How to Use FFmpeg AFIR Filter

The afir (Audio Finite Impulse Response) filter in FFmpeg is a powerful tool used for applying impulse responses (IR) to audio streams, enabling effects like convolution reverb, cabinet simulation, and precise frequency equalization. This article provides a quick, practical guide on how the afir filter works, its basic syntax, and real-world command examples to help you implement it in your audio processing projects.

Understanding the AFIR Filter

The afir filter works by convolving an input audio signal with an impulse response file. An impulse response is a recording of how a space (like a cathedral) or a piece of hardware (like a guitar amplifier) reacts to a single spike of sound. By combining your source audio with an IR, you can simulate that specific acoustic environment or hardware coloration.

The filter requires at least two audio inputs: 1. Input 0: The original, dry audio stream. 2. Input 1: The impulse response (IR) audio file.

Basic Syntax and Command Structure

Because the afir filter requires multiple inputs, you must use FFmpeg’s -filter_complex flag. Here is the basic command template:

ffmpeg -i input.wav -i impulse_response.wav -filter_complex "[0:a][1:a]afir[out]" -map "[out]" output.wav

In this command: * -i input.wav is your dry audio. * -i impulse_response.wav is the IR file. * [0:a][1:a] feeds the audio of the first input and the audio of the second input into the filter. * afir processes the convolution. * -map "[out]" ensures the output of the filter is saved to the final file.

Common Parameters

You can customize the behavior of the afir filter by passing key-value parameters.

• dry: Sets the gain of the dry (unprocessed) signal. Default is 1.0.
• wet: Sets the gain of the wet (processed) signal. Default is 1.0.
• gweight: Sets the global gain weight. This is useful for preventing digital clipping if the output signal becomes too loud.
• irgain: Adjusts the gain of the impulse response before processing.

Practical Examples

1. Adjusting the Wet/Dry Balance

To create a natural-sounding reverb, you usually want more of the dry signal and less of the wet signal. You can adjust this using the dry and wet parameters:

ffmpeg -i dry_vocal.mp3 -i room_ir.wav -filter_complex "[0:a][1:a]afir=dry=10:wet=0.5[out]" -map "[out]" output.mp3

2. Preventing Audio Clipping

Convolution can significantly increase the volume of your output, resulting in unpleasant distortion. You can apply the gweight parameter to scale down the overall volume:

ffmpeg -i guitar.wav -i cab_ir.wav -filter_complex "[0:a][1:a]afir=gweight=0.5[out]" -map "[out]" processed_guitar.wav

3. Working with Stereo Signals

If your input is stereo and your IR file is stereo, the afir filter will automatically map the left channel of the IR to the left channel of the input, and the right channel of the IR to the right channel of the input. Ensure both files share the same sample rate (e.g., 44100 Hz or 48000 Hz) for the best results, as FFmpeg will automatically resample the IR to match the input if they do not match.