Overview

LMS Norm fract32 filter

Discussion

The LMS fract32 module implements a single channel, N-order FIR adaptive filter using a Normalized Least Means Squares algorithm. 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 adaptation constant mu is visible as a user variable.

Type Definition

typedef struct _ModuleLMSNormFract32
{
    ModuleInstanceDescriptor instance;            // Common Audio Weaver module instance structure
    INT32 numTaps;                                // Length of the filter
    FLOAT32 mu;                                   // mu Step size
    INT32 stateIndex;                             // Index of the oldest state variable in the array of state variables
    INT32 prev_stateptr;                          // Index of the oldest state variable in the array of state variables
    INT32 postShift;                              // Coefficient Scaling
    fract32 mufract32;                            // mu Step size
    fract32 E;                                    // saves previous frame energy
    fract32 x0;                                   // saves previous input sample
    fract32* coeffsfract32;                       // Coefficient array
    fract32* state;                               // State variable array
} ModuleLMSNormFract32Class;

Variables

Properties

Name	Type	Usage	isHidden	Default value	Range	Units
numTaps	int	const	0	31	1:1:5000	samples
mu	float	parameter	0	0.02	0:0.999	linear
stateIndex	int	state	1	0	Unrestricted
prev_stateptr	int	state	1	0	Unrestricted
postShift	int	derived	1	0	Unrestricted
mufract32	fract32	derived	1	0.02	Unrestricted
E	fract32	state	1	0	Unrestricted
x0	fract32	state	1	0	Unrestricted
coeffsfract32	fract32*	derived	0	[31 x 1]	Unrestricted
state	fract32*	state	1	[62 x 1]	Unrestricted

Pins

Input Pins

Name: in1

Description: audio input

Data type: fract32

Channel range: 1

Block size range: Unrestricted

Sample rate range: Unrestricted

Complex support: Real

Name: in2

Description: desired input

Data type: fract32

Channel range: 1

Block size range: Unrestricted

Sample rate range: Unrestricted

Complex support: Real

Output Pins

Name: out1

Description: audio output

Data type: fract32

Name: out2

Description: error output

Data type: fract32

MATLAB Usage

File Name: lms_norm_fract32_module.m

 M=lms_norm_fract32_module(NAME, MAXTAPS)
 Creates a normalized LMS FIR filter for use in the Audio Weaver environment.
 The module has 2 single channel input pins and 2 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