There are some use cases for this. For example, you can use it to set
defaults of automatically inserted filters (like af_lavrresample). It's
also useful if you have a non-trivial VO configuration, and want to use
--vo to quickly change between the drivers without repeating the whole
configuration in the --vo argument.
See the changes in input.rst for explanations.
Technically speaking, this also gets rid of some undefined behavior:
passing NULL as a vararg (execl()) is always a bug.
This partially reverts commit 7d152965. It turns out that at least some
ALSA drivers (at least snd-hda-intel) report incorrect audio delay with
non-native sample rates, even if the sample rate is only very slightly
different from the native one.
For example, 48000Hz is fine on my hda-intel system, while both 8000Hz
and 47999Hz lead to a delay off by 40ms (according to mpv's A/V
difference display), which suggests that something in ALSA is
calculating the delay using the wrong sample rate.
As an additional problem, with ALSA resampling enabled, using
48001Hz/float/2ch fails, while 49000Hz/float/2ch or 48001Hz/s16/2ch
work. With resampling disabled, all these cases work obviously, because
our own resampler doesn't just refuse any of these formats.
Since some people want to use the ALSA resampler (because it's highly
configurable, supports multiple backends, etc.), we still allow enabling
ALSA resampling with an ao_alsa suboption.
It appears PTS sorting was useful only for avi files (and VfW-muxed
mkv). Maybe it was historically also important for decoders with broken
or non-existent PTS reordering (win32 codecs?). But now that we handle
demuxers which outputs DTS only correctly, it just seems dead weight.
Disable it by default. The --pts-association-mode option is now forced
to always use the decoder's PTS value. You can still enable the old
default (auto) or force sorting. But we will probably remove this option
entirely at some point.
Make demux_mkv export timestamps at DTS when it's in VfW mode. This is
needed to get correct timestamps with the new default mode. demux_lavf
already does that.
This was needed to determine PTS from DTS, but the previous commits
make it unnecessary.
The builtin genpts hack was used for DVD, because libavformat's genpts
essentially went amok on DVD timestamp resets. See commit 65d87091 for
details.
This drops the --pp option, which was probably broken for a while. The
option automatically inserted the "pp" filter. The value passed to it
was ignored (which is probably broken, it always selected maximal
quality).
Inserting this filter can be done simply with --vf=pp, so this is not
needed anymore.
This commit adds a new build system based on waf. configure and Makefile
are deprecated effective immediately and someday in the future they will be
removed (they are still available by running ./old-configure).
You can find how the choice for waf came to be in `DOCS/waf-buildsystem.rst`.
TL;DR: we couldn't get the same level of abstraction and customization with
other build systems we tried (CMake and autotools).
For guidance on how to build the software now, take a look at README.md
and the cross compilation guide.
CREDITS:
This is a squash of ~250 commits. Some of them are not by me, so here is the
deserved attribution:
- @wm4 contributed some Windows fixes, renamed configure to old-configure
and contributed to the bootstrap script. Also, GNU/Linux testing.
- @lachs0r contributed some Windows fixes and the bootstrap script.
- @Nikoli contributed a lot of testing and discovered many bugs.
- @CrimsonVoid contributed changes to the bootstrap script.
Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns
occur (i.e. running out of data). Add some options that control
simulated buffer and outburst sizes.
All this is useful for debugging and self-documentation. (Note that
ao_null always was supposed to simulate an ideal AO, which is the reason
why it fools people who try to use it for benchmarking video.)
This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
We found that the stretching - although it usually improves the looks of
the fonts - is incorrect.
On DVD, subtitles can cover the full area of the picture, and they have
the same pixel aspect as the movie itself.
Too bad many commercially released DVDs use bitmap fonts made with the
wrong pixel aspect (i.e. assuming 1:1) - --stretch-dvd-subs will make
these more pretty then.
This removes "--hwdec=crystalhd".
I doubt anyone even tried to use this. But even if someone wants to
use it, the decoders can still be explicitly invoked with e.g.:
--vd=lavc:h264_crystalhd
The only advantage our special code provided was fallback to
software decoding. (But I'm not sure how the ffmpeg crystalhd
pseudo-decoder actually behaves.)
Removing this will allow some simplifications as soon as we don't need
vdpau_old.c anymore.
This uses vdpau OpenGL interop to convert a vdpau surface to a texture.
Note that this is a bit weak and primitive. Deinterlacing (or any other
form of vdpau postprocessing) is not supported. vo_opengl chroma scaling
and chroma sample position are not supported. Internally, the vdpau
video surfaces are converted to a RGBA surface first, because using the
video surfaces directly is too complicated. (These surfaces are always
split into separate fields, and the vo_opengl core expects progressive
frames or frames with weaved fields.)
VA-API's OpenGL/GLX interop is pretty bad and perhaps slow (renders a
X11 pixmap into a FBO, and has to go over X11, probably involves one or
more copies), and this code serves more as an example, rather than for
serious use. On the other hand, this might be work much better than
vo_vaapi, even if slightly slower.
Most hardware decoding APIs provide some OpenGL interop. This allows
using vo_opengl, without having to read the video data back from GPU.
This requires adding a backend for each hardware decoding API. (Each
backend is an entry in gl_hwdec_vaglx[].) The backends expose video data
as a set of OpenGL textures.
Add infrastructure to support this. The next commit will add support for
VA-API.
We had some code for checking profiles earlier, which was removed in
commits 2508f38 and adfb71b. These commits mentioned that (working) hw
decoding was sometimes prevented due to profile checking, but I can't
find the samples anymore that showed this behavior. Also, I changed my
opinion, and I think checking the profiles is something that should be
done for better fallback to software decoding behavior.
The checks roughly follow VLC's vdpau profile checks, although we do
not check codec levels. (VLC's profile checks aren't necessarily
completely correct, but they're a welcome help anyway.)
Add a --vd-lavc-check-hw-profile option, which skips the profile check.
Roughly follows MPlayer svn commits 36492 and 36493. We also remove
the volume peak reporting. (There are much better libavfilter filters
for this, I think.)
af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
Now that talloc has been removed, the license can be switched back to
GPLv2+. Actually, there never was a GPLv2+ licensed MPlayer (fork or
not) until now, but removal of some GPLv2-only code makes this possible
now. Rewrite the Copyright file to explain the reasons for the licenses
MPlayer and forks use. The old Copyright file didn't contain anything
interesting anymore, and all information it contained is available at
other places in the source tree.
The reason for the license change itself is that it should improve
interoperability with differently licensed code in general.
This essentially reverts commit 1752808.
The argument or this change is that --loop should set how often the
file is played, not the number of additional repeats.
Based on pull request 277, with additions to the manpage and removal
of "--loop=0".
Signed-off-by: wm4 <wm4@nowhere>
