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?



Thanks to Andy Furniss on the ffmpeg-users mailing list, I've got this one solved. One needs to explicitly tell ffmpeg not to scale the loudness:

ffmpeg -drc_scale 0 -i foo.m2ts foo.wav

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

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