Evolutionary and variable step size strategies for multichannel filtered-x affine projection algorithms

This study is focused on the necessity to improve the performance of the affine projection (AP) algorithm for active noise control (ANC) applications. The proposed algorithms are evaluated regarding their steady-state behaviour, their convergence speed and their computational complexity. To this end, different strategies recently applied to the AP for channel identification are proposed for multichannel ANC. These strategies are based either on a variable step size, an evolving projection order, or the combination of both strategies. The developed efficient versions of the AP algorithm use the modified filtered-x structure, which exhibits faster convergence than other filtering schemes. Simulation results show that the proposed approaches exhibit better performance than the conventional AP algorithm and represent a meaningful choice for practical multichannel ANC applications.This work was supported by a grant of the Romanian National Authority for Scientific Research, CNCS-UEFISCDI, project number PN-II-ID-PCE-2011-3-0097, Spanish Ministerio de Ciencia e Innovacion TEC2009-13741 and Generalitat Valenciana PROMETEO 2009/2013.Gonzalez, A.; Albu, F.; Ferrer Contreras, M.; Diego Antón, MD. (2013). Evolutionary and variable step size strategies for multichannel filtered-x affine projection algorithms. IET Signal Processing. 7(6):471-476. https://doi.org/10.1049/iet-spr.2012.0213S47147676Shin, H.-C., Sayed, A. H., & Song, W.-J. (2004). Variable Step-Size NLMS and Affine Projection Algorithms. IEEE Signal Processing Letters, 11(2), 132-135. doi:10.1109/lsp.2003.821722Paleologu, C., Benesty, J., & Ciochina, S. (2008). A Variable Step-Size Affine Projection Algorithm Designed for Acoustic Echo Cancellation. IEEE Transactions on Audio, Speech, and Language Processing, 16(8), 1466-1478. doi:10.1109/tasl.2008.2002980Shin, H.-C., & Sayed, A. H. (2004). Mean-Square Performance of a Family of Affine Projection Algorithms. IEEE Transactions on Signal Processing, 52(1), 90-102. doi:10.1109/tsp.2003.820077Kong, S.-J., Hwang, K.-Y., & Song, W.-J. (2007). An Affine Projection Algorithm With Dynamic Selection of Input Vectors. IEEE Signal Processing Letters, 14(8), 529-532. doi:10.1109/lsp.2007.891325Seong-Eun Kim, Se-Jin Kong, & Woo-Jin Song. (2009). An Affine Projection Algorithm With Evolving Order. IEEE Signal Processing Letters, 16(11), 937-940. doi:10.1109/lsp.2009.2027638Kim, K.-H., Choi, Y.-S., Kim, S.-E., & Song, W.-J. (2011). An Affine Projection Algorithm With Periodically Evolved Update Interval. IEEE Transactions on Circuits and Systems II: Express Briefs, 58(11), 763-767. doi:10.1109/tcsii.2011.2168023Bouchard, M. (2003). Multichannel affine and fast affine projection algorithms for active noise control and acoustic equalization systems. IEEE Transactions on Speech and Audio Processing, 11(1), 54-60. doi:10.1109/tsa.2002.805642Kong, N., Shin, J., & Park, P. (2011). A two-stage affine projection algorithm with mean-square-error-matching step-sizes. Signal Processing, 91(11), 2639-2646. doi:10.1016/j.sigpro.2011.06.003MoonSoo Chang, NamWoong Kong, & PooGyeon Park. (2010). An Affine Projection Algorithm Based on Reuse Time of Input Vectors. IEEE Signal Processing Letters, 17(8), 750-753. doi:10.1109/lsp.2010.2053355Arablouei, R., & Doğançay, K. (2012). Affine projection algorithm with selective projections. Signal Processing, 92(9), 2253-2263. doi:10.1016/j.sigpro.2012.02.018Gonzalez, A., Ferrer, M., de Diego, M., & Piñero, G. (2012). An affine projection algorithm with variable step size and projection order. Digital Signal Processing, 22(4), 586-592. doi:10.1016/j.dsp.2012.03.00

Performance improvement of adaptive filters for echo cancellation applications

This work focuses on performance improvement of adaptive algorithms for both line and acoustic echo cancellation applications. Echo in telephone networks, Line Echo, is observed naturally due to impedance mismatches at the long-distance/local-loop interface. Acoustic echo is due to the acoustic coupling between the microphone and the speaker of a speakerphone. The Affine Projection (APA) and the Fast Affine Projection (FAP) algorithms are two examples of reliable and efficient adaptive filters used for echo cancellation...This thesis presents, Variable Regularized Fast Affine Projections (VR-FAP) algorithm, with a varying, optimal regularization value which provides the desirable property of both fast and low misadjustment of the filter --Abstract, page iii

Acoustic Echo Cancellation and their Application in ADF

In this paper, we present an overview of the principal, structure and the application of the echo cancellation and kind of application to improve the performance of the systems. Echo is a process in which a delayed and distorted version o the original sound or voice signal is reflected back to the source. For the acoustic echo canceller much and more study are required to make the good tracking speed fast and reduce the computational complexity. Due to the increasing the processing requirement, widespread implementation had to wait for advances in LSI, VLSI echo canceller appeared. DOI: 10.17762/ijritcc2321-8169.150513

A Practical Data-Reuse Adaptive Algorithm for Acoustic Echo Cancellation

Publication in the conference proceedings of EUSIPCO, Bucharest, Romania, 201

Adaptive Algorithms for Intelligent Acoustic Interfaces

Modern speech communications are evolving towards a new direction which involves users in a more perceptive way. That is the immersive experience, which may be considered as the “last-mile” problem of telecommunications. One of the main feature of immersive communications is the distant-talking, i.e. the hands-free (in the broad sense) speech communications without bodyworn or tethered microphones that takes place in a multisource environment where interfering signals may degrade the communication quality and the intelligibility of the desired speech source. In order to preserve speech quality intelligent acoustic interfaces may be used. An intelligent acoustic interface may comprise multiple microphones and loudspeakers and its peculiarity is to model the acoustic channel in order to adapt to user requirements and to environment conditions. This is the reason why intelligent acoustic interfaces are based on adaptive filtering algorithms. The acoustic path modelling entails a set of problems which have to be taken into account in designing an adaptive filtering algorithm. Such problems may be basically generated by a linear or a nonlinear process and can be tackled respectively by linear or nonlinear adaptive algorithms. In this work we consider such modelling problems and we propose novel effective adaptive algorithms that allow acoustic interfaces to be robust against any interfering signals, thus preserving the perceived quality of desired speech signals. As regards linear adaptive algorithms, a class of adaptive filters based on the sparse nature of the acoustic impulse response has been recently proposed. We adopt such class of adaptive filters, named proportionate adaptive filters, and derive a general framework from which it is possible to derive any linear adaptive algorithm. Using such framework we also propose some efficient proportionate adaptive algorithms, expressly designed to tackle problems of a linear nature. On the other side, in order to address problems deriving from a nonlinear process, we propose a novel filtering model which performs a nonlinear transformations by means of functional links. Using such nonlinear model, we propose functional link adaptive filters which provide an efficient solution to the modelling of a nonlinear acoustic channel. Finally, we introduce robust filtering architectures based on adaptive combinations of filters that allow acoustic interfaces to more effectively adapt to environment conditions, thus providing a powerful mean to immersive speech communications

Adaptive Algorithms for Intelligent Acoustic Interfaces

Modern speech communications are evolving towards a new direction which involves users in a more perceptive way. That is the immersive experience, which may be considered as the “last-mile” problem of telecommunications. One of the main feature of immersive communications is the distant-talking, i.e. the hands-free (in the broad sense) speech communications without bodyworn or tethered microphones that takes place in a multisource environment where interfering signals may degrade the communication quality and the intelligibility of the desired speech source. In order to preserve speech quality intelligent acoustic interfaces may be used. An intelligent acoustic interface may comprise multiple microphones and loudspeakers and its peculiarity is to model the acoustic channel in order to adapt to user requirements and to environment conditions. This is the reason why intelligent acoustic interfaces are based on adaptive filtering algorithms. The acoustic path modelling entails a set of problems which have to be taken into account in designing an adaptive filtering algorithm. Such problems may be basically generated by a linear or a nonlinear process and can be tackled respectively by linear or nonlinear adaptive algorithms. In this work we consider such modelling problems and we propose novel effective adaptive algorithms that allow acoustic interfaces to be robust against any interfering signals, thus preserving the perceived quality of desired speech signals. As regards linear adaptive algorithms, a class of adaptive filters based on the sparse nature of the acoustic impulse response has been recently proposed. We adopt such class of adaptive filters, named proportionate adaptive filters, and derive a general framework from which it is possible to derive any linear adaptive algorithm. Using such framework we also propose some efficient proportionate adaptive algorithms, expressly designed to tackle problems of a linear nature. On the other side, in order to address problems deriving from a nonlinear process, we propose a novel filtering model which performs a nonlinear transformations by means of functional links. Using such nonlinear model, we propose functional link adaptive filters which provide an efficient solution to the modelling of a nonlinear acoustic channel. Finally, we introduce robust filtering architectures based on adaptive combinations of filters that allow acoustic interfaces to more effectively adapt to environment conditions, thus providing a powerful mean to immersive speech communications

Study of L0-norm constraint normalized subband adaptive filtering algorithm

Limited by fixed step-size and sparsity penalty factor, the conventional sparsity-aware normalized subband adaptive filtering (NSAF) type algorithms suffer from trade-off requirements of high filtering accurateness and quicker convergence behavior. To deal with this problem, this paper proposes variable step-size L0-norm constraint NSAF algorithms (VSS-L0-NSAFs) for sparse system identification. We first analyze mean-square-deviation (MSD) statistics behavior of the L0-NSAF algorithm innovatively in according to a novel recursion form and arrive at corresponding expressions for the cases that background noise variance is available and unavailable, where correlation degree of system input is indicated by scaling parameter r. Based on derivations, we develop an effective variable step-size scheme through minimizing the upper bounds of the MSD under some reasonable assumptions and lemma. To realize performance improvement, an effective reset strategy is incorporated into presented algorithms to tackle with non-stationary situations. Finally, numerical simulations corroborate that the proposed algorithms achieve better performance in terms of estimation accurateness and tracking capability in comparison with existing related algorithms in sparse system identification and adaptive echo cancellation circumstances.Comment: 15 pages,15 figure