* commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7':
Use modern avconv syntax for codec selection in documentation and tests
Merged-by: James Almer <jamrial@gmail.com>
* commit 'a0797950527120c85263c910eb6ba08fddcfdcb3':
fate/mp3: specify the number of output samples instead of filesize
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '6ec688e1bc76dd93151cbca1c340162ae4b10d77':
mp3: enable packed main_data decoding in MP4
Conflicts:
libavcodec/mpegaudiodec_template.c
Only the parts needed to support the available sample are merged
the remaining error checks are left in place
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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: Luca Barbato <lu_zero@gentoo.org>
This makes only tests actually using avconv depend on it.
The remaining tests already depend on what they need.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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
