converting aac to mp3 via command line

Question

Original Question

Is there some cli tool to convert aac files from the iTunes Store (no DRM) to mp3?

I installed libavcodec-unstripped-52 and get a little further.

When I try to encode MP3 files with it: It says [mp3 @ 0x997cde0]Header missing.

lame also had problems since the tags are ID3v2, is it possible that the libmp3lame does not support that either?

Should I convert the tags or what can I do to get this working?

This went through, but there are no headers in the resulting MP3 file. How can I get the headers in there?

score 42 · Accepted Answer · edited Jul 17 '15 at 08:23

42

The most appropriate command line tool is the ffmpeg utility available to be installed via the software center/synaptic manager.

A command line example would be something like this for Constant Bitrate Mode (CBR):

ffmpeg -i inputfile.m4a -c:a libmp3lame -ac 2 -b:a 190k outputfile.mp3

Or even better for Variable Bitrate Mode (VBR):

ffmpeg -i inputfile.m4a -c:a libmp3lame -ac 2 -q:a 2 outputfile.mp3

More information on mp3 encoding with FFmpeg can be seen here:

edited Jul 17 '15 at 08:23

andrew.46

answered Apr 16 '11 at 14:49

fossfreedom

1

okay, that looks very good. My ffmpeg just says "Unknown encoder 'mp3'". I have lame installed. – Martin Ueding Apr 16 '11 at 14:53
solved it by installing libavcodec-unstripped-52, but now I got the next problem. (I added it to the original question.) – Martin Ueding Apr 16 '11 at 15:15
have a look in synaptic - have you libfaad2 installed? Generally I use sudo apt-get install ubuntu-restricted-extras to ensure all relevant codecs are installed – fossfreedom Apr 16 '11 at 15:42
libfaad2 is installed, but I got by the mp3 with with libavcodec-unstripped-52 I believe. – Martin Ueding Apr 16 '11 at 15:51
2

I'm on natty - this is how I converted a downloaded aac file
sudo apt-get install libavcodec-extra-52 followed by ffmpeg -i inputfile.m4a -acodec libmp3lame -ac 2 -ab 160 outputfile.mp3
– fossfreedom Apr 16 '11 at 18:32
Maverick here, but I got it with the same line. Thanks! – Martin Ueding Apr 16 '11 at 19:43
Maybe one more comment, it should actually be -ab 160k and not -ab 160. The -ab parameter takes bits, not kilobits. – Malte Skoruppa Jul 11 '14 at 00:36

score 8 · Answer 2 · edited May 02 '15 at 20:04

8

If your Linux complain that he doesn't have an mp3 codec, try this:

ffmpeg -i inputfile.m4a -acodec libmp3lame -ac 2 -ab 160k outputfile.mp3

Note that the -ab parameters takes in bits per second, not kilobits per second.

edited May 02 '15 at 20:04

Henry

answered Sep 09 '13 at 23:39

maxmurd

1

As above, note it should actually be -ab 160k and not -ab 160. The -ab parameter takes bits, not kilobits. – Malte Skoruppa Jul 11 '14 at 00:45