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 _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
