There were occasional shader compilation and rendering failures if FBOs
were unavailable. This is caused by the FBO caching code getting active,
even though FBOs are unavailable (i.e. dumb-mode).
Boken by commit 97fc4f.
Fixes#2432.
Fixes linker failure. How did this ever work? Apparently it did most of
the time, but apparently we just got the first case where it didn't.
Fixes#2433.
Bump it to 80, and 2 vsyncs. This is another measure against vsync
jitter. Admittedly this is a bit simplistic (and we should probably
estimate a stable estimated vsync phase instead), but for now this will
do.
It's not needed, because the additional data is not appended, but is the
total size of the audio buffer. The maximum size is the static audio
drop size (or twice, if the audio is duplicated).
Calculate the A/V difference directly in the display sync code, instead
of the awkward current way, which reuses the fields for audio sync.
We still set time_frame, because it makes falling back to audio sync
somewhat smoother.
When dropping or repeating frames, we essentially influence when the
frame after the next frame will be shown, not the next frame. This led
to dropping/repeating frames 2 times, because the A/V difference had a
delay of one frame. Compensate it with the expected value.
av_free_packet() got finally deprecated. Use av_packet_unref() instead,
which has almost the same semantics, has existed for a while, and is
available in all FFmpeg and Libav versions we support.
This was not very reliable.
In the normal vo_opengl case, this didn't deal well enough with vsync
jitter. Vsync timings can jitter quite extremely, up to a whole vsync
duration, in which case the "missed" frame counter keeps growing, even
though nothing is wrong. This behavior also messes up the A/V difference
calculation, but as long as it's within tolerance, it won't provoke
extra frame dropping/repeating. Real misses are harder to detect, and I
might add such detection later.
In the vo_opengl_cb case, this was additionally broken due to the
asynchronity between renderer and VO threads.
This is all kinds of stupid - update_avsync_after_frame() will multiply
this value with the speed at a later point, and we only update this
field for this function. (This should be refactored.)
Until recently, the channel layout code happened to catch this, but now
an explicit check is needed. Otherwise, it'd try to pad the missing
channels with NA in the channel map fallback code.
This is intended for the case when CoreAudio returns only unknown
channel layouts, or no channel layout matches the number of channels the
CoreAudio device forces. Assume that outputting stereo or mono to the
first channels is safe, and that it's better than outputting nothing.
It's notable that XBMC/kodi falls back to a static channel layout in
this case. For some messed up reason, the layout it uses happens to
match with the channel order in ALSA's/mpv's "7.1(alsa)" layout.
Share some code between ca_init_chmap() and ca_get_active_chmap(), which
also makes it look slightly nicer. No functional changes, other than the
additional log message.
If no channel layouts were determined (which can actually happen with
some "strange" devices), the selection code was falling back to mono,
because mono is always added as a fallback. This doesn't seem quite
right.
Allow a fallback to stereo too, if no channel layout could be retrieved
at all. So we always assume that mono and stereo work, if no other
layouts are available.
(I still don't know what the CoreAudio stereo layout is supposed to do.
It could be used to swap left and right channels. It could also be used
to pad/move the channels, but I have never seen that. And it can be set
to non-stereo channels, which breaks mpv. Whatever.)
The main reason is that ao_coreaudio_exclusive needs this for some OSX
devices. They want packed audio, and special-casing this in the
coreaudio code would be too much of a pain.
The maximum of channels we can support is 64 (because FFmpeg uses 64 bit
masks for channel layouts), but since struct mp_audio can get pretty
big (has static allocations of 2 pointers for each channel for planar
mode), it's less wasteful to stay lower for now.
av_get_default_channel_layout() fails with channel counts larger than 8.
The channel layout doesn't need to make sense, so pick an arbitrary
fallback.
libswresample also has options for setting the channel counts directly,
but better not introduce new concepts in the code. Also, libavresample
doesn't have these options.
Change it so that it will always return a bitmask with the correct
number of channels set if an unknown channel map is passed. This didn't
work for channel counts larger than 8, as there are not any standard
channel layouts defined with more than 8 channels (both in mpv and
FFmpeg). Instead, it returned 0.
This will help when raising the maximum allowed channel count in mpv.
Some code in af_lavrresample relies on it, more or less.
One change is that unknown channel maps won't result in lavc standard
channel layouts anymore, just a set of random speakers. This should be
fine, as the caller of mp_chmap_to_lavc_unchecked() should handle these
cases. For mp_chmap_reorder_to_lavc() this is not so clear anymore, but
should also be ok.
For normal channel maps, simply dropping NA channels is still the
correct and wanted behavior.
Currently, the mpv maximum channel count is 8. This commit is
preparation for raising this limit.
mNumberChannelDescriptions being 0 is pretty much an error, but if it
can happen, then the code checking the chmap below will trigger UB, as
chmap is not initialized at all.
Also, simplify the code a little: we never change the number of
channels, so this is just fine.
This code removes filters which can not take spdif inout. This was made
so that PCM filters are transparently dropped in spdif mode.
This entered an endless loop with:
--af=lavcac3enc:::2 --audio-channels=5.1
The forced number of output channels is incompatible with spdif. It's
trying to insert af_lavrresample as conversion filter to compensate for
it. Of course this doesn't work, which triggers the PCM filter removal.
Then it goes on normally - since the new state is exactly as before, it
will try the same thing again, forever.
Fix by reusing the retry counter, which is a very dumb but very
effective measure against these cases of filter negotiation failure. We
could try to be more clever (for example, if the removed filter is a
conversion filter, we can be sure this won't work, and error out
immediately). But better keep it simple and robust.
This speeds up redraws considerably (improving eg. <60 Hz material on a 60 Hz
monitor with display-sync active, or redraws while paused), but slightly
slows down the worst case (eg. video FPS = display FPS).
The IME is not useful for key-bindings. Handle the base ASCII chars
instead and don't show the IME window. For the sake of libmpv users, the
IME should only be disabled on mpv's GUI thread and not application-
wide.
No IME on the GUI thread should also mean that VK_PROCESSKEY will never
have to be handled, so the logic for that can be removed as well.
Older systems have certain EGL extension definitions missing. We
redefine them to make the build system easier, and because it's trivial.
But we forgot to define the EGL_LINUX_DMA_BUF_EXT identifier. (I hope
it's the only missing one.)
This makes the bitrate properties unavailable, instead of
returning 0 when:
1. No track is selected, or
2. Not enough packets have been read to have a bitrate estimate yet
Some mkv files can have this. The chapter times are still timestamps
(and thus not affected by the start time), but it misplaces the OSD
chapter ticks.
The equalizer code as it exists in vo_opengl works perfectly fine. The
situation in vo_opengl_cb is pretty different. The playback thread can't
communicate with the renderer thread synchronously (essentially to give
the API user more flexibility). So the equalizer communication has to be
done in an asynchronous way too.
There were two problems. First, the eq capabilities can change with the
pixel format, and the renderer initializes them on config only. This
means equalizers were disabled on the first config run, and options like
--video-output-levels or --brightness would not work. So we just
initialize the caps with a known superset. The player will not correctly
indicate when setting an eq doesn't work, but we're fine with it, as it
is a relatively cosmetic issue.
Second, it copied back the eq settings in the "wrong" moment (what
for?), which overwrote the settings in some cases.
Third, the eq was not reset correctly on vo init. This is needed to make
it behave the same as vo_opengl.
