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Old 11-13-2005, 09:14 PM   #1
pi314159
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Registered: Nov 2003
Location: Ohio
Distribution: Ubuntu, Slack, Athena Linux (derivative of RedHat)
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Angry ffmpeg complains about codec


I'm trying to convert a batch of .m4a files to .mp3, and from what I see in the documentation (and the example here ), it looks like ffmpeg should be able to do it. Unfortunately, this happens:
Code:
pi314159$ ffmpeg -i Flood\ -\ They\ Might\ Be\ Giants\ -\ Twisting.m4a -target mp3 twisting.mp3

ffmpeg version CVS, build 3276800, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --build i486-linux-gnu --enable-gpl --enable-pp --enable-zlib --enable-vorbis --enable-libogg --enable-theora --enable-a52 --enable-dts --enable-dc1394 --enable-libgsm --disable-debug --prefix=/usr
  built on Sep 29 2005 03:25:16, gcc: 4.0.2 20050808 (prerelease) (Ubuntu 4.0.1-4ubuntu8)
Input #0, mov,mp4,m4a,3gp,3g2, from 'Flood - They Might Be Giants - Twisting.m4a':
  Duration: 00:01:56.8, start: 0.000000, bitrate: 127 kb/s
  Stream #0.0: Audio: mp4a / 0x6134706D, 44100 Hz, stereo
Assuming PAL for target.
Unknown target: mp3
Or, when I try to copy the example from the site I linked above:

Code:
pi314159$ ffmpeg -i Flood\ -\ They\ Might\ Be\ Giants\ -\ Twisting.m4a -acodec mp3 -ac 2 -ab 128 twisting.mp3

ffmpeg version CVS, build 3276800, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --build i486-linux-gnu --enable-gpl --enable-pp --enable-zlib --enable-vorbis --enable-libogg --enable-theora --enable-a52 --enable-dts --enable-dc1394 --enable-libgsm --disable-debug --prefix=/usr
  built on Sep 29 2005 03:25:16, gcc: 4.0.2 20050808 (prerelease) (Ubuntu 4.0.1-4ubuntu8)
Input #0, mov,mp4,m4a,3gp,3g2, from 'Flood - They Might Be Giants - Twisting.m4a':
  Duration: 00:01:56.8, start: 0.000000, bitrate: 127 kb/s
  Stream #0.0: Audio: mp4a / 0x6134706D, 44100 Hz, stereo
Output #0, mp2, to 'twisting.mp3':
  Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, 128 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0
This is making me think that ffmpeg isn't understanding the mp3 format, but how can that possibly be? Even if it's a problem with the m4a format, I should be okay according to this , since I have libfaad installed.

For the record, these files I'm trying to convert are ones I ripped from CD in iTunes, so they're in the annoying AAC/m4a format, if that makes any difference.

Ideas, anyone?

pi
 
Old 11-14-2005, 03:38 PM   #2
keefaz
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Registered: Mar 2004
Distribution: Slackware
Posts: 6,552

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You could convert your file to wav, then to mp3

Code:
faad -w sound.m4a > sound.wav
lame --preset standard sound.wav sound.mp3
As a note, there is a setting in iTunes preferences to use mp3 encoding rather than AAC

Last edited by keefaz; 11-14-2005 at 03:40 PM.
 
  


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