External subtitles are always read as eager, so they do not need to be
changed on pause/unpause. Don't do the refresh seek since it will just
buffer forever. Fixes f40bbfec4f.
This replaces the previous commit and makes more sense. The internal
demux marked tracks as eager depending on their type and for subtitles
it would always lazily read them unless there happened to be no
available av stream. However, we want the sub stream to be eager if the
player is paused. The existing subtitle is still preserved on the
screen, but if the user changes tracks that's when the problem occurs.
So to handle this case, propagate the mpctx->paused down to the stream
selection logic. This modifies both demuxer_refresh_track and
demuxer_select_track to take that boolean value. A few other parts of
the player use this, but we can just assume false there (no change in
behavior from before) since they should never be related to subtitles.
The core player code is aware of its own state naturally, and can always
pass the appropriate value so go ahead and do so. When we change the
pause state, a refresh seek is done on all existing subtitle tracks to
make sure their eager state is the appropriate value (i.e. so it's not
still set to eager after a pause and a track switch). Slightly invasive
change, but it works with the existing logic instead of going around it
so ultimately it should be a better approach. We can additionally remove
the old force boolean from sub_read_packets since it is no longer
needed.
No wonder wm4 wanted to get rid of this. This option requires touching a
bunch of crap in the core player code. --stream-record works perfectly
fine and is a lot nicer so there's no need for this to exist anymore.
We should account for ao queue when setting playback position on EOF,
previously we were using the pts value corresponding to the start of
the ao queue, rather than the currently playing audio.
This fixes time-remaining being a negative number when mpv seeks to EOF
while playback is paused.
There's an edge cause with gapless audio and pausing. Since, gapless
audio works by sending an EOF immediately, it's possible to pause on the
next file before audio actually finishes playing and thus the sound gets
cut off. The fix is to simply just always do an ao_drain if the ao is
about to set a pause on EOF and we still have audio playing.
Fixes#8898.
In debug mode the macro causes an assertion failure.
In release mode it works differently and tells the compiler that it can
assume the codepath will never execute. For this reason I was conversative
in replacing it, e.g. in mpv-internal code that exhausts all valid values
of an enum or when a condition is clear from directly preceding code.
mpv's core already keeps track of whether or not it thinks a track is an
image. Some VOs (i.e. wayland) would benefit from knowing if what is
currently being displayed is an image or not so add a new VOCTRL that
signals this anytime we load a new file with a VO. Additionally, let's
add a helper enum for signaling the kind of content that is being
displayed. There is now MP_CONTENT_NONE (strictly for force window being
used on a track with no image/video), MP_CONTENT_IMAGE, and
MP_CONTENT_VIDEO. See the next commit for the actual usage of this (with
wayland).
Normally with the keep-open option, mpv is supposed to set the
eof-reached property to true so clients can possibly do interesting
things at this step. However, there was actually an edge case where this
property notification did not occur. If you use keep-open and then seek
in the file past the end (so mpv stops), property notification is not
actually sent in this case. Internally, mpv does a very exact seek at
this step which also ends playback, but it does not set STATUS_EOF to
the ao and vo before the core idle state is updated. To fix this edge
case, it's simply just a matter of explictly setting STATUS_EOF after
seek_to_last_frame in handle_keep_open. This logic will only ever
trigger if keep-open is being used and the seek goes past the end of the
file, so we know that there will always be an EOF here.
The stop-screensaver option is currently limited to a simple yes/no
option. While the no option does always disable mpv trying to stop the
screensaver, yes does not mean the screensaver is always stopped. The
screensaver will be enabled again depending on certain conditions (like
if the player is paused). Simply introduce a new value for this option,
always, which does exactly what the name implies: the screensaver will
always be disabled.
Was tripping -Wparantheses as the && after the || was not explicitly
wrapped. However, due to operator precedence the intended effect was
still correct.
Not all deprecated symbols were removed. Only three events were removed for now
since these are not used internally.
This bumps the library version to 2.0.
Allow --keep-open=always to work with --loop-playlist, where
before this patch it would work only on the last playthrough of the
playlist. This patch allows it to work on all playthroughs.
Fixes#9470.
When playing a new file, if paused_for_cache was true before being reset
the AO would never be unpaused because that state was thrown away, leaving it stuck.
Fix this by updating the pause state before resetting that variable. Note that this
does not make the second update_internal_pause_state() call redundant.
This fixes the same bug fb5d976cb0 was supposed to.
Pause can be changed during a file change, such as with for example
--reset-on-next-file=pause, or in hooks, or by being quick, and in this
case the AO's pause state was not updated correctly. mpctx->ao_chain is
only set if playback is fully initialized, while the AO itself in
mpctx->ao can be reused across files.
Fix this by always running set_pause_state() if the pause option is
changed. Could cause new bugs since running this used to be explicitly
avoided outside of the loaded state. The handling of time_frame is
potentially worrisome.
Regression due to recent audio refactor; before that, the AO didn't have
a separate/persistent pause state.
Fixes: #8079
Since b74c09efbf, audio-only files let you seek to arbitrary points
beyond the end of the file (but still displayed the time clamped to the
nominal file duration). This was confusing and just not wanted. The
reason is probably that the commit removed setting the audio PTS for
data before the seek target, so if you seek past the end of the file,
the audio PTS is never set. This in turn means the logic to determine
the current playback time has no PTS at all, and thus falls back to the
seek PTS.
This happened in the past for other reasons (like efe43d768f). I have
enough of this, so I'm just changing the code to clamp the seek
timestamp to a "known" range. Do this when seeking ends, because in the
fallback case, the playback time shouldn't be stuck at e.g. "end +
seek_argument". Also do it when initiating a new seek (mp_seek), because
if the previous seek hasn't finished yet, it shouldn't add them up and
allow it to go "out of range" either. The latter is especially relevant
for hr-seeks.
Doing this clamping is problematic because the duration is a possibly
invalid value from the demuxer, or just missing. Especially with
timestamp resets, fun sometimes happens, and in these situations it
might be better not to clamp.
One could argue you should just use the last audio timestamp returned by
the decoder or demuxer (even if that directly conflicts with --end), but
that sounds even more hairy.
In summary: what a dumb waste of time, what the fuck.
Add a property that returns whether the window is focused, currently
only for X11 and Wayland.
My use cause for this is having an equivalent of pause-when-minimize.lua
for tiling window managers: make mpv play only while it's in the current
workspace or is focused (I'm fine with either one but prefer focus).
On X I do this by observing display-names, which is empty when the
rectangles of the display and mpv don't intersect, but on Wayland its
value doesn't change when mpv leaves the current workspace (and the same
check doesn't work since the geometries still intersect).
This could later be made writable as requested in #6252.
Note that on Wayland se shouldn't consider an unactivated window with
keyboard input focused.
The wlroots compositors I tested set activated after changing the
keyboard focus, so if you set wl->focused only in
keyboard_handle_enter() and keyboard_handle_leave() to avoid adding the
"has_keyboard_input" member, focused isn't set to true when first
opening mpv until you focus another window and focus mpv again.
Conversely, if that order can't be assumed for all compositors, we
should toggle wl->focused when necessary in keyboard_handle_enter() and
keyboard_handle_leave() as well as in handle_toplevel_config().
This replaces the two buffers (ao_chain.ao_buffer in the core, and
buffer_state.buffers in the AO) with a single queue. Instead of having a
byte based buffer, the queue is simply a list of audio frames, as output
by the decoder. This should make dataflow simpler and reduce copying.
It also attempts to simplify fill_audio_out_buffers(), the function I
always hated most, because it's full of subtle and buggy logic.
Unfortunately, I got assaulted by corner cases, dumb features (attempt
at seamless looping, really?), and other crap, so it got pretty
complicated again. fill_audio_out_buffers() is still full of subtle and
buggy logic. Maybe it got worse. On the other hand, maybe there really
is some progress. Who knows.
Originally, the data flow parts was meant to be in f_output_chain, but
due to tricky interactions with the playloop code, it's now in the dummy
filter in audio.c.
At least this improves the way the audio PTS is passed to the encoder in
encoding mode. Now it attempts to pass frames directly, along with the
pts, which should minimize timestamp problems. But to be honest, encoder
mode is one big kludge that shouldn't exist in this way.
This commit should be considered pre-alpha code. There are lots of bugs
still hiding.
This can happen if a file contains headers only, or if decoding of all
data failed, and such. Interestingly, it also happened when doing "mpv
--loop emptyfile.png", because demux_mf still detects file formats by
file extension.
In this situation, the player burned a lot of CPU by restarting playback
after doing nothing. Although such "degenerate" behavior can't be
avoided in all situations (what if you loop a file with 1 audio
sample?), detecting this seems to make sense.
For now, this actually decrements the loop count by 1, and then errors
out with a warning. Works for --loop and --ab-loop, while
--loop-playlist already avoids this situation with an existing
mechanism.
Commit fcf0b80dc9 fixed this the first time. Commit 85576f31a9
"accidentally" removed this code again. The commit message justifying
the removal is correct, except it doesn't take other side-effects of the
state machine into account. I obviously didn't remember what exactly
this was about. So add a comment explaining it this time.
Just apply it again; the thing the latter commit fixed still works.
Fixes: #7819
This brings the displayed percentage closer to the exact number and
allows mpv to more frequently display 100% when it finishes playing a
typical video or audio file.
If the user manages to run a "loadfile x append" command before the loop
in mp_play_files() is entered, then the player could start playing
these. This isn't expected, because appending files to the playlist in
idle mode does not normally start playback. It could happen because
there is a short time window where commands are processed before the
loop is entered (such as running the command when a script is loaded).
The idle mode semantics are pretty weird: if files were provided in
advance (on the command line), then these should be played immediately.
But if idle mode was already entered, and something is appended to the
playlist using "append", i.e. without explicitly triggering playback,
then it should remain in idle mode.
Try to follow this by redefining PT_STOP to strictly mean idle mode.
Remove the playlist->current check from idle_loop(), since only the
stop_play field counts now (cf. what mp_set_playlist_entry() does).
This actually introduces the possibility that playlist->current, and
with it playlist-pos, are set to something, even though playback is not
active or being started. Previously, this was only possible during state
transitions, such as when changing playlist entries.
Very annoyingly, this means the current way MPV_EVENT_IDLE was sent
doesn't work anymore. Logically, idle mode can be "active" even if
idle_loop() was not entered yet (between the time after mp_initialize()
and before the loop in mp_play_files()). Instead of worrying about this,
redo the "idle-active" property, and deprecate the event.
See: #7543
Move the "old" mostly command line parsing and option management related
code to m_config_frontend.c/h. Move the the code that enables other part
of the player to access options to m_config_core.c/h. "frontend" is out
of lack of creativity for a better name.
Unfortunately, the separation isn't quite clean yet. m_config_frontend.c
still references some m_config_core.c implementation details, and
m_config_new() is even left in m_config_core.c for now. There some odd
functions that should be removed as well (marked as "Bad functions").
Fixing these things requires more changes and will be done separately.
struct m_config is left with the current name to reduce diff noise.
Also, since there are a _lot_ source files that include m_config.h, add
a replacement m_config.h that "redirects" to m_config_core.h.
When the demuxer cache read until the end of the stream, and was
finished and completely inactive, the cache properties were not updated
anymore via MP_EVENT_CACHE_UPDATE.
Unfortunately, many cache properties depend on the current playback
position, such as cache-duration or fw-bytes. This is especially visible
on the OSC. If everything was cached, seeking around didn't update the
displayed forward cache duration.
That means checking demuxer_reader_state.idle is not enough. You also
need to check whether the current playback position changed.
Fix this by explicitly using the current playback position, and update
the properties if it changed "enough". "Enough" is 1 second of media
time in this example, which may or may not be appropriate.
In general, this could probably be done better. There are many other
triggers that change the cache state and that are not covered. For now
I'm content with getting rid of the obvious problems.
I think the OSC problem in particular was caused by changing it from
polling to using property change notifications.
Instead of having f_decoder_wrapper create its own copy of the entire
mpv option tree, create a struct local to that file and move all used
options to there.
movie_aspect is used by the "video-aspect" deprecated property code. I
think it's probably better not to remove the property yet, but
fortunately it's easy to work around without needing special handling
for this option or so.
correct_pts is used to prevent use of hr-seek in playloop.c. Ignore
that, if you use --no-correct-pts you're asking for trouble anyway. This
is the only behavior change.
See manpage additions. This has been a topic in MPlayer/mplayer2/mpv
since forever. But since libavcodec multi-threaded decoding was added,
I've always considered this pointless. libavcodec requires you to
"preload" it with packets, and then you can pretty much avoid blocking
on it, if decoding is fast enough.
But in some cases, a decoupled decoder thread _might_ help. Users have
for example come up with cases where decoding video in a separate
process and piping it as raw video to mpv helped. (Or my memory is
false, and it was about vapoursynth filtering, who knows.) So let's just
see whether this helps with anything.
Note that this would have been _much_ easier if libavcodec had an
asynchronous (or rather, non-blocking) API. It could probably have
easily gained that with a small change to its multi-threading code and a
small extension to its API, but I guess not.
Unfortunately, this uglifies f_decoder_wrapper quite a lot. Part of this
is due to annoying corner cases like legacy frame dropping and hardware
decoder state. These could probably be prettified later on.
There is also a change in playloop.c: this is because there is a need to
coordinate playback resets between demuxer thread, decoder thread, and
playback logic. I think this SEEK_BLOCK idea worked out reasonably well.
There are still a number of problems. For example, if the demuxer cache
is full, the decoder thread will simply block hard until the output
queue is full, which interferes with seeking. Could also be improved
later. Hardware decoding will probably die in a fire, because it will
run out of surfaces quickly. We could reduce the queue to size 1...
maybe later. We could update the queue options at runtime easily, but
currently I'm not going to bother.
I could only have put the lavc wrapper itself on a separate thread. But
there is some annoying interaction with EDL and backward playback shit,
and also you would have had to loop demuxer packets through the
playloop, so this sounded less annoying.
The food my mother made for us today was delicious.
Because audio uses the same code, also for audio (even if completely
pointless).
Fixes: #6926
I may (optionally) move decoding to a separate thread in a future
change. It's a bit attractive to move the entire decoder wrapper to
there, so if the demuxer has a new packet, it doesn't have to wake up
the main thread, and can directly wake up the decoder. (Although that's
bullshit, since there's a queue in between, and libavcodec's
multi-threaded decoding plays cross-threads ping pong with packets
anyway. On the other hand, the main thread would still have to shuffle
the packets around, so whatever, just seems like better design.)
As preparation, there shouldn't be any mutable state exposed by the
wrapper. But there's still a large number of corner-caseish crap, so
just use setters/getters for them. This recorder thing will inherently
not work, so it'll have to be disabled if threads are used.
This is a bit painful, but probably still the right thing. Like
speculatively pulling teeth.
This was a hack that attempted to line up external audio tracks with
video. The problem is that if you do a keyframe seek backwards, video
will usually seek much farther back than audio (due to much higher
keyframe aka seek point distances). The hack somehow made seeking a 2
step process.
This existed in 4 different forms in the history of this code base, and
it was always very cumbersome. We mostly needed this for ytdl_hook (I
think?), which uses the 4th form, which is nicely confined to
demux_timeline and is unrelated to the "external" audio tracks in the
high level player.
Since this is (probably) not really widely needed anymore, get rid of
it. Better do this now, than when somehow rewriting all the seeking code
(which might happen in this decade or the next or so) and when it
wouldn't be easily revertable anymore in case we find we "really" need
it unlike expected.
There is no issue if hr-seeks are used. Also, you can still use edl
files to "bundle" multiple streams as if it was a single stream (this is
what ytdl_hook does now).
Try to deal with various corner cases. But when I fix one thing, another
thing breaks. (And it's 50/50 whether I find the breakage immediately or
a few months later.) So results may vary.
The default for--hr-seek is changed to "default" (not creative enough to
find a better name). In this mode, audio seeking is exact if there is no
video, or if the video has only a single frame. This change is actually
pretty dumb, since audio frames are usually small enough that exact
seeking does not really add much. But it gets rid of some weird special
cases.
Internally, the most important change is that is_coverart and is_sparse
handling is merged. is_sparse was originally just a special case for
weird .ts streams that have the corresponding low-level flag set. The
idea is that they're pretty similar anyway, so this would reduce the
number of corner cases. But I'm not sure if this doesn't break the
original intended use case for it (I don't have a sample anyway).
This changes last-frame handling, and respects the duration of the last
frame only if audio is disabled. This is mostly "coincidental" due to
the need to make seeking past EOF trigger player exit, and is caused by
setting STATUS_EOF early. On the other hand, this might have been this
way before (see removed chunk close to it).
This tries to fix#7206 (hr-seeking past EOF does not stop playback)
again. Commit 57fbc9cd76 should have fixed this, but trying it again
(using that git revision), it often did not work. Whatever the fuck.
So add another dumb special case that will break within weeks. Note that
the check in handle_eof() had no effect, since execute_queued_seek() is
called later, which cancels EOF in the same case.
Hr-seek past the last frame instantly enters EOF, which means
handle_playback_time() will not set playback_pts to the video PTS (as
all video frames are skipped), which leads to the playback time being
taken from the last seek target. This results in confusing behavior,
especially since the seek time will be clipped to the file duration for
display, but not for further relative seeks.
Obviously, the time should be set to the last video frame, so use the
last video frame as fallback if both audio and video have ended. Also,
since the same problem exists with audio-only playback, add a fallback
for audio PTS too. We don't know which was the "last" fragment of media
played (to decide whether to use the audio or video PTS as the
fallback), but it doesn't matter since the maximum works.
This could lead to some undesired effects. In particular the audio PTS
is basically a bad guess, and is for example not clipped against --end.
(But the ridiculous way audio syncing and clamping currently works, I'm
not going to touch that shit unless I rewrite it completely.) The cover
art case is slightly broken: using --keep-open with keyframe seeks will
result in 0 as playback PTS (the video PTS). OK, who cares, it got late.
Also casually get rid of last_vo_pts, since that barely made any sense
at all.
Fixes: #7487
AOs can report audio underruns, but only ao_alsa and ao_sdl (???)
currently do so. If the AO was marked as not reporting it, the cache
state was used to determine whether playback was interrupted due to slow
input.
This caused problems in some cases, such as video with very low video
frame rate: when a new frame is displayed, a new frame has to be
decoded, and since there it's so much further into the file (long frame
durations), the cache gets into an underrun state for a short moment,
even though both audio and video are playing fine. Enlarging the audio
buffer didn't help.
Fix this by making all AOs report underruns. If the AO driver does not
report underruns, fall back to using the buffer state.
pull.c behavior is slightly changed. Pull AOs are normally intended to
be used by pseudo-realtime audio APIs that fetch an audio buffer from
the API user via callback. I think it makes no sense to consider a
buffer underflow not an underrun in any situation, since we return
silence to the reader. (OK, maybe the reader could check the return
value? But let's not go there as long as there's no implementation.)
Remove the flag from ao_sdl.c, since it just worked via the generic
mechanism. Make the redundant underrun message verbose only.
push.c seems to log a redundant underflow message when resuming (because
somehow ao_play_data() is called when there's still no new data in the
buffer). But since ao_alsa does its own underrun reporting, and I only
use ao_alsa, I don't really care.
Also in all my tests, there seemed to be a rather high delay until the
underflow was logged (with audio only). I have no idea why this happened
and didn't try to debug this, but there's probably something wrong
somewhere.
This commit may cause random regressions.
See: #7440
As requested I guess. It behaves quite similar to the --loop* options.
Not quite happy with the idea that 1) the option is mutated on each
operation (but at least it's consistent with --loop* and doesn't require
more properties), and 2) the ab-loop command will do nothing once all
loop iterations are done. As a concession, the OSD shows something about
"disabled".
Fixes: #7360
Due to asynchronicity, we generally can't guarantee that a video frame
matches up with other events such as playback time change exactly (since
decoding, presentation, and property update all happen at different
times). This is a complaint in the referenced bug report, where
screenshot filenames in each-frame screenshot did not use the correct
timestamp, and instead was lagging behind by 1 frame.
But in this case, synchronicity was already pretty much forced with wait
calls. The only problem was that the playback time was updated at a
later time, which results in the observed 1 frame lag. Fix this by
moving the place where the screenshot is triggered in this mode.
Normal screenshots may still have the old problem. There is no effort
made to guarantee the timestamps absolutely line up, same as with the
OSD. (If you want a guarantee, you need to use a video filter, such as
libavfilter's drawtext. These will obviously use the proper timestamp,
instead of going through the somewhat asynchronous property etc. system
in the player frontend.)
Fixes: #7433
This simply didn't set the direction flag in most situations, which
meant the timestamps used in the subtitle renderer were nonsense,
leading to invisible subtitles.
This works only for text subtitles that are cached in the ASS_Track
state. Reading new subtitles is broken because the demuxer layer has
trouble returning subtitle packets backwards, and I think for rendering
bitmap subtitles, the pruning direction would have to be adjusted. (Not
sure if reversing the timestamps before the subtitle renderer backend is
even the right thing to do. At least for sd_ass.c, it seems to make
sense, because it caches subtitles with "normal" timestamps.)
Dudemanguy
llyyr
Thomas Weißschuh
sfan5
Guido Cella
LaserEyess
Leo Izen
Avi Halachmi (:avih)
wm4
Guido Cella
Ricardo Garcia