• Proceedings of the KIEE Conference
• /
• 1994.11a
• /
• pp.332-334
• /
• 1994

This paper proposes a robust active noise control (ANC) system. The system constructs filter with a standard model. An adaptive litter, connected to the system in parallel, compensates the modeling error. When the adaptive filter does not work, the proposed system is robust while the conventional system loses the noise control capability. But proposed ANC system shows that the performance is comparable with that of the conventional ANC system under normal conditions.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
• /
• v.17 no.6
• /
• pp.147-153
• /
• 2003

In this paper, implementation of active noise barriers(ANB) to attenuate exterior noise which is propagated through open windows is presented. The leaky multiple channel LMS algorithms are used for adaptive filters tc improve the convergence property, and a new type of the active noise barrier is proposed. The attenuation effects of conventional active noise control(ANC) systems using leaky multiple channel LMS algorithms and the proposed system are compared by experiments using a TMS320C33 digital signal processor. Noise attenuation levels at the points of error microphones are similar for both systems, but average noise attenuation effects of the proposed system for an entire space of an experimental enclosure are much better than conventional ANC systems.

• Journal of KSNVE
• /
• v.4 no.2
• /
• pp.187-198
• /
• 1994

This study is to describe the effects of the duct termination conditions conditions upon the active noise attenuation system. The adaptive filtering algorithm using FIR filter is implemented for duct noise attenuation. To avoid the instability caused by the acoustic feedback, two methods are considered. One is to use a compensating FIR filter. The other is to utilize uni-directional detecting microphone and uni-directional control speaker. Experimental results show that the reflections of sound from duct terminations greatly reduce the performance of ANC system. The directionality of detecting microphone and control speaker is a major factor to decide ANC performance. When there are some reflections from both duct terminations, the noise attenuation using finite FIR filter is not enough to model the duct plant. Especially, the reflection from the upstream termination reduces the noise attenuation in the frequencies related to the distance between control speaker and upstream termination. The performance of the noise attenuation is found to be largely enhanced by using uni-directional coupler, both on detecting microphone and control speaker, even if the duct system has an arbitrary termination conditions.

• Proceedings of the KIEE Conference
• /
• 2001.07d
• /
• pp.1964-1966
• /
• 2001

Active Noise control(ANC) is rapidly becoming the most effective way to reduce noises that can otherwise be very difficult and expensive to control. This research presents ANFIS (Adaptive Network Fuzzy Inference System) controller for adaptively noise cancelling in a duct. ANC system generates secondary control sound pressure with same amplitude and with opposite phase as noise to be eliminated. ANFIS controller is trained to optimize its parameters for adaptively cancelling noise. That is ANFIS train its parameters by gradient descent and LSE method so called hybrid method. This paper present ANFIS in active noise control which provides an improvement convergence speed and limitation of linearity condition. It can model nonlinear functions of arbitrary complexity and ANFIS can construct an input-ouput mapping based on both human knowledge in the form of Takagi and Sugeno's fuzzy if-then rules and stipulated input-output data pairs. This paper also shows that the proposed ANFIS active noise control system successfully cancelled noise.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2000.06a
• /
• pp.269-273
• /
• 2000

This paper describes Active Noise Control (ANC) using Discrete Wavelet Transform (DWT) Domain Least Mean Square (LMS) Method. DWT-LMS is one of the transform domain input decorrelation LMS and improves the convergence speed of adaptive filter especially when the input signal is highly correlated. Conventional transform domain LMS's use Discrete Cosine Transform (DCT) because it offers linear band signal decomposition and fast transform algorithm. Wavelet transform can project the input signal into the several octave band subspace and offers more efficient sliding fast transform algorithm. In this paper, we propose Wavelet transform domain LMS algorithm and shows its performance is similar to DCT LMS in some cases using ANC simulation.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.9
• /
• pp.172-176
• /
• 2013

This paper presents a modified filtered-x least mean square (FxLMS) algorithm to improve the stability of active noise control (ANC) system in realistic environment. A real-time ANC system employing modified FxLMS is designed and implemented on digital signal processor (DSP) board. The ANC system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz and the performance is measured in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone with maximum 20 dB SPL attenuation can be generated around the location of error microphone.

• Proceedings of the KIEE Conference
• /
• 2004.05a
• /
• pp.22-25
• /
• 2004

In active noise control (ANC) system, the convergence behavior of the Filtered- X Least Mean Square (FXLMS) algorithm may be affected by nonlinear distortion in the secondary path as in the power amplifiers (e.g., saturation), loudspeakers and transducers. This distortion may yields degrading the error reduction performance of the ANC systems. In this paper, the authors of this paper propose a more improved and stable FXLMS algorithm to compensate for the undesirable nonlinearity of the secondary-path, whereby the third-order Volterra model was employed for the identification of the nonlinear secondary-path. In particular, the proposed approach was based on the modification of the conventional FXLMS algorithm. Finally, the simulation results showed that the proposed approach yields better convergence property and more stable performance in the ANC systems.

• Transactions of the Korean Society of Automotive Engineers
• /
• v.13 no.4
• /
• pp.152-159
• /
• 2005

The method of induction noise reduction can be classified by using passive control or active control method. However, the passive control method has a demerit to reduce the effect of noise reduction to low frequency (below) 500Hz) range and to be limited in a space of the engine room. Whereas, the active control method can overcome the demerit of passive control method. The algorithm of active control is mostly used in LMS (Least-Mean-Square) algorithm because it can obtain the complex transfer function easily in real-time. Especially, Filtered-X LMS (FXLMS) algorithm is applied to an ANC system. However, the convergence performance of LMS algorithm could not match if the FXLMS algorithm is applied to an active control of the induction noise under rapidly accelerated driving conditions. So, in order to solve the problem in this study, the Moving Bandpass Filter(MBPF) was proposed and implemented. The ANC using MBPF for the reduction of the induction noise shows that more noise reduction as 4dB than without MBPF.

• The Journal of the Acoustical Society of Korea
• /
• v.34 no.5
• /
• pp.377-384
• /
• 2015

An active noise control (ANC) algorithm for noise canceling headphone is proposed. In this study, the feedback ANC operated with the filtered-x least mean square algorithm (FxLMS) algorithm is used to attenuate the undesired noise. Also an adaptive residual music canceller (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Simulation results show that a high quality of music sound can be consistently achieved in a time-varying secondary path situation.

• Proceedings of the KIEE Conference
• /
• 1992.07a
• /
• pp.323-326
• /
• 1992

In this paper, the active noise control(ANC) system using the inverse modeling techiniques is presented. The nonlinearity and time delay of the transfer function from the secondary speaker to the error microphone makes the ANC system have poor performance. To solve this problem, the inverse model technique and filtered-X LMS algorithm is using proposed. This proposed ANC system is implemented using DSP chip and operated in on-line. The experimental results show that this ANC system has better noise canceling performance than that used LMS only about 5-15[db]