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Old 01-24-2009, 10:36 PM   #1
Registered: Feb 2005
Location: Calgary, AB Canada
Distribution: Kubuntu 6.10
Posts: 44

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faac issues creating m4b book files for iPod

I have an mp3 eBook I own on CD. I want to create an m4b file so I can use it on my iPod (classic 120Gb) and use all the audiobook features. I found a script on the net that I am trying to use to create an m4b file from all the mp3 files (34 of them) on the CD. I copied them all to my system (Kubuntu 8.10) and used the following script:

mp3wrap file_name *.mp3
mplayer -vc null -vo null -ao
faac -R 44100 -B 32 -C 2 -X -w -q 100 --artist "author"
--album "title" --title "title" --track "1" --genre
"Spoken Word" --year "year" -o file_name.m4b

The output of mp3wrap plays great in Amarok. The output of mplayer also plays great in Amarok. The output of faac sounds like Alvin and the Chipmunks recorded it. Even the audiobook slow setting on my iPod sounds like Alvin et. al. All the things like title and author are filled with the correct information. I'm finding almost nothing on faac with a Google search. I tried varying the -B 99 parameter, but never got a good result. Any help would be appreciated? TIA


The -R 44100 setting must be -R 22050 to work.

Last edited by Calgarian; 01-25-2009 at 01:06 PM. Reason: SOLVED
Old 01-26-2009, 02:09 AM   #2
David the H.
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Change the faac command to "-B 16" (or just remove it entirely, 16 is the default). Your standard CD quality pcm audio is 16 bits, not 32 bits.

BTW, if this is just spoken-word, you can probably also reduce the quality setting as well by half or more, and perhaps even change it to mono as well. You don't need that much of a bitrange if you aren't encoding music.

Last edited by David the H.; 01-26-2009 at 02:14 AM.
Old 01-26-2009, 08:41 AM   #3
Registered: Feb 2005
Location: Calgary, AB Canada
Distribution: Kubuntu 6.10
Posts: 44

Original Poster
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Thanks for the reply. I did change the setting to -B 16 and forgot to edit that in my message. The -B 16 and the -R 22050 did the trick. I really don't understand the impact of the changes as my knowledge of audio is very, very weak. However, it worked and I got to it by just playing around some.
Old 01-27-2009, 08:45 AM   #4
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Registered: Jan 2009
Location: Germany, near Frankfurt
Distribution: FreeBSD 7.1
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An alternative with lame in one single step (and the guarantee that all input files have the same samplerate):

for f in *.mp3; do lame --resample 22.05 -m j "$f" - | \
    lame --decode -t --mp3input - -; done | \
  faac -B 16 -R 22050 -C 2 -q 100 -b 64 -X -w \
    -o output.m4b [more options like --artist etc.] -
Could not make lame to resample and decode to pcm in one step so I had to use a pipe for this. Don't forget the several hyphens for input/output

The resample-rate (22.05 khz) and the stereo/mono-mode ("-m j" or "-m m") of lame must be the same for faac ("-R 22050" and "-C 2"/"-C 1").


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