We proposed a shape of a noise shaping filter of the Filtered-E LMS (Least Mean Square) algorithm that reduces the NR value within the effective operating frequency band of the ANC. The usefulness of the proposed scheme is verified by simulation and showed that the NR value of the residual noise is lower than the filters used in the conventional psychoacoustic ANC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1533-1545
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• 1996

In this paper, the design and implementation of echo canceller with new variable step size algorithm is discussed. The method used in the new algorithm is to periodically adopt the test function which helps an optimal coefficient tracking. This algorithm outperforms LMS and VS algorithms in convergence speed and steady state error. As the period of test function is decreased, the speed of convergence is improved, but the number of calculation is increased, then the trade off between these parameters must be considered. Simulation results show new algorithm outperforms LMS and VS algorithms in convergence rate. For the design of hardware, circuit is designed with VHDL, and synthesized with Act1 withc is a FPGA library of ActelTM in use of synovation of InterGraph$^{TM}$. Verification of the synthesized circuit is carried out with simulator DLAB. The circuit based on the algorithm which is suggested in this paper calculated 7 radix places of inary number. A simulation data for the verification is based on the data of algorithm simulation. When the same input data is applied to the both simulation, output results of circuit simulation had slight difference in compare with that of algorithm simulation. The number of used gate is about 5,500 and We have 5.53MHz in maximum frequency.y.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8C
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• pp.697-708
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• 2010

In this paper, we compare the timing synchronization performance of the adaptive filter to that of the correlator in direct sequence spread spectrum (DS/SS) systems. The test variables used in the code synchronization are statistically analyzed for both schemes, and then the obtained results are used in calculating the detection and false alarm probabilities. Based on the derived probabilities, the synchronization performance is compared and the simulation is followed. Analysis and simulation results show that the correlator outperforms the adaptive filter under most synchronization environments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.2
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• pp.237-245
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• 2012

This paper describes an acoustic echo canceller with double-talk by the blind signal separation. The acoustic echo canceller is deteriorated or diverged in the double-talk period. So we use the blind signal separation to estimate the near-end speech signal and to eliminate the estimated signal from the residual signal. The blind signal separation extracts the near-end signal with dual microphones by the iterative computations using the 2nd order statistical character. Because the mixture model of blind signal separation is multi-channel in the closed reverberation environment, we used the copied coefficients of echo canceller without computing the separation coefficients. By this method, the acoustic echo canceller operates irrespective of double-talking. We verified performances of the proposed acoustic echo canceller by simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods thoroughly, and then operates stably in the normal state without the divergence of coefficients after ending the double-talking. And it shows the ERLE of averagely 20dB higher than the normal LMS algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.27-31
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• 2012

In this paper, we constructed an array of speaker system with directivity by using FXLMS(Filtered-X LMS) algorithm and confirmed its directivity. The front $0^{\circ}$ characteristics of the controlled speaker was suppressed by interfering it with the control signal produced with filter coefficients optimized with respect to the $180^{\circ}$ characteristics of the rear speakers. The directivity of the array of rear speakers was measured and the damping effect of the signal from the front speaker array was confirmed. The frequency characteristics and directivity was investigated by using the adaptive filter coefficients on damping, the damping on the control point was verified in all the frequency range. In 100Hz, 200Hz, 1000Hz regime, the damping effect was observed in the range of front $60^{\circ}{\sim}100^{\circ}$.

• Journal of Korea Society of Industrial Information Systems
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• v.27 no.2
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• pp.11-24
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• 2022

For the questions asked for LMS-based online evaluation the professor directly set exam questions, or use the automatic question-taking method according to the level of difficulty using the question bank divided by category. Among them, it is important to manage the difficulty of questions in an objective and efficient way, above all, in the automatic question-taking method according to difficulty. Because the questions presented to the evaluators may be different. In this paper, we propose an difficulty re-adjustment algorithm that considers not only the correct rate of a problem but also the time taken to solve the problem. For this, a logistic regression classification algorithm was used of machine learning, and a reference threshold was set based on the predicted probability value of the learning model and used to readjust the difficulty of each item. As a result, it was confirmed that there were many changes in the difficulty of each item that depended only on the existing correct rate. Also, as a result of performing group evaluation using the adjustment difficulty problem, it was confirmed that the average score improved in most groups compared to the difficulty problem based on the percentage of correct answers.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.38-46
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• 1999

In this paper, two methods were proposed to improve the performance of adaptive feedback cancellation in hearing aids. One is “Orthogonality principle acoustic feedback cancellation method(Orthogonality principle method)” to track optimal solution with monitoring the instantaneous gradient, the other is a method using the CLMS algorithm(CLMS method). In many simulation conditions, adaptive feedback cancellation method proposed in this paper was much better than Greenberg method by Sum-method LMS algorithm which is known the most excellent method by now in case of system mismatch, SNR and segmental SMR. Also. Orthogonality principle method is as good as CLMS method in terms of adaptive feedback cancellation in many simulation conditions.

• The Journal of the Acoustical Society of Korea
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• v.37 no.4
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• pp.206-214
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• 2018

This paper proposes an impact point estimation algorithm of the rifle using microphone sensors. The proposed algorithm resolves the time synchronization problem by expanding the existing ToA (Time of Arrival) method to TDoA (Time Difference of Arrival) method and verifies the performance of the algorithm through the actual shooting experiments. By comparing analysis of the actual and the estimated impact points by the algorithm, it is confirmed that the proposed algorithm has excellent performance by estimating the impact point accurately within the tolerance range.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.84-90
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• 2000

The bearing detection of radiated target noise is very important at underwater acoustic measurement and passive detection. It differs the arrival tines of received signal at each sensor. Therefore, the bearing can be obtained from the time delay. This paper proposes a new algorithm using the RLSL adaptive filter for TDE. The proposed method is particularly attractive when there is a limitation of priori information about the received signal spectra and when the delay is subject to variation. As the simulation results, it is shown that the proposed algorithm has better convergence characteristics and TDE speed, and so that the usefulness of proposed algorithm is confirmed.