1
I'm trying to convert my imported music collection to flac for compatibility reasons. Anyway, I'm converting using this command:
ffmpeg -i file.m4a file.flac
Now technically, this works fine, and converts the file to flac, but studying the output I notice that the original file (in this one example, the actual number varies between tracks) shows this for the audio stream:
Audio: alac (alac / 0x63616C61), 44100 Hz, stereo, s16p, 968 kb/s (default)
and the new flac version shows this for the audio stream:
Audio: flac, 44100 Hz, stereo, s16, 128 kb/s (default)
I notice two things: the first is that "s16p" was changed to "s16" though I don't know what that means, and don't care too much, but what bothers me, is the massive lowering of the bitrate! From 968 down to 128. Now, each track has a slightly different bitrate, but they all end up outputting to 128. Is this a limitation of flac? And if not, then how do I instruct ffmpeg to output at the original bitrate?
Also, if it affects the quality, then what are s16 and s16p referring to?
EDIT: I technically got what I was looking for, though I still don't know what s16 means, and I'm not 100% sure that what I did worked properly, check my answer for an explanation.
Thank you for informing me the about the output quirk. In the answer I provided, do you see any problems arising? Essentially what I'm doing is
ffmpeg -i song.m4a -ab 968k song.flac
. Which makes the output show what I want, but are there any unintended side effects from running ffmpeg like this? – Matthew – 2019-07-07T21:20:00.927The bitrate specification has no effect. The FLAC encoder does not look at it, so it's unnecessary. – Gyan – 2019-07-08T04:39:17.577