It makes no sense to set the packet duration, because libavcodec doesn't
know the timebase. And in fact, no subtitle decoder accesses the packet
duration, except text subtitle converters, which are not relevant here.
So this code did nothing - drop it.
Also fix a blatantly incorrect comment.
If duration<0, it means the duration is unknown. Disable framedropping,
because end_time makes no sense in this case.
Also, strictly never drop the first frame.
This fixes weird behavior with the cover-art case (for the 100th time).
Commit 846257da introduced an accidental feature: if you kept seeking
(so playback never really resumes), the audio would never be played.
This was nice, but commit 4c25b000 accidentally removed it again (due
to the video_next_pts being earlier available than it used to be, so
audio could be played before the player executed the next queued seek).
Implicitly reintroduce the old behavior again by not decoding a second
video frame immediately. Usually, the second frame is used to compute
the frame duration needed to for accurate framedropping, but since the
first frame after a seek is never dropped, we don't need this.
Now the video code will queue the new frame to the VO immediately, and
since fill_audio_out_buffers() is called in the playloop before
write_video() and execute_queued_seek(), it never gets the chance to
enter STATUS_READY, and seeks will be silent.
This also has a nice side-effect: since the second frame is not decoded
and filtered, seeking becomes slightly faster (back to the same level
as with framedrop disabled).
It seems this still sometimes plays a period of audio when keeping a
seek key down. In my tests, this appeared to happen because the seek
finished before the next key repeat was sent.
In theory, timestamps can be negative, so we shouldn't just return -1
as special value.
Remove the separate code for clearing decode buffers; use the same code
that is used for normal seek reset.
Commit 5afc025c broke this. The reason is that mpctx->delay is updated
when a new video frame is added. This value is also needed to resync
audio, but it will be for the wrong PTS. They must be consistent with
each other, and if they aren't, initial sync will be off by N video
frames, which results at least in worse user experience.
This can be reproduced by for example heavily switching between normal
and 2x speed, or similar.
Fix by readding the video_next_pts field (keeping its use minimal,
instead of reverting the commit that removed it).
If video reaches EOF, and audio is also EOF (or is otherwise not
meaningful, like audio disabled), then the playback position was briefly
set to 0. Fix this by not trying to use a bogus audio PTS.
CC: @mpv-player/stable (maybe)
This simplifies the code, and fixes an odd bug: the second-last frame
was displayed for a very short duration if framedrop was enabled. The
reason was that basically the time difference between second-last and
last frame were skipped, because at this point EOF was already
signaled. Also see commit b0959488 for a similar issue in the
same code.
This removes the messiness of the next_frame 2-frame queue, and
strictly runs the "new frame" code when a frame is moved to the first
position of the queue, instead of somehow messing with return codes.
This also merges update_video() into video_output_image().
No functional changes. init_vo() is now needed a bit further down, and
moving it keeps definition and use close. adjust_sync() will be used by
a function further up in one of the following commits.
Add two new script environment variables 'video_in_dw' and
'video_in_dh', representing the display resolution of video. Along
with video resolution, sample ratio aspect can be calculated in
scripts.
Currently it's impossible to change sample ratio aspect with single
vapoursynth filter since '_SARNum' and '_SARDen' frame properties
from output clip will be ignored. A following 'dsize' filter is
necessary for this purpose.
This is independent of terminfo/termcap, and supports more keys.
Originally, the goal was just extending the set of supported key
sequences, but since the terminfo stuff actually makes this much harder,
and since it's a big blob of bloated legacy crap, just drop it. Instead,
use hardcoded tables.
It's pretty easy to get on the same level as the old code (with fewer
LOC), and we avoid additional error situations, such as mallocs which
could fail (the old code just ignores malloc failures). We also try to
support some xterm escape sequences, which are in relatively widespread
use. (I'm not sure about the urxvt ones.)
Trying to deal with xterm shift/ctrl/alt modifiers is probably a bit
overcomplicated, and only deals with prefixes - xterm randomly uses
prefix sequences for some keys, and suffixes for others (what the heck).
Additionally, try to drop unknown escape codes. This basically relies
on a trick: in almost 100% of all situations, a read() call will
actually return complete sequences (possibly because of pipe semantics
and atomic writes from the terminal emulator?), so it's easy to drop
unknown sequences. This prevents that they trigger random key bindings
as the code interprets the part after ESC as normal keys.
This also drops the use of terminfo for sending smkx/rmkx. It seems
even vt100 (to which virtually everything non-legacy is reasonably
compatible with) supports the codes we hardcode, so it should be fine.
This commit actually changes only the code if terminfo/termcap are not
found. The next commit will make this code default.
Surprisingly, WaitFor* works on console handles. We can simply run the
code for reading the console in a thread, and don't have to worry about
crazy win32 crap in the rest of the player's input code anymore.
This also fixes the issue that you couldn't unpause the player from the
terminal, because the player would stop polling for input.
We already redirect all terminal output through our own wrappers (for
the sake of UTF-8), so we might as well use it to handle ANSI escape
codes.
This also changes behavior on UNIX: we don't retrieve some escape codes
per terminfo anymore, and just hardcode them. Every terminal should
understand them.
The advantage is that we can pretend to have a real terminal in the
normal player code, and Windows atrocities are locked away in glue
code.
The original intention was probably to avoid unnecessarily high numbers
of wakeups. Change it to wait at most 25% of buffer time instead of 75%
until refilling. Might help with the dsound problems in issue #1024, but
I don't know if success is guaranteed.
So talking to a certain Intel dev, it sounded like modern VA-API drivers
are reasonable thread-safe. But apparently that is not the case. Not at
all. So add approximate locking around all vaapi API calls.
The problem appeared once we moved decoding and display to different
threads. That means the "vaapi-copy" mode was unaffected, but decoding
with vo_vaapi or vo_opengl lead to random crashes.
Untested on real Intel hardware. With the vdpau emulation, it seems to
work fine - but actually it worked fine even before this commit, because
vdpau was written and designed not by morons, but competent people
(vdpau is guaranteed to be fully thread-safe).
There is some probability that this commit doesn't fix things entirely.
One problem is that locking might not be complete. For one, libavcodec
_also_ accesses vaapi, so we have to rely on our own guesses how and
when lavc uses vaapi (since we disable multithreading when doing hw
decoding, our guess should be relatively good, but it's still a lavc
implementation detail). One other reason that this commit might not
help is Intel's amazing potential to fuckup anything that is good and
holy.
This could be used by VO implementations to report a recent vsync time
to the generic VO code, which in turn will use it and the display FPS
to estimate at which point in time the next vsync will happen.
This uses glXGetVideoSyncSGI() to check how many vsyncs happened since
the last flip_page() call. It allows checking a pattern of vsync
increments of at most 2 elements. For example, to check ~24 fps playback
on a ~60 Hz monitor, this can be used:
--vo=opengl:check-pattern=[3-2]:waitvsync
Whether the reported results are accurate or just plain wrong may depend
on the driver and if the waitvsync sub-option is used. There are no
guarantees.
This option is undocumented, and may be removed again in the near or
distant future.
For debugging (drawing fun plots with TOOLS/stats-conv.py).
Also move last_flip under the correct comment: it's not protected by the
lock, and can be accessed by the VO thread only.
Playing with high framedrop could make it run out of surfaces. In
theory, we wouldn't need an additional surface, if we could just clear
the vo_vaapi internal surface - but doing so would probably be a pain,
so I don't care.
Only reports the most recently entered output if the window is displayed on
2 or more outputs. Should be changed to the lowest fps of all outputs the
window is visible. Until no one complains this will have to wait.
Look for the VO framedropping for more information on this topic.
After a new file is loaded, playback never starts instantly. Rather, it
takes some playloop iterations until initial audio and video have been
decoded, and the outputs have been (lazily) initialized. This means you
will get status line updates between the messages that inform of the
initialized outputs. This is a bit annoying and clutters the terminal
output needlessly.
Fix this by never printing the status line before playback isn't fully
initialized. Do this by reusing the --term-playing-msg code (which
prints a message once playback is initialized). This also makes sure the
status line _is_ shown during playback restart when doing seeks.
It's possible that the change will make the output more confusing if for
some reason is stuck forever initializing playback, but that seems like
an obscure corner case that never happens, so forget about it.
print_status() is called at a later point anyway (and before sleeping),
so this code has little effect. This code was added in commit a4f7a3df5,
and I can't observe any problems with idle mode anymore.
Now print_status() is called from a single place only, within osd.c.
If the Xrandr configuration changes, re-read it. So if you change
display modes or screen configuration, it will update the framedrop
refresh rate accordingly.
This passes the rootwin to XRRSelectInput(), which may or may not be
allowed. But it works, and the documentation (which is worse than used
toilet paper, great job Xorg) doesn't forbid it, or in fact say anything
about what the window parameter is even used for.