Sadly, avprobe comes with the libav library, which is considered abandonware, since its last stable release was in February 2018 (there have been a few updates in early 2019 on their git). His provided script does exactly that, relying on avprobe to extract the bit rate from the M4A before doing the actual conversion to MP3. PS: I am working on an intel Mac, but also have a Ubuntu answer above is (at the time of writing this) the only one that truly addresses the OP's main issue, which is preserving the bit rate in the original M4A - neither more nor less - on the resulting MP3. If this isn't possible, is there some sort of script which might detect the required quality as it converts files individually? (Of course there is likely to be some extra losses from converting from lossy to lossy file formats, above that which would be expected when converting from a lossless to lossy format.) (At the moment, the files are being converted to about 50k.)ĭoes anyone know how I can do this? What I really want to do is tell ffmpeg to convert all m4a files in a directory into mp3's while retaining the current audio quality as best it can.

Similarly it makes no sense to destroy all my 320k files by converting them to something much lower than 96k. It seems to make no sense to force 320k, since some files will become many times larger than they need be. Similarly I don't want to convert using a constant bitrate, such as 320k, because some of the files I am converting are 320k m4a's and some are as low quality as 96k m4a's. I tried doing something simple like: ffmpeg -i FILE.m4a FILE.mp3 but this seems to reduce the bitrate to a very low value, which isn't what I want.

I have a load of audio files (about 1000) which I want to convert from m4a to mp3 so I can use play them on a CD player which has a USB port.

M4a to mp3