LMS

Overview

Normalized LMS FIR filter

Discussion

The LMS module implements a single channel N-order FIR adaptive filter.
The module takes 2 single channel inputs: desired filter output and filter input (the reference signal).
For every data sample the filter output is compared to the desired output and this information is used to update the filter taps according to the normalized LMS algorithm. The module always has 2 single channel ouputs: adaptive filter output and error signal, (desired filter output - actual filter output. The module also optionally outputs the filter coefficients on a third output pin. The array coeffs holds the filter coefficients and is of size numTaps x 1. The coefficients are stored in normal order. The state variable array is of size numTaps x 1. The adaptation constant mu is visible as a parameter.

Module Pack

Advanced

ClassID

classID = 1292

Type Definition

typedef struct _ModuleLMS
{
ModuleInstanceDescriptor instance;            // Common Audio Weaver module instance structure
INT32 maxTaps;                                // Maximum length of the filter
INT32 numTaps;                                // Current length of the filter
FLOAT32 mu;                                   // Adaptation constant
INT32 stateIndex;                             // Index of the oldest state variable in the array of state variables
FLOAT32* coeffs;                              // Coefficient array
FLOAT32* state;                               // State variable array
} ModuleLMSClass;

Variables

Properties

Name	Type	Usage	isHidden	Default Value	Range	Units
maxTaps	int	const	0	1024	1:1:5000	samples
numTaps	int	parameter	0	32	1:1:1024	samples
mu	float	parameter	0	0	0:1	linear
stateIndex	int	state	1	0	Unrestricted
coeffs	float*	derived	0	[1024 x 1]	Unrestricted
state	float*	state	1	[1027 x 1]	Unrestricted

Pins

Input Pins


Name	dataIn
Description	Filter input
Data type	float
Channel range	1
Block size range	Unrestricted
Sample rate range	Unrestricted
Complex support	Real


Name	reference
Description	Reference input: desired filter output
Data type	float
Channel range	1
Block size range	Unrestricted
Sample rate range	Unrestricted
Complex support	Real

Output Pins


Name	dataOut
Description	Filter output
Data Type	float


Name	errorSignal
Description	Error signal = difference between filter output and reference input
Data Type	float


Name	coeffs
Description	Filter coefficients
Data Type	float

Matlab Usage

File Name: lms_module.m 
 M=lms_module(NAME, MAXTAPS, OUTPUTCOEFFS) 
 Creates a normalized LMS FIR filter for use in the Audio Weaver environment. 
 The module has 2 single channel input pins and up to 3 single channel  
 output pins:  
    Input Pin 1:  dataIn  - Adaptive filter desired output 
    Input Pin 2:  reference - Adaptive filter input 
    Output Pin 1:  dataOut - output of the adaptive filter 
    Output Pin 2:  errorSignal - difference between dataIn and dataOut 
    Output Pin 3:  Coefficients (optional) 

 Arguments: 
    NAME - name of the module. 
    MAXTAPS - maximum length of the filter (number of taps). 
    OUTPUTCOEFFS - Boolean which specifies whether a third output 
       pin holding the coefficients should be added. By default, 
       OUTPUTCOEFFS=0.