On Sat, May 16, 2015 at 10:27:07 +0000, Carl Eugen Hoyos wrote: > Andy Furniss <adf.lists <at> gmail.com> writes: > > > IIRC this has come up before. The issue seems > > to be that sometimes -ac 2 normalises and > > sometimes it doesn't (depending on what codec > > is used). > > > > You can see whether or not the matrix is > > normalised with -loglevel debug. > > I was under the impression that the output option > "-ac 2" always normalizes the input. > Do you have an example command line that shows > the opposite?
Without listening to the audio content, just taking Andy's words and ffmpeg log output: I can confirm by comparing $ 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 @ 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 @ 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. (Unfortunately, my math tells me that the former matrix is bound to cause overflows with value-restricted numerical formats. There's something I don't seem to understand there. And is LFE really ignored when downmixing?) Moritz _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
