Issue was caused by the change in FFmpeg options: some of them were
renamed, some moved to another class.
Made some tweaks to how options are passed to the FFmpeg which now
seems to be the same as ffmpeg.c.
Issue was caused by missing GLOBALHEADER flag set for audio stream.
Also made it so frame is getting filled with default, that's what
happening in ffmpeg.c.
CODEC_ID_* have been replaced with AV_CODEC_ID_* in new libavcodec
versions.
Update the code to use those new identifiers.
Added a compatibility code to ffmpeg_compat.h
Issue was caused by wrong PTS calculation. This commit
makes this calculation closer to what's happening in
FFmpeg itself.
Seems everything is working now including newer FFmpeg,
but there's one thing which still doesn't work: writing
avi files with h264 codec and Vorbis audio doesn't play
correct in mplayer here. But didn't manage to get this
working even using FFmpeg CLI, so this might be just a
bug in FFmpeg/mplayer. Since this file works fine in
blender just fine wouldn't consider this is crucial thing
to look into at this moment.
* Removed audio-only options from ffmpeg render settings (added some versionning code too)!
* Moved the Mixdon button from the Scene->Audio pannel to the Render->Render panel.
CHANGE: writing an animation in the ogg movie format now defaults to .ogv (ogg/video) which is recommended by Xiph.org for video
CHANGE: for .ogg files a check is added whether Blender can read it as a movie (is avi or ffmpeg movie), otherwise assume audio
CHANGE: the anim player now filters for the same extensions as the file browser
After planar codecs support minimal FFmpeg was
bumped to 0.10 which was not so much nice because
it was only released only later last year.
Didn't find a way to make compatibility code local
in ffmpeg_compat, so there're some ifdefs in
audaspace and writeffmpeg.
Not entirely happy, but having a bit of ifdefs in
code better than lots of real PITA for platform
maintainers.
Patch by Jehan Pages (pardon for mis-typing, emacs-nox works not so good with
urf-8 buffers here), with some own modifications. Thanks!
From the patch tracker:
The problem is because of several versions of ffmpeg, but even more because of
the fork situation libav/ffmpeg. So there are some installed versions out there
where you *must* use a float sample for some codec; whereas oppositely on some
other installations, you *must* use the int sample. So for some people, one
works not the other, and reciprocally.
As a consequence, you can't just have a switch codec-based, like in current
code, which decides on the float or int implementation, you must necessarily
have a runtime test because you won't know until then if ogg vorbis will use
one or another sample (note: that's true also for AC3 as I fixed the exact same
bug in DVDStyler for AC3 encoding a few months ago; and I guess it would be same
for AAC).
Some notes from self:
- New FFmpeg requires using FLTP for AAC, AC3 and Vorbis, it's not supported
by audaspace and result in this case would be just wrong. Throw an error
in cases FLTP is trying to be used.
- Moved strict_std_compliance a bit upper. When we'll support FLTP both
FLT and FLTP for AAC would need to be using FF_COMPLIANCE_EXPERIMENTAL.
- It is nice to have such check of supported by codec formats anyway.
Dudes, please try building blender with all default features
before doing such a commits. It helps just a lot when bisecting
issues later.
Also solved const qualifier discard happens in recent monofont
commit.
Patch makes it possible to compile blender with recent ffmpeg
and libav libraries, mainly by getting rid of deprecated API.
Original patch by Campbell Barton with own modifications to
support compilation with older ffmpeg versions.
This patch could break compatibility of FFV1 videos playing
back in older players, mainly because of alpha support changes.
Preserving compatibility with such players became a headache
and think it's high time to get rid of workarounds here.
Issue was caused by rc_initial_buffer_occupancy being set for context
Commented this option for now -- it's not used in ffmpeg.c.
Now encoding seems to be working nice for MPEG4 and other codecs as well.
However, if there're some gurus around please check :)
Patch by David M (erwin94), thanks!
Also made Vorbis codec using float sample_fmt, otherwise it didn't work
with new FFmpeg.
Perhaps we can make it more clear by explicitly separating audio_input_buffer
for float and integer buffers, but as far as it works i'm not so fussed about
this atm.
