sound:filter - filter a signal
Last update: 31.07.2026Version: 0.0.0
Description
Filters a 1D signal, using a linear, time-invariant digital filter. The filter is an Infinite Impulse Response (IIR) filter, defined by its vectors of coefficients B (numerator) and A (denominator) in the z-domain. If A contains only one coefficient, equal to 1, it is a Finite Impulse Response (FIR) filter. If A is not specified, it will be set to value 1. If the first value of the A coefficients (A[0]) is not set to 1.0, all coefficient values will be divided by A[0] to normalize the filter.
Inputs
| Input | Name | Expected type(s) | Description |
|---|---|---|---|
| Pin 0 Required |
signal_to_filter | field |
Signal to filter. |
| Pin 1 Required |
filter_b_coefficients | vector<double> |
Coefficients B of the filter. |
| Pin 2 | filter_a_coefficients | vector<double> |
(Optional) Coefficients A of the filter. |
Outputs
| Output | Name | Expected type(s) | Description |
|---|---|---|---|
| Pin 0 | filtered_signal | field |
Filtered signal, as a field. |
Configurations
| Name | Expected type(s) | Default value | Description |
|---|---|---|---|
| mutex | bool |
false | If this option is set to true, the shared memory is prevented from being simultaneously accessed by multiple threads. |
Scripting
Category: sound
Plugin: sound
Scripting name: None
Full name: None
Internal name: filter_signal
License: avrxp_snd_level1
Changelog
- Version 0.0.0: Initial release.
