[FFmpeg-devel] [RFC] Audio normalization

[Jan Ehrhardt]

Clément Bœsch in gmane.comp.video.ffmpeg.devel (Fri, 22 Feb 2013
00:22:38 +0100):
>How to test it:
>
>    ./ffplay -f lavfi -i
>	'amovie=in.mp3,ebur128=video=1:metadata=1[r128-before][a];
>	[a]volume=metadata=lavfi.r128.I,ebur128=video=1[r128-after][out1];
>	[r128-before] pad=iw*2 [padded]; [padded][r128-after] overlay=w'

This one exits immediately in my cross-compiled ffplay.exe, bit that may
be a problem with the cross-compiling.

>Also, there is bug: it seems ffmpeg and ffprobe don't appreciate very much the
>segmentation through {min,max}_samples in lavfi, and exit very quickly without
>giving much info (AFAICT it seems the fifo is empty after a first request
>frame). How to reproduce: after the patchset,
>    ffmpeg -f lavfi -i 'amovie=in.mp3,ebur128=metadata=1' -f null -

I am now experimenting with ffmpeg & things like

-filter_complex
	"[0:v]setpts=PTS-STARTPTS[v0]; \
	[0:a]asetpts=PTS-STARTPTS,ebur128=metadata=1[a0]; \
	[a0]volume=metadata=lavfi.r128.I,ebur128[a1]" \
	-map [v0] -map [a1]

This produces a correct output in ffmpeg, but alas without volume
correction. But my intuition says something like this should be
possible. Would that be an idea?

Jan