Xiaojun Qiu

An Active Impulsive Noise Control Algorithm With Logarithmic Transformation

To overcome the limitations of the existing algorithms for active impulsive noise control, an algorithm based on minimizing the squared logarithmic transformation of the error signal is proposed in this correspondence. The proposed algorithm is more robust for impulsive noise control and does not need the parameter selection and thresholds estimation according to the noise characteristics. These are verified by theoretical analysis and numerical simulations.

Active noise attenuation in ventilation windows

The feasibility of applying active noise control techniques to attenuate low frequency noise transmission through a natural ventilation window into a room is investigated analytically and experimentally. The window system is constructed by staggering the opening sashes of a spaced double glazing window to allow ventilation and natural light. An analytical model based on the modal expansion method is developed to calculate the low frequency sound field inside the window and the room and to be used in the active noise control simulations. The effectiveness of the proposed analytical model is validated by using the finite element method. The performance of the active control system for a window with different source and receiver configurations are compared, and it is found that the numerical and experimental results are in good agreement and the best result is achieved when the secondary sources are placed in the center at the bottom of the staggered window. The extra attenuation at the observation points in the optimized window system is almost equivalent to the noise reduction at the error sensor and the frequency range of effective control is up to 390 Hz in the case of a single channel active noise control system.

A study of time-domain FXLMS algorithms with control output constraint

A single input, single output active noise control system using the time-domain Filtered-X LMS algorithm with output constraint is investigated. The constraint on the output of the control filter is applied by three different methods: the leakage algorithm based on the transformation method using a penalty function; the re-scaling algorithm based on the active set method; and the simple practical (clipping) algorithm which just clips the output if a constraint is encountered. A comparison of the three algorithms shows that the re-scaling algorithm can usually work successfully under the constraint, while the leakage algorithm usually needs a large leakage coefficient to satisfy the constraint with a resulting performance loss. The clipping algorithm has potential problems both with the stability and convergence speed.

An Improved Active Noise Control Algorithm Without Secondary Path Identification Based on the Frequency-Domain Subband Architecture

Common active noise control (ANC) algorithms need to identify the secondary path transfer functions between the output of the adaptive control filters and the error sensors, and then use the information to guide the direction of control filter coefficient updating. Recently, Zhou et al. proposed an ANC algorithm without secondary path identification, and we improve their algorithm in this paper. For single-tone and narrowband noise control, the direction of control filter coefficient updating has four choices 180deg, 0deg, and plusmn90deg. We test the four update directions and select the one that works the best. If for all four update directions, the system converges slowly or diverges, we adjust the step size and test again with the new step size. The multitone and broadband noise control problems are converted into several single-tone and narrowband noise control problems by means of a frequency-domain delayless subband architecture. Compared to Zhous algorithm, our proposed method yields good performance and converges quickly. Simulation results confirm the effectiveness of our proposed algorithm.

A waveform synthesis algorithm for active control of transformer noise: implementation

Abstract For the active control of the transformer noise, a newly developed adaptive algorithm based on waveform synthesis was proposed in [19], where a comparison of the performance of the proposed algorithm with the FXLMS algorithm made on a single channel system showed the feasibility of the algorithm. This paper describes the implementation of the proposed algorithm on a multiple channel adaptive control system, which is used to control the noise radiated by a small transformer in an anechoic chamber. The implementation shows that the proposed algorithm requires less memory and less computation load than a typical implementation of the FXLMS algorithm and that a controller realised with the proposed algorithm can effectively reduce transformer noise and be quite robust.

Sound absorption of a finite micro-perforated panel backed by a shunted loudspeaker

Deep back cavities are usually required for micro-perforated panel (MPP) constructions to achieve good low frequency absorption. To overcome the problem, a close-box loudspeaker with a shunted circuit is proposed to substitute the back wall of the cavity of the MPP constructions to constitute a composite absorber. Based on the equivalent circuit model, the acoustic impedance of the shunted loudspeaker is formulated first, then a prediction model of the sound absorption of the MPP backed by shunted loudspeaker is developed by employing the mode solution of a finite size MPP coupled by an air cavity with an impendence back wall. The MPP absorbs mid to high frequency sound, and with properly adjusted electrical parameters of its shunted circuit, the shunted loudspeaker absorbs low frequency sound, so the composite absorber provides a compact solution to broadband sound control. Numerical simulations and experiments are carried out to validate the model.

Stereo Acoustic Echo Cancellation Employing Frequency-Domain Preprocessing and Adaptive Filter

This paper proposes a windowing frequency domain adaptive filter and an upsampling block transform preprocessing to solve the stereo acoustic echo cancellation problem. The proposed adaptive filter uses windowing functions with smooth cutoff property to reduce the spectral leakage during filter updating, so the utilization of the independent noise introduced by preprocessing in stereo acoustic echo cancellation can be increased. The proposed preprocessing is operated in short blocks with low processing delay, and it uses frequency-domain upsampling to meet the minimal block length requirement given by the band limit of simultaneous masking. Therefore, the simultaneous masking can be well utilized to improve the audio quality. The acoustic echo cancellation simulations and the audio quality evaluation show that, the proposed windowing frequency domain adaptive filter performs better than the conventional frequency domain adaptive filter in both mono and stereo cases, and the upsampling block transform preprocessing provides better audio quality and stereo acoustic echo cancellation performance than the half-wave preprocessing at the same noise level.

Lattice form adaptive infinite impulse response filtering algorithm for active noise control

In some situations of active noise control, infinite impulse response (IIR) filters are more suitable than finite impulse response (FIR) filters owing to the poles in the transfer function. A number of algorithms have been derived for applying IIR filters in active noise control; however, most of them use the direct form IIR filter structure, which faces the difficulties of checking stability and relatively slow convergence speed for noise composed of narrow-band components with large power disparity. To overcome these difficulties along with using the direct form IIR filters, a new adaptive algorithm is proposed in this paper, which uses and updates the lattice form adaptive IIR filter in an active noise control system. Full mathematical derivations of the proposed algorithm are presented, and the comparison between the proposed algorithm and the commonly used filtered-u LMS and filtered-v LMS algorithms shows the superiority of the proposed algorithm.

Time Difference of Arrival Estimation Exploiting Multichannel Spatio-Temporal Prediction

To localize sound sources in room acoustic environments, time differences of arrival (TDOA) between two or more microphone signals must be determined. This problem is often referred to as time delay estimation (TDE). The multichannel cross-correlation-coefficient (MCCC) algorithm, which is an extension of the traditional cross-correlation method from two- to multiple-channel cases, exploits spatial information among multiple microphones to improve the robustness of TDE. In this paper, we propose a multichannel spatio-temporal prediction (MCSTP) algorithm, which can be viewed as a generalization of the MCCC principle from using only spatial information to using both spatial and temporal information. A recursive version of this new algorithm is then developed, which can achieve similar performance as MCSTP, but is computationally more efficient. Experimental results in reverberant and noisy environments demonstrate the advantages of this new method for TDE.

An Overlap-Save Frequency-Domain Implementation of the Delayless Subband ANC Algorithm

An efficient active noise control algorithm based on the delay less subband adaptive filter was previously proposed by Park (ldquoA delayless subband active noise control system for wideband noise control, IEEE transactions on speech and audio processing, vol. 9, no. 8, pp. 892-897, November 2001), whereby the computational complexity is reduced significantly by using a short impulse response filter to model the secondary path transfer function in a subband-decomposed form. In this correspondence, an overlap-save frequency-domain implementation of the algorithm is proposed. Simulations show that the proposed implementation has better performance than the original.