A transform domain optimization to increase the convergence speed of the multichannel filtered-X least-mean-square algorithm

Bouchard, M.; Paillard, Bruno

doi:10.1121/1.417204

Cited by 10 publications

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“…These include decreasing the convergence sensitivity to ill-conditioning by transforming the broadband input signal to the frequency domain with a discrete Fourier or cosine transformation [16]. This approach was "rst described in the context of a single-input/single-output adaptive "lter [17}19] to increase the convergence speed when the input signal was not spectrally white.…”

Section: Introductionmentioning

confidence: 99%

A Principal Component Algorithm for Feedforward Active Noise and Vibration Control

Cabell

Fuller

1999

Journal of Sound and Vibration

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A principal component least mean square (PC-LMS) adaptive algorithm is described that has considerable benefits for large control systems used to implement feedforward control of single frequency disturbances. The algorithm is a transform domain version of the multichannel filtered-x LMS algorithm. The transformation corresponds to the principal components (PCs) of the transfer function matrix between the sensors and actuators in a control system at a single frequency. The method is similar to other transform domain LMS algorithms because the transformation can be used to accelerate convergence when the control system is ill-conditioned. This ill-conditioning is due to actuator and sensor placement on a continuous structure. The principal component transformation rotates the control filter coefficient axes to a more convenient coordinate system where (1) independent convergence factors can be used on each coordinate to accelerate convergence, (2) insignificant control coordinates can be eliminated from the controller, and (3) coordinates that require excessive control effort can be eliminated from the controller. The resulting transform domain algorithm has lower computational requirements than the filtered-x LMS algorithm. The formulation of the algorithm given here applies only to single frequency control problems, and computation of the decoupling transforms requires an estimate of the transfer function matrix between control actuators and error sensors at the frequency of interest. The robustness of the PC-LMS algorithm to modeling errors is shown to be identical to the filtered-x LMS algorithm, and the statistical properties of the principal components are used to select a subset of the PCs to control, depending on the control application. The variation of the PCs with frequency is investigated, and an online identification procedure is described to compute the transfer function matrix while the controller is operating. The feasibility of the PC-LMS method was demonstrated in real-time noise control experiments involving 48 microphones and 12 control actuators mounted on a closed cylindrical shell. Convergence of the PC-LMS algorithm was more stable than the filtered-x LMS algorithm. In addition, the PC-LMS controller produced more noise reduction with less control effort than the filtered-x LMS controller in several tests.

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Section: Introductionmentioning

confidence: 99%

A Principal Component Algorithm for Feedforward Active Noise and Vibration Control

Cabell

Fuller

1999

Journal of Sound and Vibration

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“…Also in [109], the self-orthogonalizing frequency domain adaptive filter is studied, which improves the convergence speed. It uses the autocorrelation matrix to compute the filter weights and the discrete cosine transform, [110], to cancel some correlation of the input signal.…”

Section: A1 Real Time Room Equalization In Frequency Domainmentioning

confidence: 99%

Linear and nonlinear room compensation of audio rendering systems

Criado¹

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Common audio systems are designed with the intent of creating real and immersive scenarios that allow the user to experience a particular acoustic sensation that does not depend on the room he is perceiving the sound. However, acoustic devices and multichannel rendering systems working inside a room, can impair the global audio effect and thus the 3D spatial sound.In order to preserve the spatial sound characteristics of multichannel rendering techniques, adaptive filtering schemes are presented in this dissertation to compensate these electroacoustic effects and to achieve the immersive sensation of the desired acoustic system. Adaptive filtering offers a solution to the room equalization problem that is doubly interesting. First of all, it iteratively solves the room inversion problem, which can become computationally complex to obtain when direct methods are used. Secondly, the use of adaptive filters allows to follow the time-varying room conditions.In this regard, adaptive equalization (AE) filters try to cancel the echoes due to the room effects. In this work, we consider this problem and propose effective and robust linear schemes to solve this equalization problem by using adaptive filters. To do this, different adaptive filtering schemes are introduced in the AE context. These filtering schemes are based on three strategies previously introduced in the literature: the convex combination of filters, the biasing of the filter weights and the block-based filtering. More specifically, and motivated by the sparse nature of the acoustic impulse response and its corresponding optimal inverse filter, we introduce different adaptive equalization algorithms.In addition, since audio immersive systems usually require the use of multiple transducers, the multichannel adaptive equalization problem should be also taken into account when new single-channel approaches are presented, in the sense that they can be straightforwardly extended to the ii Abstract multichannel case.On the other hand, when dealing with audio devices, consideration must be given to the nonlinearities of the system in order to properly equalize the electroacoustic system. For that purpose, we propose a novel nonlinear filtered-x approach to compensate both room reverberation and nonlinear distortion with memory caused by the amplifier and loudspeaker devices.Finally, it is important to validate the algorithms proposed in a realtime implementation. Thus, some initial research results demonstrate that an adaptive equalizer can be used to compensate room distortions. ResumenLos sistemas de audio actuales están diseñados con la idea de crear escenarios reales e inmersivos que permitan al usuario experimentar determinadas sensaciones acústicas que no dependan de la sala o situación donde se esté percibiendo el sonido. Sin embargo, los dispositivos acústicos y los sistemas multicanal funcionando dentro de salas, pueden perjudicar el efecto global sonoro y de esta forma, el sonido espacial 3D.Para poder preservar las características espaciales ...

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2001

Signal Processing for Active Control

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A transform domain optimization to increase the convergence speed of the multichannel filtered-X least-mean-square algorithm

Cited by 10 publications

References 0 publications

A Principal Component Algorithm for Feedforward Active Noise and Vibration Control

A Principal Component Algorithm for Feedforward Active Noise and Vibration Control

Linear and nonlinear room compensation of audio rendering systems

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