Granted if anyone see's a way in which this may be problematic, please tell me, and I make no guarantee's, but here is what I came up with (this recursively converts all the m4a alac files in the current directory): I dug around for a couple hours, and found out about ffprobe, and at first I ignored it, because I knew the command I wrote would end up being long, but with no other choice, I present to you my final, working, powershell command.
Is this a limitation of flac? And if not, then how do I instruct ffmpeg to output at the original bitrate?Īlso, if it affects the quality, then what are s16 and s16p referring to?ĮDIT: 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. Now, each track has a slightly different bitrate, but they all end up outputting to 128.
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 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:Īudio: alac (alac / 0圆3616C61), 44100 Hz, stereo, s16p, 968 kb/s (default)Īnd the new flac version shows this for the audio stream:Īudio: flac, 44100 Hz, stereo, s16, 128 kb/s (default) Anyway, I'm converting using this command: I'm trying to convert my imported music collection to flac for compatibility reasons.
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