This is for users who don't like changes. I'm hoping it will make the
process of cleaning up key bindings less bumpy.
It should be mentioned in the release notes of the next release.
It was more complicated than it had to be: the audio thread already
determines whether audio has ended, so we can use that. Remove the
separate logic for draining.
Commit 957097 attempted to use PA_STREAM_FAIL_ON_SUSPEND to make
ao_pulse exit if the stream was started suspended.
Unfortunately, PA_STREAM_FAIL_ON_SUSPEND is active even during playback.
If you pause mpv, pulseaudio will close the actual audio device after a
while (or something like this), and unpausing won't work. Instead, it
will spam "Entity killed" error messages.
Undo this change and check for suspended audio manually during init.
CC: @mpv-player/stable
This doesn't look to be needed anymore. Fullscreening with both the NSView
and the NSWindow API works correctly. I guess this was forgotten in from older
code which changed presentation options directly for going fullscreen.
--x11-netwm=yes now forces NetWM fullscreen, while --x11-netwm=auto
(detect whether NetWM fullsctreen support is available) is the old
behavior and still the default.
See #888.
Unfortunately using dispatch_sync for synchronization turned out to be really
bad for us. It caused a wide array of race conditions, deadlocks, etc.
Moving to a very simple mutex. It's not clear to me how to do liveresizing
with this, for now it just flickers with is unacceptable (maybe I'll draw
black instead).
This should fix all the threading cocoa bugs. Reopen if it's not the case!
Fixes#751Fixes#1129
Also recreate ASS_Library on every file played. This means we can move
the code out of main.c as well.
Recreating the ASS_Library object has no disadvantages, because it
literally stores only the message callback, the (per-file) font
attachment as byte arrays, and the set of style overrides. Hopefully
this thing can be removed from the libass API entirely at some point.
The only reason why the player core creates the ASS_Renderer, instead
of the subtitle renderer, is because we want to cache the loaded fonts
across ordered chapter transitions, so this probably still has to stay
around for now.
When the VO was moved it its own thread, responsibility for redrawing
was given to the VO thread itself. So if there was a condition that
indicated that redrawing was required, like expose events or certain
VOCTRLs, the VO thread was redrawing itself.
This worked fine, but there are some corner cases where this works
rather badly. E.g. if I fullscreen the player and hit panscan controls
with mpv's default autorepeat rate, playback stops. This happens because
the VO redraws itself after every panscan change command. Running each
(repeated) command takes so long due to redrawing and (involuntary)
waiting on vsync, that it never leaves the input processing loop while
the key is held down. I suspect that in my case, redrawing in fullscreen
mode just gets slow enough that it takes 2 vsyncs instead of 1 on
average, and the processing time gets larger than the autorepeat delay.
Fix this by taking redraw control from the VO, and instead let the
playloop issue a "real" redraw command to the VO if needed. This
basically reverts redraw handling to what it was before moving the VO to
a thread.
CC: @mpv-player/stable
Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The
main use of this was that you could pass a bitmask of these flags to
uninit_player(). Except in some situations where you wanted to
uninitialize nearly everything, this wasn't really useful. Moreover, it
was quite annoying that subsystems had most of the code in a specific
file, but the uninit code in loadfile.c (because that's where
uninit_player() was implemented).
Simplify all this. Remove the flags; e.g. instead of testing for the
INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code
to separate functions, e.g. uninit_audio_out().
Sometimes, ao_pulse starts in suspended mode, which means playback is
essentially paused in pulseaudio. This gives the impression that mpv is
hanging, since it times video against the audio playback progress, and
audio never makes progress in this state.
I'm not sure if this will help - possibly it does with mixed
pulseaudio/alsa setups. However, if the alsa setup has the pulseaudio
plugin, alsa will hang too. But there's still a chance we get less
blame for pulseaudio messes.
This gets rid of this warning:
Could not update timestamps for skipped samples.
This required an API addition to FFmpeg (otherwise it would instead
doing arithmetic on the timestamps itself), so whether it works depends
on the FFmpeg version.
For the sake of libmpv. Might make things much easier for the user,
especially on Windows. On the other hand, it's a bit sketchy that a
command exists that makes the player access arbitrary memory regions.
(But do note that input commands are not meant to be "secure" and never
were - for example, there's the "run" command, which obviously allows
running random shell commands.)
Somewhat more flexible: now there's a separate overlay struct, and you
don't need to coerce all state into struct sub_bitmap. Also, removing
the previous mapping (munmap call) is now all in one place, the
replace_overlay function.
Makes the next commit easier to implement.
Use percent-pos instead, which is exactly the same, except with the
range 0.0-100.0.
I'm not sure how this got there; it was probably introduced and then
removed again as percent-pos got more precise.
CC: @mpv-player/stable
Another fallout resulting from the changes whether or not to wait for
mapping the window. In this case, it obviously makes no sense to wait
for mapping, because the root window is always mapped. Mapping will
never happen, and it would wait forever.
Fixes#1139.
CC: @mpv-player/stable
The messages "Audio: no audio" and "Video: no video" could be printed
twice each if initializing them failed. Prevent his silliness.
CC: @mpv-player/stable
Apparently this is what users want. When playing with normal speed,
nothing is done. When playing slower than normal, resampling is used
instead, because scaletempo (which does the pitch correction) adds
too many artifacts.
Although the "af" command already could do this, it seems it's better
to introduce a lower level mechanism for now. This avoids some messy
issues, since that code would recursive call reinit_audio_chain().
To be used by the next commit.
There's no real reason why audio_init_filter() should exist. Just use
af_init or af_reinit directly. (We lose a useless message; the same
information is printed in a quite close place with more details.)
Requires less code, and the way the filter chain is marked as having
failed to initialize allows just switching off audio instead of
crashing if trying to insert a volume filter in mixer.c fails, and
recreating the old filter chain fails too.
Simple fix for issue #1137.
Since all sub-bitmaps are packed on a larger texture, there's still a
"fall off" on the border due to the linear scaling. This could be
fixed by constraining each sub-bitmap to its own texture, or by
clamping on the shader level, but I don't care for now.
At least on kwin, we decide to proceed without waiting for the window
being mapped (due to the frame exts hack, see commit 8c002b79). But that
leaves us with a window size of 0x0, which causes VdpOutputSurfaceCreate
to fail. This prints some warnings, although vo_vdpau recovers later and
this has no other bad consequences.
Do the following things to deal with this:
- set the "known" window size to the suggested window size before the
window is even created
- allow calling XGetGeometry on the window even if the window is not
mapped yet (this should work just fine)
- make the output surface minimum size 1x1
Strictly speaking, only one of these would be required to make the
warning disappear, but they're all valid changes and increase robustness
and correctness. At no point we use a window size of 0x0 as magic value
for "unset" or unknown size, so keeping it unset has no purpose anyway.
CC: @mpv-player/stable
Commit 50e131b43e happened to make it work for DVD (because the higher
bits of the ID are masked in the DVD case), but failed for Bluray. This
probably fixes it, although I don't have a sample to multiple streams to
confirm it really does it right.
CC: @mpv-player/stable
This would play some silence in case video was slower than audio. If
framedropping is already enabled, there's no other way to keep A/V
sync, short of changing audio playback speed (which would give worse
results). The --audiodrop option inserted silence if there was more
than 500ms desync.
This worked somewhat, but I think it was a silly idea after all. Whether
the playback experience is really bad or slightly worse doesn't really
matter. There also was a subtle bug with PTS handling, that apparently
caused A/V desync anyway at ridiculous playback speeds.
Just remove this feature; nobody is going to use it anyway.
When embedding, if the parent window is destroyed, it will cause mpv's
window to be destroyed as well. Since WM_USER wakeups are sent to the
window, destroying the window will prevent wakeups and cause uninit to
hang.
Fix this by quitting the event loop on WM_DESTROY. Events should only be
processed for the lifetime of the window, from CreateWindowEx to
WM_DESTROY. After the event loop is finished, mp_dispatch_queue_process
can handle any remaining requests.
Might matter when libavformat tries to do tiny seekbacks in an
unseekable stream, and the seekback buffer isn't large enough. In this
case, seeking would fail, and would drop the current buffer. The
seekback would end up dropping future data.
This change probably doesn't have any observable effects. libavformat
normally has its own stream buffer, and demux_mkv.c tries carefully
never to seek back.
