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Overview

Cascade of second order filters realized as a single module

Discussion

This module implements a general purpose second order cascade filter that is capable of realizing many different filter types. The module performs internal smoothing allowing the filters to be updated without introducing clicks. In all cases, the module implements an underlying cascade second order filters. First order filters are realized by setting some of the second order coefficients to zero.

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On the processors of 2159x where 8 IIR accelerators are available with dual SHARC+ cores, 4 IIR accelerators are statically reserved for each core. The processing load is distributed as IIRA channels = floorf(numChannels/3) * 2. i.e. if the numChannels are 10 then 6 channels are processed in 4 IIR accelerators in parallel with 4 channels in core processing. On SHARC+ processor, additional memory is allocated for accelerator TCB with size of numAcceleratorChannels*13. Also separate coefficient/state buffer is allocated as per the accelerator requirement.

Type Definition

Code Block
typedef struct _ModuleSecondOrderFilterSmoothedCascadeV2
{
    ModuleInstanceDescriptor instance;            // Common Audio Weaver module instance structure
    INT32 numStages;                              // Number of cascaded stages of the second order filter.
    FLOAT32 smoothingTime;                        // Time constant of the smoothing process.
    INT32 updateActive;                           // Specifies whether the filter coefficients are updating (=1) or fixed (=0).
    FLOAT32 smoothingCoeff;                       // Smoothing coefficient. This is computed based on the smoothingTime, sample rate, and block size of the module.
    INT32 bulkParamsUpdate;                       // State variable to handle bulk changes in parameters.
    INT32* filterType;                            // Selects the type of filter that is implemented by the module: Bypass=0, Gain=1, Butter1stLPF=2, Butter2ndLPF=3, Butter1stHPF=4, Butter2ndHPF=5, Allpass1st=6, Allpass2nd=7, Shelf2ndLow=8, Shelf2ndLowQ=9, Shelf2ndHigh=10, Shelf2ndHighQ=11, PeakEQ=12, Notch=13, Bandpass=14, Bessel1stLPF=15, Bessel1stHPF=16, AsymShelf1stLow=17, AsymShelf1stHigh=18, SymShelf1stLow=19, SymShelf1stHigh=20, VariableQLPF=21, VariableQHPF=22 Resonant=23.
    FLOAT32* freq;                                // Cutoff frequency of the filter, in Hz.
    FLOAT32* gain;                                // Amount of boost or cut to apply, in dB if applicable.
    FLOAT32* Q;                                   // Specifies the Q of the filter, if applicable.
    FLOAT32* targetCoeffs;                        // Matrix of filter target coefficients. The size of the matrix is 5 x numStages. Each column contains the variables for a biquad arranged as [b0; b1; b2; a1; a2].
    FLOAT32* currentCoeffs;                       // Matrix of filter current coefficients. The size of the matrix is 5 x numStages. Each column contains the variables for a biquad arranged as [b0; b1; b2; a1; a2].
    FLOAT32* state;                               // State variables. 2*numChannels per section.
    void * hardware_specific_struct_pointer;      // This is the internal array used for ADI IIR accelerator. Size is determined internally
} ModuleSecondOrderFilterSmoothedCascadeV2Class;

Variables

Properties

Name

Type

Usage

isHidden

Default value

Range

Units

numStages

int

const

0

2

1:1:32

smoothingTime

float

parameter

0

10

0:1:1000

msec

updateActive

int

parameter

1

1

0:1

smoothingCoeff

float

derived

1

0.06449

Unrestricted

bulkParamsUpdate

int

state

1

0

Unrestricted

filterType

int*

parameter

0

[1 x 2]

0:23

freq

float*

parameter

0

[1 x 2]

10:0.1:23990

Hz

gain

float*

parameter

0

[1 x 2]

-24:0.1:24

dB

Q

float*

parameter

0

[1 x 2]

0:0.1:20

targetCoeffs

float*

derived

0

[5 x 2]

Unrestricted

currentCoeffs

float*

state

0

[5 x 2]

Unrestricted

state

float*

state

1

[2 x 2]

Unrestricted

hardware_specific_struct_pointer

void *

state

1

Unrestricted

Pins

Input Pins

Name: in

Description: audio input

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Sample rate range: Unrestricted

Complex support: Real

Output Pins

Name: out

Description: audio output

Data type: float

MATLAB Usage

File Name: second_order_filter_cascade_v2_module.m

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