Simulink Desktop Real-Time and Least Mean Square adaptive algorithm

Question

I need to implement an LMS-based adaptive audio-cancellation algorithm on the Simulink Desktop Real-Time toolbox.

The physical system is composed of a microphone recording a noise source and another microphone recording the residual noise after the control process (antinoise being injected by a speaker controlled by Simulink).

For the (adaptive) LMS algorithm to work properly I need to be able to work on a sample-by-sample basis, that is at each sampled time instant I need to update the adaptive filter using the synchronised current sample value of both microphones. I realise some delay is inevitable but I was wondering whether it's possible on Simulink Desktop Real-Time to reduce the buffer size of the inputs to one sample and thus work on a sample-by-sample basis.

Thanks for your help in advance.

Answer 1

You can always implement the filter on a sample by sample basis.

But you still need a history of input values to perform the actual LMS calculation on. On a sample by sample basis this would just mean using a simple FIFO buffer.

If you have access to the DSP Toolbox then there is already an LMS Filter block that will do this for you.