Not needed anymore. I'm not opposed to having asm, but inline asm is too
much of a pain, and it was planned long ago to eventually get rid fo all
inline asm uses.
For the note, the inline asm use that was removed with the previous
commits was almost worthless. It was confined to video filters, and most
video filtering is now done with libavfilter. Some mpv filters (like
vf_pullup) actually redirect to libavfilter if possible.
If asm is added in the future, it should happen in the form of external
files.
No change in speed (or even slightly faster, though I tested with
progressive solid color video only), and normally we use libavformat's
vf_pullup anyway.
I didn't test the speed, but by default, this filter diverts to
libavfilter already. So this would help only if libavfilter is disabled,
or libavfilter doesn't have vf_noise (like on Libav). For these cases,
we still provide the (possibly but not necessarily) slower C
implementation of vf_noise.
This makes it multiple times slower. However, the output format (packed
YUV) isn't handled efficiently by anything to begin with, and I have no
clue we even have this filter. I guess it's one of these filters which
find some use sometimes, but are not of higher importance, which
justifies removing the faster inline asm.
This can be used to easily extent the status line for one's own needs.
I'm not experienced with lua so a few things could probably be done a
better way.
This is for the sake of multi-key commands again. This could break:
SPACE ignore
SPACE-SPACE command
while this worked:
SPACE-SPACE command
SPACE ignore
The reason being that if the shorter command was first in the list,
it would obviously match, and searching was stopped.
This is for the sake of multi-key combinations (see github issue #718).
Now a multi-key sequence isn't matched if any of the previous keys were
actually mapped.
We only need to track key up/down for a single key. There may be some
minor loss of robustness, but this can (probably) happen only if a VO or
user sends strange or complicated sequences of events. In the normal
case, what we do is more than enough. Most simplification comes from the
fact that mpv is not a game console, and users normally execute only one
action at once.
The window close button is usually mapped to the CLOSE_WIN pseudo-key.
Until now, --input-test treated this pseudo-key like any other key (like
the rest of the input handling code), so you couldn't close the window
in this mode. The manpage had silly instructions and warnings how to
deal with this.
Just always quit when CLOSE_WIN is received, and improve the
instructions.
The input code always supported combinations of multiple keys (even in
MPlayer, although there the code was active really only for mouse
buttons). This was arcance and also made the code more complicated. I
only know of a single person who ever made use of this feature.
Remove this feature, and repurpose some of the support code (e.g.
parsing, display of key combinations, etc.) to handle such multi-
combinations as sequences, instead of keys to be pressed at the same
time. This is much simpler and implements the feature requested in
github issue #718.
This commit will probably cause a bunch of regressions, since the input
handling code has some weird corner cases. I couldn't find any problems
when testing, though.
This avoids too many realloc() calls if the caller is appending to an
audo buffer. This case is actually quite noticeable when using something
that buffers a large amount of audio.
For some reason, the buffered_audio variable was used to "cache" the
ao_get_delay() result. But I can't really see any reason why this should
be done, and it just seems to complicate everything.
One reason might be that the value should be checked only if the AO
buffers have been recently filled (as otherwise the delay could go low
and trigger an accidental EOF condition), but this didn't work anyway,
since buffered_audio is set from ao_get_delay() anyway at a later point
if it was unset. And in both cases, the value is used _after_ filling
the audio buffers anyway.
Simplify it. Also, move the audio EOF condition to a separate function.
(Note that ao_eof_reached() probably could/should whether the last
ao_play() call had AOPLAY_FINAL_CHUNK set to avoid accidental EOF on
underflows, but for now let's keep the code equivalent.)
This replaces translate_key_input with a solution that gives mpv more
control over how keyboard input is converted to unicode. As a result:
- Key up/down events are generated the same way for all keys.
- Dead keys generate their base character instead of being combined with
the following character.
- Many Ctrl and Ctrl+Alt key combinations that were previously broken
are fixed, since it's possible to discover the base keys.
- AltGr doesn't produce special characters when mp_input_use_alt_gr is
false.
This also fixes some logic to do with detecting AltGr and adds proper
UTF-16 decoding.
This was reported with PulseAudio 2.1. Apparently it still has problems
with reporting the correct delay. Since ao_pulse.c still has our custom
get_delay implementation, there's a possibility that this is our fault,
but this seems unlikely, because it's full of workarounds for issues
like this. It's also possible that this problem doesn't exist on
PulseAudio 5.0 anymore (I didn't explicitly retest it).
The check is general and works for all push based AOs. For pull based
AOs, this can't happen as pull.c implements all the logic correctly.
This should probably be an AO function, but since the playloop still has
some strange stuff (using the buffered_audio variable instead of calling
ao_get_delay() directly), just leave it and make it more explicit.
This collects statistics and other things. The option dumps raw data
into a file. A script to visualize this data is included too.
Litter some of the player code with calls that generate these
statistics.
In general, this will be helpful to debug timing dependent issues, such
as A/V sync problems. Normally, one could argue that this is the task of
a real profiler, but then we'd have a hard time to include extra
information like audio/video PTS differences. We could also just
hardcode all statistics collection and processing in the player code,
but then we'd end up with something like mplayer's status line, which
was cluttered and required a centralized approach (i.e. getting the data
to the status line; so it was all in mplayer.c). Some players can
visualize such statistics on OSD, but that sounds even more complicated.
So the approach added with this commit sounds sensible.
The stats-conv.py script is rather primitive at the moment and its
output is semi-ugly. It uses matplotlib, so it could probably be
extended to do a lot, so it's not a dead-end.
This was accidentally completely destroyed with commit 24f1878e. I
didn't notice it when testing, because forward seeking still worked
mostly.
The issue was that dvd_seek_to_time() actually called stream_seek(),
which was supposed to call the byte-level seek function dvd_seek(). So
we have to restore this function, and replace all generic stream calls
with stream_dvd.c internal ones. This also affects stream->pos (now a
random number as far as stream_dvd.c is concerned) and stream_skip().
The recent change of waking up the playback thread using a wakeup pipe
doesn't work on windows, because windows is horrible. So use a condition
variable instead to wake up the thread. To make things worse, jackaudio
is also horrible and "disallows" the use of mutexes, so all we can do is
implementing a half-solution that is not race condition free.
It would probably better to give up on this lock-free crap in the pull-
API audio path.
Mostly untested.
The audio subsystem now wakes up the playback thread explicitly, and we
don't need this anymore.
It still could cause dropouts and such if there are bugs in the recently
introduced audio changes, so this is a thing to watch out for.
Same change as in e2184fcb, but this time for pull based AOs. This is
slightly controversial, because it will make a fast syscall from e.g.
ao_jack. And according to JackAudio developers, syscalls are evil and
will destroy realtime operation. But I don't think this is an issue at
all.
Still avoid locking a mutex. I'm not sure what jackaudio does in the
worst case - but if they set the jackaudio thread (and only this thread)
to realtime, we might run into deadlock situations due to priority
inversion and such. I'm not quite sure whether this can happen, but I'll
readily follow the cargo cult if it makes hack happy.
I'm not quite sure why ao_pulse needs this. It was broken when a thread
to fill audio buffers was added to AO - the pulseaudio callback was
waking up the playback thread, not the audio thread. But nobody noticed,
so it can't be very important. In any case, this change makes it wake up
the audio thread instead (which in turn wakes up the playback thread if
needed).
And also add a function ao_need_data(), which AO drivers can call if
their audio buffer runs low.
This change intends to make it easier for the playback thread: instead
of making the playback thread calculate a timeout at which the audio
buffer should be refilled, make the push.c audio thread wakeup the core
instead.
ao_need_data() is going to be used by ao_pulse, and we need to
workaround a stupid situation with pulseaudio causing a deadlock because
its callback still holds the internal pulseaudio lock.
For AOs that don't call ao_need_data(), the deadline is calculated by
the buffer fill status and latency, as before.
qscale export has been completely removed from Libav 10, and FFmpeg has
an alternative API, so this code does nothing and only causes
deprecation warnings on Libav.
