[FFmpeg-user] FFmpeg produces m4a files that won't play on Marantz CD6003
baptiste.coudurier at gmail.com
Sun Apr 17 01:20:06 CEST 2011
On 4/16/11 12:07 PM, richard wrote:
> The Marantz CD6003 will play m4a files via a USB device or iPod
> connected to a USB port, but m4a files created using ffmpeg will not
> play. Testing has shown that the problem appears to be caused because
> the average bit rate (avgBitrate ) in the DecoderConfigDescriptor within
> the esds atom is set to zero. Where the avgBitrate is a non zero value,
> the m4a files play OK.
> I'm using Ubuntu 8.04 (Hardy). FFmpeg version git-N-28888-g6f73d5e built
> on Apr 5 2011 with gcc 4.2.4
> If I convert an mp3 to m4a using libfaac:
> ffmpeg -i test2.mp3 -acodec libfaac test2.m4a
> the m4a file won't play. Check the metadata using:
> mp4dump test2.m4a
> and the avgBitrate in the decConfigDescr in the esds atom = 0.
> If I convert the same mp3 to m4a using faac:
> ffmpeg -i test4.mp3 -f wav - | faac -o test4.m4a -b 152 -
> the m4a file plays OK, but the the avgBitrate in the decConfigDescr in
> the esds atom = 151859
> An m4a file that does not play (avgBitrate = 0) can be made playable by
> changing a tag (any tag) using EasyTag or mp4tags. When a tag is
> changed, EasyTag or mp4tags automatically changes the avgBitrate to the
> correct (or estimated) value (e.g. avgBitrate =128002).
> The problem with m4a files also occurs when I use get_iplayer. The
> latest patched version of get_iplayer does this:
> rtmpdump downloads flv
> ffmpeg remuxes flv to aac using: ffmpeg -i -vn -acodec copy -y
> ffmpeg remuxes aac to m4a and removes ADTS using: fmpeg -i -vn -acodec
> copy -absf aac_adtstoasc -y
> atomicparsley tags the m4a
> Again the resulting m4a file will not play, and the avgBitrate in the
> esds atom is zero. The m4a file can be made playable using mp4tags to
> alter a tag. This automatically changes the avgBitrate to the correct
> (or estimated) value, but the file only plays after a long delay. The
> -optimize option of mp4creator can be used to remove free atoms and the
> resulting m4a files plays OK without a long delay.
> Could this problem of setting the AvgBitrate to zero be resolved?
> ffmpeg does know the average bit rate, but it doesn't store it in the
> m4a file, and so a zero value results.
Should be fixed in latest git.
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