Overview

Frequency domain subband adaptive filter

Discussion

Subband based mono echo canceler / adaptive filter. This module works in conjunction with the WOLA filterbank modules and implements an adaptive filter within each subband. Each subband operates independently which provides fast adaptation time and excellent cancellation.

The length of the effective echo tail equals the number of subband taps times the block size at the input to the WOLA analysis module. For example, assume that the WOLA analysis module has an input block size of 256 samples and an FFT size of 512 samples. Then if the subband echo canceler has 10 taps, this corresponds to an effective tail length of 10 x 256 = 2560 samples.

The module has two input pins. dataIn is the desired response (typically the microphone signal in an AEC). The second input is the mono reference signal.

At instantiation time you specify the length of each subband filter. Then at run time, you specify the number of taps currently used. This allows you to easily vary the filter length at run time and verify performance and CPU load. On some systems - especially those with caches - the computational load of the filter is a function of both the length of each subband and the number of taps currently in use. For most efficient operation, the length of the filter should equal the number of taps in use.

The smoothingTime variable controls the adaptation speed of the filter. smoothingTime is in units of seconds and usually in the range of 10 to 20 seconds. The higher the smoothingTime the slower the filter will adapt but it will achieve higher overall cancellation.

The module has optional residual noise suppression which is enabled via the enableRNS variable. By default, residual noise suppression is on and the output of the echo canceler (the error signal) is further reduced in level. The residual noise suppression is configured by a 7 element interpolation table. This table operates just like the Table X-Y module with configurable (X, Y) lookup points. The lookup table specifies values in dB for the input and output function.

Type Definition

typedef struct _ModuleSbAECV1
{
    ModuleInstanceDescriptor instance;            // Common Audio Weaver module instance structure
    INT32 maxTaps;                                // Maximum length of the filter
    INT32 numTaps;                                // Current length of the filter
    FLOAT32 V;                                    // Adaptation constant
    FLOAT32 smoothingTime;                        // Time constant of the smoothing process.
    FLOAT32 A;                                    // Smoothing factor
    FLOAT32 p0;                                   // Starting value for covariance estimate
    INT32 numIter;                                // Number of iterations of adaptive loop
    INT32 enableRNS;                              // Enable residual noise suppression
    INT32 stateIndex;                             // Circular buffer index. [0 numTaps-1]
    INT32 RNS_numPoints;                          // Number of values in the lookup table. The total table size is [maxPoints 2].
    INT32 RNS_order;                              // Order of the interpolation. This can be either 2 (for linear) or 4 (for pchip).
    FLOAT32* WW1;                                 // Adaptive filter coefficients.
    FLOAT32* P;                                   // Real estimation error covariance matrix.
    FLOAT32* CC;                                  // Complex state variables (previous inputs).
    FLOAT32* Pplus;                               // Intermediate result (real scratch).
    FLOAT32* Wplus;                               // Intermediate result (complex scratch).
    FLOAT32* energyC;                             // Intermediate result (real scratch).
    FLOAT32* mu;                                  // Intermediate result (real scratch).
    FLOAT32* W2;                                  // Residual noise suppression weights.
    FLOAT32* RNS_table;                           // Lookup table. The first row is the X values and the second row is the Y values.
    FLOAT32* RNS_polyCoeffs;                      // Interpolation coefficients returned by the grid control.
} ModuleSbAECV1Class;

Variables

Properties

Name	Type	Usage	isHidden	Default value	Range	Units
maxTaps	int	const	0	16	Unrestricted	samples
numTaps	int	parameter	0	16	1:1:16	samples
V	float	const	0	0.5	Unrestricted
smoothingTime	float	parameter	0	10	0.1:1000	sec
A	float	derived	0	1	0:1
p0	float	parameter	0	1	Unrestricted
numIter	int	parameter	0	1	1:20
enableRNS	int	parameter	0	1	0:1:1
stateIndex	int	state	1	0	Unrestricted
RNS_numPoints	int	const	1	7	4:1:1000
RNS_order	int	const	1	2	2:2:4
WW1	float*	state	0	[16 x 32]	Unrestricted
P	float*	state	0	[16 x 32]	Unrestricted
CC	float*	state	0	[16 x 32]	Unrestricted
Pplus	float*	state	0	[1 x 16]	Unrestricted
Wplus	float*	state	0	[1 x 16]	Unrestricted
energyC	float*	state	0	[16 x 1]	Unrestricted
mu	float*	state	0	[1 x 16]	Unrestricted
W2	float*	state	0	[32 x 1]	Unrestricted
RNS_table	float*	parameter	1	[2 x 7]	Unrestricted
RNS_polyCoeffs	float*	state	1	[4 x 6]	Unrestricted

Pins

Input Pins

Name: dataIn

Description: Micro input

Data type: float

Channel range: 1

Block size range: Unrestricted

Sample rate range: Unrestricted

Complex support: Complex

Name: reference

Description: Reference input

Data type: float

Channel range: 1

Block size range: Unrestricted

Sample rate range: Unrestricted

Complex support: Complex

Output Pins

Name: errorSignal

Description: Error signal = difference between filter output and dataIn

Data type: float

MATLAB Usage

File Name: sb_aec_v1_module.m

 M = sb_aec_v1_module(NAME, MAXTAPS)
 Frequency domain subband echo canceler using a Kalman filter based
 algorithm.  Arguments:
    NAME - name of the module.
    MAXTAPS - maximum length of each frequency domain complex adaptive filter