LMSNormFract32

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.

Module Pack

Advanced

ClassID

classID = 1293

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