Brought in from the thread on my idea for the version markers script and as a sound engineer very passionate about the Loudness topic, I totally support this idea.
Envato has already some of the ingredients of the recipe in place. Since it needs to analyze the incoming audio files to create the waveform displaying in the player – wherever that analysis is performed during the uploading process –, applying the LUFS normalization after that analysis is completed would be just a matter of adding one single line of code to the underlying engine putting together the material before exposing it to the public.
From a programming point of view, a FFMPEG command like the following one would be enough to programmatically normalize any audio file to the desired LUFS:
ffmpeg -i input_file.mp3 -af loudnorm=I=-13:TP=-0.3:LRA=11 output_file.mp3
which manipulates input_file.mp3 to –13 LUFS, –0.3 dB True Peak and an 11 LU dynamic range and saves it as output_file.mp3.
Of course, through to the previous analysis that generated the waveform for the player, FFMPEG could also calculate the current LUFS of input_file.mp3, leaving unaltered all those files that have already a loudness lower than the target one.
The technical implementation would be very easy and, provided the fact that the waiting time for a track to be approved is already several days, this is definitely not a real-time process, so its implementation is far from being critical for the Envato’s server resources.
I have just run both the analysis and the normalization on a 4-minute track using a MacBook Pro:
– the analysis was completed in about 10 seconds
– the normalization was done in about 18 seconds
This would be a piece-of-cake of a task for Envato’s servers!