• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.9
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• pp.1456-1467
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• 1990

The time domain adaptive noise canceller (time domain ANC) with the adaptive compensator and its algorithm, so called compensated least mean squares(CLMS) algorithm, had been introduced to improve the performance of ANC[1]. In this paper the time domain ANC with the adaptive compensator is transformed into the frequency domain ANC with the adaptive ocmpensator. An compensated frequency-domain least mean squares(CFLMS) algorithm that can adapt the proposed frequency domain ANC is presented.

• The Journal of the Acoustical Society of Korea
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• v.34 no.2
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• pp.163-170
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• 2015

In this paper, it is shown that the performance of the frequency-domain multi-reference narrowband noise canceller is determined by the narrowband component to the broadband disturbance power ratio in the reference signals. To overcome this problem, a new narrowband ANC is proposed, where the update of the adaptive filter is determined based on SNR of the reference inputs being measured using the magnitude squared coherence (MSC) between the primary and the reference signals. Simulation results show that the proposed ANC has superior performance over the conventional one.

• ETRI Journal
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• v.35 no.5
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• pp.859-868
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• 2013

The active noise control (ANC) technique attenuates acoustic noise in a flexible and effective way. Traditional ANC design aims to minimize the residual noise energy, which is indiscriminative in the frequency domain. However, human hearing perception exhibits selective sensitivity for different frequency ranges. In this paper, we aim to improve the noise attenuation performance in perceptual perspective by incorporating noise weighting into ANC design. We also introduce psychoacoustic analysis to evaluate the sound quality of the residual noise by using a predictive pleasantness model, which combines four psychoacoustic parameters: loudness, sharpness, roughness, and tonality. Simulations on synthetic random noise and realistic noise show that our method improves the sound quality and that ITU-R 468 noise weighting even performs better than A-weighting.

• Journal of the Institute of Convergence Signal Processing
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• v.1 no.1
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• pp.7-14
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• 2000

Recently, the technologies of adaptive noise cancellation(ANC) are developed fast and widely due to the highly sophisticated digital signal processing algorithms and the high-speed communication networks and devices. But, thousand numbers of the adaptive filter taps are required to obtain the satisfying results in the fields of the adaptive noise cancellation and echo cancellation. In the paper, performance comparisons based on the real-time processing between frequency domain adaptive filter(FDAF) and multi-delay frequency domain adaptive filter(MDF) are carried. Those algorithms provide us with the reductions of the computational burdens and the increase of the convergence rate for the lengthy Fill adaptive filters. The time delay due to the long taps of FDAF can be reduced by adopting the MDF algorithms. The conventional ANC and cross talks ANC using FDAF are implemented on the dSP ACE 1103 real-time signal processing board.

• Journal of KSNVE
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• v.8 no.5
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• pp.914-921
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• 1998

Combusion instability due to thermoacoustic feedback in a ducted combustor usually excites severe noise and vibration, which could lead to result in the failure of the system or environmental dispute. In the present study, an active noise control(ANC) method with an adaptive algotithm is hired to suppress instability which has very discrete behavior in the frequency domain. Especially a feedback system is composed to evade hot environment of the combustor, and as a preliminary, the performance and stability of the controller is chekced by simulating the real situation with harmonic waves. Application to the real combustor showed serious reductions in sound pressure level by 20∼30 dB. It was also shown that the selected control system was very stable and effective.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2E
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• pp.56-67
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• 2004

Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.559-567
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• 2020

This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm's limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.