[FFmpeg-user] Erroneous volume level shifts using ffmpeg to convert AC3 in M2TS to WAV

[Andy Furniss]

John Pilgrim wrote:
> I using ffmpeg to extract audio from non-commercial bluray m2ts video
> files, for subsequent loudness analysis in AudioLeak or Dolby Media
> Meter. The m2ts sometimes have AC3 audio and sometimes have linear
> PCM audio. I wish to output a WAV file. The command I am using is
>
> ffmpeg -i foo.m2ts foo.wav The volume levels of the WAV files are not
> accurate compared to the corresponding AC3 files.
>
> Here's what I mean: If I extract the ac3 audio instead, using
>
> ffmpeg -i foo.m2ts -acodec copy -f ac3 foo.ac3 and run both through
> Dolby Media Meter, the loudness and peak levels of the WAV are
> shifted up/or down compared to the corresponding AC3 file.
>
> And this is happening with AC3 files with a DIALNORM metadata value
> of -31, which per the Dolby specs, should result in NO volume level
> change, as -31 is the reference point.
>
> Does anyone have any insight into what's going on, or recommendations
> for a better invocation of ffmpeg?

By default ac3dec will apply full DRC if there is any there it doesn't
use dialnorm by default.

To get full range DRC wise do

ffmpeg -drc_scale 0 -i foo.m2ts

WRT dialnorm - I am not familiar with Dolby s/w, but I think it's
possible that some decoders would scale up to a target higher than -31
depending on how they are set - so just because the stream is -31 I am
not sure that excludes the possibility that a decoder will adjust -
target level is a decoder setting.