Opus is lossy, so no. But theyβre already stored in a lossy format, so you probably wonβt notice a difference if you use the same bitrate.
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Unless the m4a files are ALAC-encoded, you shouldn't do it. m4a files tend to hold AAC-encoded audio, which is a lossy format. Opus is also a lossy format, so you'll only lose sound quality, basically. You should only convert lossless to lossy.
You can use the find
command to do the stuff other commenters have posted recursive
or fd
What platform do you have to work with?
linux mint
Disregard. The answer is 'no'. Opus is a lossy format, so there is no way to losslessly convert to it, regardless of the tools available to you.
No as both are lossy codecs you will always lose quality doing so. You can do it with ffmpeg.
parallel ffmpeg -i {} {.}.opus ::: *.m4a
You may want to tweak parameters for bitrate, etc. But this is not needed at all, as AAC had patents that are now expired.
If you get .flac, .wav or similar lossless music, you should encode that with opus. But lossy to lossy makes nearly no sense (apart from specific players not supporting them)
GNU Parallel works well for this kind of thing. A lot of audio stuff is single threaded, so unlike video transcoding running multiple conversions simultaneously is a useful thing to do. The command is simpler, too:
parallel ffmpeg -i {} {.}.opus ::: *.m4a
Thanks, learning is always good, changed it
If you get .flac, .wav or similar lossless music, you ~~should~~ could encode that with opus.
Fixed.
Unless you have a strong stance against people storing lossless files of their music? But I don't think that's quite what you meant :)
it's my music and i dont like proprietary formats
"If you get .flac, .wav or similar lossless music, you should encode that with opus."
I am looking forward to similarly useful discussions XD
If you're on linux try navigating to the folder in the terminal and running
for i in *.m4a; do ffmpeg -i "$i" "${i%.*}.opus"; done
before doing anything like that.. back up the folder.
Well sure always back up everything but this command will create new .opus files still leaving the original .m4a in the folder too, so even if it errors just delete the bad .opus files and try to resolve your ffmpeg codec issue before retrying the conversion or trying another method. Ffmpeg in my experience always converts with the "makes a new file with your file(s)" method, I've never had it do the "change the file destructively" method.
It'll also error out or prompt to overwrite an existing file unless a flag is passed that tells it to overwrite unattended.
Sure, but if it errors out no harm no foul, and don't pass any flags to overwrite the input file which I had not included above.
While I'm thinking about it, what is that flag? Because afaik ffmpeg can't overwrite and convert the file simultaneously, you have to use a temp file.
That's the point: you have to go out of your way to accidentally overwrite your input files with ffmpeg.
And no it indeed can't output to the same file as input.