FilterbankFIRV2
Overview
Efficient filterbank based FIR filter
Discussion
DSP Concepts Proprietary IP.
This module implements a long FIR filter using filterbank processing.
This yields a significant reduction in the processing load and allows much longer filters to be implemented.
The filterbank reduces computation by breaking a large convolution into multiple smaller independent convolutions.
The implementation supports mono audio channels and an arbitrary block size.
When you instantiate the module you specify the blockSize and the length of the filter L.
The larger the blockSize the more efficient the processing.
The latency through the filter is zero and is numerically equivalent to having a time domain FIR filter.
On the Inspector (or Numeric Inspector) you specify the time domain FIR filter coefficients.
The module will automatically take care of converting the time domain coefficients into the appropriate frequency domain coefficients.
For ease of use, it is better to use the Numeric Inspector and then load the coefficients from a text file.
Module Pack
Advanced
ClassID
classID =
Type Definition
typedef struct _ModuleFilterbankFIRV2
{
ModuleInstanceDescriptor instance; // Common Audio Weaver module instance structure
} ModuleFilterbankFIRV2Class;
Variables
Properties
| Name |
Type |
Usage |
isHidden |
Default Value |
Range |
Units |
| L |
int |
const |
0 |
1024 |
Unrestricted |
|
| FFTSize |
int |
const |
0 |
256 |
Unrestricted |
|
| numBlocks |
int |
const |
0 |
4 |
Unrestricted |
|
| coeffs |
float* |
parameter |
0 |
[1024 x 1] |
Unrestricted |
|
Pins
|
|
| Name |
in |
| Description |
Time domain input |
| Data type |
float |
| Channel range |
1 |
| Block size range |
256 |
| Sample rate range |
Unrestricted |
| Complex support |
Real |
Output Pins
|
|
| Name |
out |
| Description |
Time domain output |
| Data Type |
float |
Scratch Pins
|
|
| Channel Count |
1 |
| Block size |
256 |
| Sample rate |
48000 |
|
|
| Channel Count |
1 |
| Block size |
256 |
| Sample rate |
48000 |
|
|
| Channel Count |
1 |
| Block size |
257 |
| Sample rate |
48000 |
|
|
| Channel Count |
1 |
| Block size |
256 |
| Sample rate |
48000 |
|
|
| Channel Count |
1 |
| Block size |
256 |
| Sample rate |
48000 |
|
|
| Channel Count |
1 |
| Block size |
256 |
| Sample rate |
48000 |
|
|
| Channel Count |
1 |
| Block size |
512 |
| Sample rate |
48000 |
Matlab Usage
File Name: filterbank_fir_v2_subsystem.m
SYS = filterbank_fir_v2_subsystem(NAME, L, BLOCKSIZE)
Creates a subsystem which implements a long FIR filter efficiently using
a filterbank based algorithm. Arguments:
NAME - name of the subsystem
L - length of the time domain FIR filter
BLOCKSIZE - time domain block size. This allocates sizes of internal
modules and the prebuild function checks that this matches the
actual block size.
This v2 version of the module no longer requires that the input
block size is a power of 2.
The function internally uses a combination of Audio Weaver blocks
in order to achieve the desired filtering. The input and output
audio are mono.
Copyright (c) 2026 DSP Concepts, Inc.