[FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality

[Carl Eugen Hoyos]

Moritz Barsnick <barsnick <at> gmx.net> writes:

> $ ffmpeg -loglevel debug -i inter.dts -ac 2 -c:a libmp3lame -f null -
> $ ffmpeg -loglevel debug -i inter.dts -ac 2 -c:a pcm_s16le -f null -
> You can insert other arbitrary codecs at will.
> 
> The former shows a matrix:
> 1.000000 0.000000 0.707107 0.000000 0.707107 0.000000
> 0.000000 1.000000 0.707107 0.000000 0.000000 0.707107
> [auto-inserted resampler 0  <at>  0xb713840] ch:6 chl:5.1(side) 
> fmt:fltp r:48000Hz -> ch:2 chl:stereo
> fmt:fltp r:48000Hz
> 
> while the latter shows:
> 0.414214 0.000000 0.292893 0.000000 0.292893 0.000000
> 0.000000 0.414214 0.292893 0.000000 0.000000 0.292893
> [auto-inserted resampler 0  <at>  0xb3b55c0] ch:6 chl:5.1(side) 
> fmt:fltp r:48000Hz -> ch:2 chl:stereo
> fmt:s16 r:48000Hz
> 
> I think this may be the described behavior.

If this really is the issue, it should be reproducible 
with at least one of the command lines I proposed 
(namely for -acodec pcm_f32le) and it is possible to 
work-around the issue by forcing s16p as the mp3 
encoding format. The mp3 encoder accepts fltp, s16p 
and s32p.

But I would really appreciate if somebody can confirm 
that the issue is reproducible with pcm_f32le (and 
neither with s16le nor s32le).

Carl Eugen