| Commit message (Collapse) | Author | Age | Files | Lines |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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.
|
|
|
|
| |
Was forgotten in commit 1b398e99.
|
|
|
|
|
| |
This considered audio to never reach EOF when audio was disabled,
instead of always being EOF.
|
|
|
|
| |
Unlike --loop, loops a file instead of the playlist.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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 EOF problems happen at least with PulseAudio, but since it's hard
to reproduce, let ao_null optionally simulate it.
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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().
|
|
|
|
| |
It did nothing; the real check is in push.c.
|
|
|
|
|
| |
Set a zero sized input region for the osd surfaces in order to avoid getting
input for the subsurfaces which might dissapear any time.
|
| |
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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.
|
| |
|
|
|
|
|
|
|
|
|
| |
And slightly adjust the semantics of MPV_EVENT_PAUSE/MPV_EVENT_UNPAUSE.
The real pause state can now be queried with the "core-idle" property,
the user pause state with the "pause" property, whether the player is
paused due to cache with "paused-for-cache", and the keep open event can
be guessed with the "eof-reached" property.
|
|
|
|
|
|
|
| |
This property is set to "yes" if playback was paused due to --keep-open.
The change notification might not always be perfect; maybe that should
be improved.
|
|
|
|
|
|
|
|
|
| |
Currently this is (probably) equivalent to "paused-for-cache", but the
latter is a bit special, while this new property is a bit more general.
One case where they might actually be different is dvdnav menus, but I
haven't checked.
Also add property change notifications for these two properties.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
We were relying on vsscript_freeScript() to take care of proper
termination. But it doesn't do that: it doesn't wait for the filters to
finish and exit at all. Instead, it just destroys all objects, which
causes the worker threads to crash sometimes.
Also, we're supposed to wait for the frame callback to finish before
freeing the associated node.
Handle this by explicitly waiting as far as we can. Probably fixes
crashes on seeking, although VapourSynth itself might also need some
work to make this case completely stable.
|
|
|
|
|
|
|
|
|
| |
It's a bit strange to allow this, so get rid of it.
This probably breaks a bunch of user config files.
The client API still allows setting them with MPV_FORMAT_FLAG with a
value of 1 (i.e. true), but I guess this is tolerable.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
The most user visible change is that "420p" is now displayed as
"yuv420p". This is what FFmpeg uses (almost), and is also less confusing
since "420p" is often confused with "420 pixels vertical resolution".
In general, we return the FFmpeg pixel format name. We still use our own
old mechanism to keep a list of exceptions to provide compatibility for
a while.
Also, never return NULL for image format names. If the format is unset
(0/IMGFMT_NONE), return "none". If the format has no name (probably
never happens, FFmpeg seems to guarantee that a name is set), return
"unknown".
|
|
|
|
|
|
|
| |
This re-allows the previous behaviour of being able to reencode with
metadata removed, which is useful when encoding "inconsistently" tagged
data for a device/player that shows file names when tags are not
present.
|
|
|
|
|
|
|
|
| |
Before this commit, the filter attempted to keep the vsscript state
(p->se) even when the script was reloaded. Change it to destroy the
script state too on reloading. Now no workaround for LoadPlugin is
necessary, and this also fixes a weird theoretical race condition when
destroying and recreating the mpv source filter.
|
|
|
|
| |
Signed-off-by: wm4 <wm4@nowhere>
|
|
|
|
|
| |
It was disabled by default, works only for analogue radio, and I bet
nobody uses it.
|
|
|
|
|
|
|
|
|
|
|
| |
I hate tabs.
This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).
|
| |
|
| |
|
|
|
|
| |
Signed-off-by: wm4 <wm4@nowhere>
|
|
|
|
|
|
|
| |
This is a read-only property that uses VFCTRL_GET_METADATA
to retrieve mp_tags metadata from a filter specified by label
Signed-off-by: wm4 <wm4@nowhere>
|
|
|
|
|
|
|
|
| |
store it as mp_tas and add VFCTRL_GET_METADATA to access it from elsewhere
Signed-off-by: wm4 <wm4@nowhere>
old-configure test by wm4.
|
|
|
|
| |
use avfilter_graph_parse_ptr for ffmpeg
|
|
|
|
| |
Signed-off-by: wm4 <wm4@nowhere>
|
| |
|
| |
|
|
|
|
|
|
| |
rename add_metadata to the more genera/descriptive mp_tags_copy_items_from_av_dictionary
Signed-off-by: wm4 <wm4@nowhere>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
These playlist parsers are all what's left from the old mplayer playlist
parsing code. All of it is old code that does little error checking; the
type of C string parsing code that gives you nightmare.
Some playlist parsers have been rewritten and are located in
demux_playlist.c. The removed formats were not reimplemented. ASX and
SMIL use XML, and since we don't want to depend on a full blown XML
parser, this is not so easy. Possibly these formats could be supported
by writing a very primitive XML-like lexer, which would lead to success
with most real world files, but I haven't attempted that. As for NSC, I
couldn't find any URL that worked with MPlayer, and in general this
formats seems to be more than dead.
Move playlist_parse_file() to playlist.c. It's pretty small now, and
basically just opens a stream and a demuxer. No use keeping
playlist_parser.c just for this.
|
|
|
|
| |
mpv was resizing to the same size before it went to fullscreen, we don't need to schedule a resize because the compositor will send a configure event with the new dimensions and thats when we should do it.
|
|
|
|
|
|
|
|
|
| |
Basically a cheap hack to fix that the --msgmodule prefix will cause an
unwanted linebreak by making the line too long.
Suggested by Hamuko in github issue #710.
Fixes #710.
|
|
|
|
|
|
|
|
|
| |
This is needed if you want to reimplement the status line in lua
I could only test drop-frame-count because I didn't find an easy way to
trigger paused-for-cache and total-avsync-change
Signed-off-by: wm4 <wm4@nowhere>
|
|
|
|
|
| |
Apparently there is no real format negotiation, so this can actually
happen.
|
|
|
|
| |
Instead of crashing by possibly passing a NULL pointer to VapourSynth.
|
|
|
|
| |
Unfortunately, reloading on seeking causes real problems.
|
|
|
|
| |
This will be less confusing.
|
|
|
|
| |
We provide some "official" utility functions for this.
|
|
|
|
| |
Pretty much experimental for issue #661.
|
|
|
|
| |
Should make integreating with some event loops easier. Untested.
|
|
|
|
|
|
|
|
|
|
|
|
| |
Mainly meant to apply simple VapourSynth filters to video at runtime.
This has various restrictions, which are listed in the manpage.
Additionally, this actually copies video frames when converting frame
references from mpv to VapourSynth, and a second time when going from
VapourSynth to mpv. This is inefficient and could probably be easily
improved. But for now, this is simpler, and in fact I'm not sure if
we even can references VapourSynth frames after the core has been
destroyed.
|
|
|
|
|
|
| |
Commit 2c2c1203 sorted the output of --list-options, but the same code
ias also used for listing sub-options, such as --vo=scale:help. For sub-
options, the order actually matters.
|
| |
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
| |
Until now, --list-options printed options in random order. There
literally wasn't any logic in its order, they just appeared as they were
declared. So just sort them.
Note that we can't sort them in advance, because for certain things
internal to m_config, the order actually matters.
Also we're using strcasecmp(), which is bad (locale dependent), but this
is output intended for human consumption, so it's not a problem.
|
|
|
|
|
|
| |
This used to display the property type, but it was not always correct or
even available. The way the property mechanism works, we can know this
only at runtime.
|
|
|
|
|
|
|
|
|
| |
Otherwise, the client API user could not know why playback was stopped.
Regarding the fact that 0 is used both for normal EOF and EOF on error:
this is because mplayer traditionally did not distinguish these, and in
general it's hard to tell the real reason. (There are various weird
corner cases which make it hard.)
|