Use the value specified in the codec context for setting the
filterchain sample aspect ratio, when it is not specified in the
stream context.
Consistent with the ffmpeg behavior.
Fix trac issue #398.
avcodec_set_dimensions should be used for size changes to ensure
compatibility with future changes.
avctx->width/avctx->height may not be set to display-only dimensions.
Even more so since vc1dec.c would later set coded_width/height based
on this.
coded_width/coded_height should be used instead of width/height for
decoder setup.
This fixes playback of e.g. zz-mcr-nsqr.vc1 sample (containing
display width/height settings) in MPlayer and should fix a crash
with MPC: http://forum.doom9.org/showthread.php?t=162221.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Its only reason for its existence was ffmpeg's inability to properly assign
AVOptions to streams. Now this is not a problem anymore, so 'ab' should
go.
The new option doesn't depend on its placement wrt -new* options (which
don't exist anymore) and works in a similar way as per-stream AVOptions.
-[vas]codec remain as aliases to -codec:[vas]
* qatar/master:
Fix NASM include directive
dsputil_mmx: Honor HAVE_AMD3DNOW
lavf,lavd: remove all usage of AVFormatParameters from demuxers.
jack: add 'channels' private option.
VC-1: fix reading of custom PAR.
Remove redundant and dubious video codec detection by its extradata
mpeg12: remove repeat-field code disabled since May 2002
patch checklist: suggest fate instead of regression tests
Turn on resampling on sudden size change instead of bailing out during recode.
avtools: reinitialise filter chain when input video stream changes dimensions
Conflicts:
Makefile
avconv.c
doc/developer.texi
ffplay.c
libavcodec/x86/dsputil_mmx.c
libavdevice/libdc1394.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AVFormatParameters are converted into corresponding private options in
av_open_input_file/stream() compat wrappers, so accessing them from
demuxers is redundant.
Since SDL has no audio buffer fullness info, one can get a much precise audio
clock based on the last time of the audio callback and the elapsed time since.
To achieve this I introduced the audio_current_pts and audio_current_pts_drift
variables (similar to video_current_pts and video_current_pts_drift) and
calculate them in the end of the audio callback, when VideoState->audio_clock
is already updated. The reference time I use is from the start of the audio
callback, because this way the amount of time used for audio decoding is not
interfereing with calculation.
I also replaced the audio_write_get_buf_size function with a calculated
variable because when the audio frame decoding is in progress audio_buf_size
and audio_buf_index are not stable, so using them from other threads are not a
good idea.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aacenc: Fix number of coefficients used in a LFE channel.
aacenc: Fix a segfault with grouped psymodel.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In apply_unsharp(), when y is >= height, prevent out-of-buffer reading
from src, read from the last buffer line in src2 instead.
The check was implemented in the original unsharp libmpcodecs code and
lost in the port.
This also fixes output discrepancy between the two filters.
Stefano Sabatini
Michael Niedermayer
Carl Eugen Hoyos
Reimar Döffinger
Anton Khirnov
Dustin BrodyandAnton Khirnov
Dave YeoandRonald S. Bultje
Alex Converse
Kostya ShishkovandDiego Biurrun
Marton BalintandMichael Niedermayer
Baptiste CoudurierandMichael Niedermayer