No as both are lossy codecs you will always lose quality doing so. You can do it with ffmpeg.
for f in *.m4a; do ffmpeg -i "$f""${f%.m4a}.opus"; done
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:
No as both are lossy codecs you will always lose quality doing so. You can do it with ffmpeg.
for f in *.m4a; do ffmpeg -i "$f" "${f%.m4a}.opus"; doneYou 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)
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 :)
“If you get .flac, .wav or similar lossless music, you should encode that with opus.”
I am looking forward to similarly useful discussions XD
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:
Thanks, learning is always good, changed it