>> I currently use MPD (https://www.musicpd.org/) to play music. MPD can use
>> ffmpeg to decode the audio coming in (local or cloud). Since ffmpeg can
>> handle convolution files in wav format, I was looking for a way to "pass"
>> these files to ffmpeg. I would like to implement the following scenario:
>> 1. Have one convolution file per sampling rate (44.1kHz, 48kHz, 88,2kHz,
>> 96kHz, 176.4kHz and 192kHz)2. Store the convolution files locally to
>> ffmpeg3. When ffmpeg is used, it will use the convolution file that matches
>> with the incoming file's sampling rate
>> I want this to work as a global setting without the need to do it manually
>> for every audio file. I know that minimserver has implemented this but it is
>> a closed source project.
>> Any ideas?
> You will need to use FFmpeg audio convolution filter, afir.
> And write some kind of script for processing of all files.

What is a convolution file? When you get a venue with a weird frequency 
response and you come in before the sound check to record spectrum sweeps from 
the speakers with a figure 8 mic, the tool had a convolution function that 
would generate a model of the venue that served as a general starting point for 
mixing, is it something similar?

Ted Park

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