Overview

Decimator combining an FIR filter and a downsampler

Discussion

The FIR decimator module implements an FIR filter followed by a downsampler. An efficient polyphase algorithm is used which reduces the processing load. The module operates on multiple channels with all channels sharing the same set of filter coefficients. The filter coefficients are stored in the array coeffs in normal order. Each channel uses a circular buffer of length N+D-1, where D is the decimation factor.

The block size of the input pin must be divisible by the decimation factor D. This condition is checked in the prebuild function. The output block size equals the input block size divided by D. The output sample rate equals the input sample rate divided by D.

The latency through the module in samples is displayed in the Audio Weaver Designer GUI. The underlying FIR processing occurs at the input sampling rate and the latency is the delay in samples at the higher sampling rate.orate downsampling.

Type Definition

typedef struct _ModuleFIRDecimator
{
    ModuleInstanceDescriptor instance;            // Common Audio Weaver module instance structure
    INT32 D;                                      // Downsampling factor.
    INT32 N;                                      // Length of the filter.
    INT32 stateIndex;                             // Index of the oldest state variable in the array of state variables.
    INT32 stateLen;                               // Length of the circular buffer used by each channel.
    FLOAT32* coeffs;                              // Filter coefficient array in normal order.
    FLOAT32* state;                               // State variable array. The size of the array equals stateLen*numChannels.
    void * hardware_specific_struct_pointer;      // Usually NULL, but for optimizations that use 3rd party optimizations (like ne10), this may point to the required instance structure
} ModuleFIRDecimatorClass;

Variables

Properties

Name	Type	Usage	isHidden	Default value	Range	Units
D	int	const	0	2	2:1:512
N	int	const	0	32	1:1:5000	samples
stateIndex	int	state	1	0	Unrestricted
stateLen	int	const	1	33	Unrestricted
coeffs	float*	parameter	0	[32 x 1]	Unrestricted
state	float*	state	1	[33 x 1]	Unrestricted
hardware_specific_struct_pointer	void *	state	1		Unrestricted

Pins

Input Pins

Name: in

Description: audio input

Data type: float

Channel range: Unrestricted

Block size range: Unrestricted

Sample rate range: Unrestricted

Complex support: Real

Output Pins

Name: out

Description: audio output