• The Journal of the Acoustical Society of Korea
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• v.24 no.3E
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• pp.99-108
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• 2005

To investigate the equalizer performance in underwater acoustic communication m the presence of intersymbol interference (ISI) due to multipath, computer simulations are carried out in discrete multipath shallow water channels for three different horizontal ranges. For the purpose of computation simplicity, least mean square (LMS) algorithm is adopted both in linear equalizer and nonlinear equalizer, decision feedback equalizer (DFE) to cancel out ISI effects. Binary phase shift keying (BPSK) signals have been transmitted with high data rate of 2000bps through the use of equalization technique. The results demonstrate that equalization is an efficient way to achieve high transmission data rate in the shallow water channel.

• 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 Korean Institute of Communications and Information Sciences
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• v.35 no.1C
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• pp.96-102
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• 2010

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

• Journal of Electrical Engineering and Technology
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• v.12 no.4
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• pp.1627-1633
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• 2017

This paper proposes a voice enhancing ear-protection system which is based on feedback active noise control(FBANC). The proposed system selectively suppresses the background noise and preserves the talking voice by controlling the adaptive algorithm with the voice activity period detection module. The noise reduction performance of the proposed noise canceling algorithm is analytically derived for the two key performance affecting parameters, i.e., electro-acoustic coupling distance and noise bandwidth. The proposed system is also implemented with a floating-point DSP system and its performance is experimentally tested to compare with the analytically derived results. The achieved levels of noise reduction for the three different noise bandwidths cases, i.e., 10Hz, 50Hz, and 90Hz, are high to show 17.05dB, 10.54dB and 8.99dB, respectively. The feasibility of the proposed system is also shown by the peak noise reduction achieved more than 25dB while preserving the voice component in the frequency range between 200-800Hz.

• Journal of KSNVE
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• v.9 no.1
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• pp.33-41
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• 1999

This paper presents a method of actively controlling the interior noise by a trim panel with hybrid feedforward-feedback control loop. The control technique is designed to minimize the vibration of panel whose motion is limited to that of a piston (out-of-plane motion). The hybrid controller consists of an adaptive feedforward controller in conjunction with a linear quadratic Gaussian (LQG) feedback controller. In order to maintain control performance of both persistent and transient disturbances, the feedback loop speeds up the adaptation rate of feedforward controller by improving damping capacity of secondary plant related with the adaptation rule. Numerical simulation and experimental result indicate that the hybrid controller is a more effective method for reducing the vibration of the panel (and therefore the interior noise) compared to using feedforward controller.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.699-702
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• 2009

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows the improved SNR of about more than 20 dB.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.64 no.3
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• pp.169-174
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• 2015

Noise problem that occurs on the road is raising a lot of problems in the economic, social and environmental aspects. The active noise control (ANC) systems based on the filtered-X least mean square(FxLMS) algorithm have a problem with compensating the acoustic feedback of secondary route. However, newly proposed correlation-LMS(CLMS) and expanded CLMS algorithms have slightly much calculation and are minutely behind performance, these have a advantage not in measuring transfer function onerously so that we can easily adapt these in real time. The CLMS and expanded CLMS algorithm was developed to improve the real-time implementation performance under the variable input noise such as road noise environment. In this paper, we compared and analyzed their performance. From the results of the Matlab simulation for an ANC system, it is shown that expanded CLMS algorithms are more convergence speed and keep the desirable performance even in the input of road noise situation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7C
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• pp.413-420
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• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.4
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• pp.645-651
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• 2009

In this paper, we present bit error rate(BER) performance of an adaptive decision feedback equalizer(DFE) using experimental data. The experiment was performed at the shore of Geoje in November 2007. The BER of the adaptive DFE whose tap weight is updated by RLS is described with change of feedforward filter length, feedback filter length, training sequence length, and delay, which shows that the uncoded average BER is $4{\times}10^2\;and\;1.5{\times}10^{-2}$ with transmission range of 9.7km and 4km, respectively. The BER of the adaptive DFE can be lower than 10-3 by a forward error correction code and therefore the adaptive DFE may be a good candidate for a high speed AUV communications since the volume and weight of the underwater acoustic modem should be small because of the restricted space and power in the battery-operated AUV.

• Transactions of the Korean Society of Mechanical Engineers B
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• v.20 no.3
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• pp.1106-1113
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• 1996

The phase criterion of the feedback cycle of low-speed edgetones has been obtained using the jet-edge interaction model which is based on the substitution of an array of dipoles for the reaction of the wedge to the impinging jet. The edgetone is produced by the feedback loop between the downstream-convected sinuous disturbance and upstream-propagating waves generated by the impingement of the disturbance on the wedge. By estimation of the phase difference between the downstream and the upstream disturbances, the relationship between the edge distance and the wavelength is obtained according to the phase-locking condition at the nozzle lip. With a little variation depending on the characteristics of jet-edge interaction, the criterion can be approximated as follows: h/.LAMBDA. + h/.lambda. = n - 1/4, where h is the stand-off distance between the nozzle lip and the edge tip, .LAMBDA. is the wavelength of downstream-convected wave, .lambda. is the wavelength of the upstream-propagating acoustic wave and n is the stage number for the ladder-like characteristics of frequency. The present criterion has been confirmed by estimating wavelengths from available experimental data and investigating their appropriateness. The above criterion has been found to be effective up to 90.deg. of wedge angle corresponding to the cavitytones.