To breathe new life into this sort of usage, we have developed the greatest M4A to MP3 converter software, which allows you to convert your favorite video music into MP3 format and frees you to simply close your eyes (if you choose) and tune into the beat.Įven if you haven't noticed it, M4A to MP3 converter software have grown in popularity in recent years. When faced with a situation like this, all that comes to mind is a quick way to listen to the audio without exposing your eyes to the screen. Although LAME can (allegedly) go up to 640 Kbps, such MP3s wouldn't be able to be played by ordinary, low-tech devices anyway.As a music lover, you're bound to say 'music only' in your head while watching a video with catchy music. The current script just works for one directory (no globbing directory names, or the results will be unpredictable - that's a fair warning!).Ĭaveat: some M4As really have a very high bit rate, and ffmpeg complains a bit about those - after all, the MP3 specs define the maximum bitrate as 320 Kbps, with 192 Kbps (variable bit rate) being in practice the 'best' a human can hear with the kind of lossy compression provided by MP3. To-do: allow multiple directories as a command-line parameter, thus allowing a long batch to be run across the entire music library (1000s of songs!) and organise them properly according to their album names. # from M4A to MP3, respecting the bitrate, and creating a directoryĮcho "Doing conversions from directory \"$" # Takes a directory as parameter and converts everything inside it Just like I've placed the converted files inside a mp3s/ directory, but with a twist: it adds the album name as an extra subdirectory (basically, the last element of the directory fed to the script) - so everything will be grouped together again by album name :) #!/usr/bin/env bash So, copy & paste the script below, put it inside a file, add the required chmod +x file-name-chosen-for-the-script, and let it roll, by invoking it with a single parameter, which is the directory where your collection of M4A files is. There are at least a gazillion tools that do that, some even just GUI-based front ends to ffmpeg but nothing like 'being in control' and do it from the bash! This requires a slightly different (and simpler) way to parse, so I wrote my own script to do that, with a twist: in my case, I have lots of albums on my Mac, some of which are in M4A, and which require converting to MP3 for my low-end car audio player. Which should result in something like: format_bit_rate="324440" ffprobe is, at best, tricky to coerce into the proper format, but, after much googling, I sort of found a solution which does, indeed, extract the correct bit rate (at least for the first stream in the M4A - if you have many, this will probably not work): ffprobe -v quiet -of flat=s=_ -show_entries format=bit_rate The good news is that you can replace avprobe with ffprobe - after all, it comes with ffmpeg, so you don't need to add an additional package for that. 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.
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