This is not dependent on the output format.

14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.

However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.

In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.

Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: NLuca Barbato <lu_zero@gentoo.org>

This allows having the samples accessible via different paths
on the target and on the host.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMans Rullgard <mans@mansr.com>

This allows fate to run without errors with or without SAMPLES
being set.
Signed-off-by: NMans Rullgard <mans@mansr.com>

This makes only tests actually using avconv depend on it.
The remaining tests already depend on what they need.
Signed-off-by: NMans Rullgard <mans@mansr.com>

related to b7165426
Error messages and audible artifacts were fixed in that commit.

instead of current "make output file of size less than ss".

Also use it to make MP3 tests more readable (using -fs xxx where xxx is
the requested output size, not something slightly lower).

Originally committed as revision 24884 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 24881 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 24880 to svn://svn.ffmpeg.org/ffmpeg/trunk