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Original Question

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

Update 1

MP3 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?

AAC to MP3

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

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2 Answers 2

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40

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:

FFmpeg MP3 Encoding Guide

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  • 1
    okay, that looks very good. My ffmpeg just says "Unknown encoder 'mp3'". I have lame installed.
    – Martin Ueding
    Apr 16, 2011 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, 2011 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, 2011 at 15:42
  • libfaad2 is installed, but I got by the mp3 with with libavcodec-unstripped-52 I believe.
    – Martin Ueding
    Apr 16, 2011 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, 2011 at 18:32
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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.

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    As above, note it should actually be -ab 160k and not -ab 160. The -ab parameter takes bits, not kilobits.
    – Malte Skoruppa
    Jul 11, 2014 at 0:45

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