Multidelay Block Frequency Domain Adaptive Filter
   HOME

TheInfoList



OR:

The multidelay block frequency domain adaptive filter (MDF) algorithm is a block-based frequency domain implementation of the (normalised) Least mean squares filter (LMS) algorithm.


Introduction

The MDF algorithm is based on the fact that convolutions may be efficiently computed in the frequency domain (thanks to the
fast Fourier transform A fast Fourier transform (FFT) is an algorithm that computes the discrete Fourier transform (DFT) of a sequence, or its inverse (IDFT). Fourier analysis converts a signal from its original domain (often time or space) to a representation in th ...
). However, the algorithm differs from the
fast LMS algorithm Fast or FAST may refer to: * Fast (noun), high speed or velocity * Fast (noun, verb), to practice fasting, abstaining from food and/or water for a certain period of time Acronyms and coded Computing and software * ''Faceted Application of Subje ...
in that block size it uses may be smaller than the filter length. If both are equal, then MDF reduces to the FLMS algorithm. The advantages of MDF over the (N)LMS algorithm are: * Lower algorithmic complexity * Partial de-correlation of the input (which 'may' lead to faster convergence)


Variable definitions

Let N be the length of the processing blocks, K be the number of blocks and \mathbf denote the 2Nx2N Fourier transform matrix. The variables are defined as: : \underline(\ell) = \mathbf\left \mathbf_, e(\ell N),\dots,e(\ell N-N-1) \rightT : \underline_k(\ell) = \mathrm \left\ : \underline(\ell) = \left \underline_0(\ell), \underline_1(\ell), \dots, \underline_(\ell) \right/math> : \underline(\ell) = \mathbf\left \mathbf_, d(\ell N),\dots,d(\ell N-N-1) \rightT With normalisation matrices \mathbf_1 and \mathbf_2: : \mathbf_1 = \mathbf\begin \mathbf_ & \mathbf_ \\ \mathbf_ & \mathbf_ \\ \end\mathbf^H : \tilde_2 = \mathbf\begin \mathbf_ & \mathbf_ \\ \mathbf_ & \mathbf_ \\ \end\mathbf^H : \mathbf_2 = \operatorname \left\ In practice, when multiplying a column vector \mathbf by \mathbf_1, we take the inverse FFT of \mathbf, set the first N values in the result to zero and then take the FFT. This is meant to remove the effects of the circular convolution.


Algorithm description

For each block, the MDF algorithm is computed as: : \underline(\ell) = \mathbf_1 \underline(\ell) \underline(\ell-1) : \underline(\ell) = \underline(\ell) - \underline(\ell) : \mathbf_\mathbf(\ell) = \underline^H(\ell)\underline(\ell) : \underline(\ell) = \underline(\ell-1) + \mu\mathbf_2\mathbf_\mathbf^(\ell) \underline^H(\ell) \underline(\ell) It is worth noting that, while the algorithm is more easily expressed in matrix form, the actual implementation requires no matrix multiplications. For instance the normalisation matrix computation \mathbf_\mathbf = \underline^H(\ell)\underline(\ell) reduces to an element-wise vector multiplication because \underline{\mathbf{X(\ell) is block-diagonal. The same goes for other multiplications.


References

* J.-S. Soo and K. Pang,
Multidelay block frequency domain adaptive filter
” ''IEEE Transactions on Acoustics, Speech, and Signal Processing'', vol. 38, no. 2, pp. 373–376, 1990. * H. Buchner, J. Benesty, W. Kellermann, "An Extended Multidelay Filter: Fast Low-Delay Algorithms for Very High-Order Adaptive Systems". ''Proc. IEEE
International Conference on Acoustics, Speech, and Signal Processing ICASSP, the International Conference on Acoustics, Speech, and Signal Processing, is an annual flagship conference organized of IEEE Signal Processing Society. All papers included in its proceedings have been indexed by Ei Compendex. The first ICAS ...
(ICASSP)'', 2003. * A free implementation of the MDF algorithm is available i
Speexmain source file


See also

*
Adaptive filter An adaptive filter is a system with a linear filter that has a transfer function controlled by variable parameters and a means to adjust those parameters according to an optimization algorithm. Because of the complexity of the optimization algorit ...
*
Recursive least squares Recursive least squares (RLS) is an adaptive filter algorithm that recursively finds the coefficients that minimize a weighted linear least squares cost function relating to the input signals. This approach is in contrast to other algorithms such ...
* For statistical techniques relevant to LMS filter see
Least squares The method of least squares is a standard approach in regression analysis to approximate the solution of overdetermined systems (sets of equations in which there are more equations than unknowns) by minimizing the sum of the squares of the res ...
. Digital signal processing Filter theory