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I tried to convert a file with pacpl, but I get the well-known "256" error. With the -v flag, the FAQ of pacpl tells me:

"The file you are trying to convert is a lossless .m4a file. The format is not yet supported by FAAC/FAAD."

Since faac/faad seems to be used in every other converting tool on Ubuntu, how can I successfully convert formats?

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How is the lossless m4a a 24 bit sample? CD's are 16 bit, and SACD's are not rippable by any means, which means it was pointlessly "upconverted" to 24 bit, which is stupid. – user126919 Jan 29 '13 at 3:25
up vote 10 down vote accepted

You can convert an m4a file to flac with the ffmpeg command-line tool:

To install ffmpeg:

sudo apt-get install ffmpeg

To convert:

ffmpeg -i filein.m4a -f flac fileout.flac

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works for me:) Hopefully the pacpl devs will fix the problem – Graslandpinguin Jan 15 '12 at 22:38
`` for file in *.m4a; do echo $file; ffmpeg -i "$file" -f flac "basename "$file" .m4a.flac"; done `` To do a batch conversion of all *.m4a files in directory. – zetdotpi Jun 13 '13 at 0:04
sudo aptitude install libav-tools

for file in *.m4a; do avconv -i "$file" -f flac "`basename "$file" .m4a`.flac"; done
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While both the answers involving ffmpeg/avconv (which I think are essentially the same tool) both work, they currently have a flaw. Namely that lossless m4a is often 24 bit sample, and currently ffmpeg/avconv will generally force the conversion to end up in 16 bit sample.

I believe using sndfile-convert (libsndfile) does not have this problem. Likewise, I believe it can be avoided by using mplayer to decode the m4a before encoding it with ffmpeg or flac. I think soundKonverter on KDE may do this for you.

In any case, whatever you do, I suggest checking whether the original and the converted file have the same bit depth of samples.

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But sndfile-convert can not read iTunes AAC-LC. – Martin Schröder Nov 30 '12 at 18:49

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