| Commit message (Collapse) | Author | Age | Files | Lines |
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It is now the AO's responsibility to handle period size alignment. The
ao->period_size alignment field is unused as of the recent audio
refactor commit. Remove it.
It turns out that ao_alsa shows extremely inefficient behavior as a
consequence of the removal of period size aligned writes in the
mentioned refactor commit. This is because it could get into a state
where it repeatedly wrote single samples (as small as 1 sample), and
starved the rest of the player as a consequence. Too bad. Explicitly
align the size in ao_alsa. Other AOs, which need this, should do the
same.
One reason why it broke so badly with ao_alsa was that it retried the
write() even if all reported space could be written. So stop doing that
too. Retry the write only if we somehow wrote less.
I'm not sure about ao_pulse.
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I guess the audio timestamp corruption problem is probably solved now.
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The code is OK, and it could be restored if it's needed again. But it is
unused now, so remove it.
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Unused, was terrible garbage. It was (or at least its implementation
was) always a make-shift solution, and just gross bullshit. It is unused
now, so delete it.
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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.
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fix lua5.1-5.2 support with luajit （mxe default upstream lua5.3 ）
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Shouldn't change the behavior if not used. Will probably be used in a
later commit.
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That's dumb.
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Do not make resetting the "access" filters reset the queue itself. This
is more flexible, and will be used in a later commit.
Also, if the queue is not in the reset state while the input access
filter is reset, make it immediately request data again. This is more
consistent, because it'll enter the state it "should" be, rather when
the filter's process function is called at an (essentially) random point
in the future. This means the filter graph will resume work on its own
if the queue was not reset before filter reset.
This could affect the only current user of f_async_queue, the code for
the --vd-queue-enable/--ad-queue-enable feature in f_decoder_wrapper.
But it looks like this already uses it in a compatible way.
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This is a kind of bad hack (with bad implementation) to paint over other
problems of the filter system. The main problem is that some filters
might be left with pending frames if the filter runner is "paused",
which we don't want. To be used in a later commit.
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This requires a slightly more recent libass than before
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For debugging.
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It's relevant in some obscure corner cases (EDL file that has a segment
without audio). Didn't test what's actually going on (is ad_lavc.c
behaving wrong? is libavcodec behaving wrong or in an unexpected way? is
lavc_process wrong?) and just patched it over with some bullshit, so the
fix might be too complicated, and could be reworked at some later point.
This sure is a real data flow fuckmess.
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Doesn't take paused_for_cache into account. For consistency; unlikely to
matter at all in practice.
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Not used yet; probably will, just dumping this to get it out of my
sight.
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Allow mp_aframe_clip_timestamps() to discard a spdif frame if it's
entirely out of the timestamp range. Just a minor thing that might make
handling these dumb formats easier.
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From what I can tell, this has been copy-pasted from times when
ao_coreaudio still used its own ringbuffer, instead of the common code.
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Such failure.
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Transparent windows on X11/EGL/native Mesa GL didn't work for various
reasons. From what I remember, the current code did work with nvidia at
least. Mesa has made attempts to fix this, but they never really made it
in.
But it turns out you can make EGL/Mesa list the EGLConfigs that use X11
RGBA visuals, and context_x11egl.c contains code that explicitly selects
them if alpha is requested (see pick_xrgba_config()).
The reason EGL/Mesa did not list them (and thus breaking transparency)
is because we requested a EGL_ALPHA_SIZE != 0 if alpha is requested. But
the transparent EGLConfigs use EGL_ALPHA_SIZE == 0. That's because EGL
doesn't actually support the concept of transparent windows; the alpha
size parameter is something else (memory rendering without FBOs or
something, I don't care enough to look up the real reasons).
This still won't work on Wayland. Every EGL backend needs platform
specific code. (Good job, EGL, such an awesome platform independent
standard.)
Fixes: #6590
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Previously get_state() would keep setting the cork status
while paused, but it only does for that after underflows now.
Correct this oversight by creating the stream corked for start()
to uncork it at a later time.
fixes #8026
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Oversight in b0f0be7. The user_sbc value would update but not last_sbc
if no presentation events were received. This would result in an
incorrect sbc_passed value (in practice, this should always be 1 since,
as far I know, all wayland compositors are currently only capable of
double buffering). When bring the window back into view, it would result
in a single frame of very high vsync jitter. Although in most cases it
was imperceptible, rarely I was able to completely break playback (i.e.
constant mistimed/dropped frames). Fix this by simply incrementing
last_sbc by 1 if the window is hidden. The buffer swap call did still
occur. The user just didn't see it.
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This reverts commit 41243e7c4f98b410195397b6758f9796acd9de57.
This fixes image format detection. FFmpeg has an utter called "image2",
which is designed to read patterns in filenames (so you can play
something like "%*.jpg" for all jpg files in the current directory).
"image2" is not what we want; it's just broken with custom I/O like
mpv uses it, and we don't want to "accidentally" interpret filenames
as pattern. That's why mpv blacklists it.
Unfortunately, "image2" is sometimes the format that FFmpeg's probe API
returns as best match. Thus demux_lavf fails to detect the file type,
and after some more futile attempts, we end up at demux_mf, which uses
detection by file extension. (Not sure why. I guess MPlayer did that,
and foudn that sufficient.) If the file extension is wrong (which
happens a lot because apparently the world is full of idiots who don't
manage to get the most simple things right), the image "loads", but
decoding obviously fails.
There's no easy way around this. The FFmpeg API has no mechanism to
exclude a specific format from probing (like image2, which breaks stuff
for us). Out of the 5 probe functions the API provides, none can probe
a specific format or include or exclude specific formats. The main
problem is that AVInputFormat.read_probe is a private symbol.
FFmpeg itself has no problem opening such files. It turns out that it
works, because even though image2 by itself uses detection by file
extension, it uses private API to further probe the exact format. It
explicitly excludes itself to prevent recursion.
But fortunately, that also means that it's impossible to get the image2
format if no filename is passed to the prober. (No filename, no file
extension.) Apparently we pass it in because it helps in corner cases.
Until almost 3 years ago, we passed the filename only when normal
probing already failed. Restore this by this revert. It makes
incorrectly named files work. The revert also makes the (apparently
forgotten) comment above the touched line of code true again.
Yes, quite possible that this breaks some mp3s again. You can't win
with FFmpeg. Thanks FFmpeg for making us fail at opening simple image
files and/or the most widely used file format for audio.
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Some absurd useless stuff.
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Why on earth did I ever bother with this dumb crap? If we do not have
any presentation statistics, just set the relevant vo_sync_info values
to -1 to disable it. It's much simpler than using mp deltas and trying
to keep up with mpv's clock. This also appears to fix audio/video
desynchronization if you start a video with the pause flag, move it out
of view, and then unpause it. Technically harmless since the video
wasn't even in view and putting back in view recovered it, but a quieter
terminal is better.
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Well, whatever. Only results in an error message being printed, because
there is no other error reporting mechanism, and the general policy is
to keep trying with the rest of the data (i.e. not report EOF).
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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.
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Accepting ":" in addition to "," seems confusing and dumb. It only
causing problems when you want to pass a value that contains ":". Remove
support for ":", it is now treated like any other normal character. This
affects all options that are listed as "Key/value list" in the option
list.
It's possible that this breaks for someone who happened to use ":" as
separator. But this was undocumented, and never recommended. Originally,
the option treated many other characters in a special way, but this was
changed in commit a3d561f950e74fe. I'm, not sure why ":" was explicitly
included. Maybe because -the absurd -vf/--af syntax uses ":" as list
separator. But "," was always recommended and used in examples for
key/value options.
Fixes: #8021 (if you consider it a bug)
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this fixes some race condition where the content of the layer wasn't
updated because the size didn't changed and the layer was already marked
as updated. just force an update on reconfig.
Fixes #7989
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this is preparation for new backends. currently this is done via the
libmpv API but for future new new VOs not based on libmpv we need these
VOCTRL events.
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remove the libmpv observer for the macOS specific options and use a
config cache + change callback for runtime changes. this is also a
preparation for new backends and generalises even more, since libmpv
functions can't and shouldn't be used in usual vo backends. for feature
parity the config cache is used.
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this was requested on an old issue, but the comment has since been
deleted. i though it was useful enough to add it. it's also just a one
line change.
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i don't know what i was thinking there, but force unwrapping is a very
bad idea.
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move all backend independent code parts in their own folder and files,
to simplify adding new backends. the goal is to only extend one class
and add the backend dependent parts there. usually only the (un)init,
config and related parts need to be implemented per backend. furthermore
all needed windowing and related events are propagated and can be
overwritten. the other backend dependent part is usually the surface for
rendering, for example the opengl oder metal layer.
in the best case a new backend can be added with only a few hundred
lines.
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The software rendering API makes libswscale directly write into supplied
user memory. As such, weird memory corruption bugs on non-optimal buffer
configurations are exposed to the user.
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It was possible for sway to get incorrectly sized borders if you resized
the mpv window in a creative manner (e.g. open a video in a non-floating
mode, set window scale to 2, then float it and witness wrong border
sizes). This is possibly a sway bug (Plasma doesn't have these border
issues at least), but there's a reasonable workaround for this.
The reason for the incorrect border size is because it is possible for
mpv to ignore the width/height from the toplevel listener and set its
own size. This new size can differ from what sway/wlroots believes the
size is which is what causes the sever side decorations to be drawn on
incorrect dimensions.
A simple trick is to just explicitly commit the surface after a resize
is performed. This is only done if mpv is not fullscreened or maximized
since we always obey the compositor widths/heights in those cases.
Sending the commit signals the compositor of the new change in the
surface and thus sway/wlroots updates its internal coordinates
appropriately and borders are no longer broken.
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This allows users to control whether full dialogue subtitles are displayed
with an audio track already in their preferred subtitle language.
Additionally, this improves handling for the forced flag, automatically
selecting between forced and unforced subtitle streams based on the user's
settings and the selected audio.
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The original goal was to simplify all this logic to make it less fragile
and breaky. Unfortunately, that didn't exactly happen and things might
actually be more complicated in some ways (well in other ways it's
simplier). There's a lot of negotiation back and forth between the
client and the compositor regarding sizes. The client (aka mpv) can do a
resize on its own. But also the compositor can request its own resize
(which we should be nice and listen to of course). The older method had
a lot of breakfalls/edgecases that were gradually patched up as time
went on, but that approach is really fragile. This refactor should,
hopefully, be on a more solid foundation.
Don't call any of the xdg toplevel state changing functions
(fullscreen, maximized, etc.) directly. Use the toggle wrapper
functions. These signal that the state was changed which is later
handled in the toplevel listener.
Introduce a new vdparams variable that stores the actual dimensions of
the video. This does create some new (but neccesary) complexity.
wl->vdparams stores what the actual dimensions of the video are
(according to mpv). wl->window_size stores the last size of the window
(so it includes any manual resizes for instance). wl->geometry is the
actual size of the output that gets displayed on the screen.
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Add support for reading a byte range from a stream via
the `slice://` protocol.
Syntax is `slice://start[-end]@URL` where end is a maximum
(read until end or eof).
Size suffixes support in `m_option` is reused so they can
be used with start/end.
This can be very useful with e.g. large MPEGTS streams with
corruption or time-stamp jumps or other issues in them.
Signed-off-by: Mohammad AlSaleh <CE.Mohammad.AlSaleh@gmail.com>
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Fixes #8014.
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We've had some serious issues with GNOME in the past, but since then
their compositor has undergone some major internal improvements. The
most severe one [1], random vsync spikes and mistimed frames, can no
longer be reproduced by the original author of the issue. There are some
minor UI-related things (lack of window decorations for instance since
there is no xdg-decoration support), but users don't seem to complain
about that too much and they aren't revelant to playback.
3.38 isn't out quite yet, but that should also fix playback issues when
on a multimonitor setup (the fix is in the master branch at the moment).
In terms of playback, the only real concerning issue is the lack of idle
inhibit so a warning is still displayed. But GNOME has their own
workaround that users can use for that so if anyone happens to complain,
we can just point them to that.
[1] https://gitlab.gnome.org/GNOME/mutter/-/issues/957
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When using presentation time, we have to be sure to update the ust when
no presentation events are received to make sure playback is still
smooth and in sync. Part of the recent presentation time refactor was to
use the presentation discarded event to signal that the window is
hidden. Evidently, this doesn't work the same everywhere for whatever
reason (drivers?? hardware??) and at least one user experienced issues
with playback getting out of sync since (presumably) the discarded event
didn't occur when hiding the window. Instead, let's just go back to the
old way of checking if the last_ust is equal to the ust value of the
last member in the wayland sync queue. Fixes #8010.
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The motivation for this change was a segfault caused by e107342 which
has complicated reasons for occuring (i.e. I'm not 100% sure but I think
it is a really weird race). The major part of this commit is moving the
initialization of presentation listener to the frame_callback function.
Calling it in swap_buffers worked fine but in practice it meant a lot of
meaningless function calls if a window was hidden (the presentation
would just be immediately discarded). By calling it in frame_callback,
we ensure the listener is only created when it is possible to receive a
presentation event.
Of course calling the presentation listener in feedback_presented or
feedback_discarded was considered, but ultimately these events are too
slow. Receiving the ust/msc/sbc triplet here and then passing it to mpv
results in higher vsync judder since there is (likely) not enough time
before the next pageflip. By design, the frame callback is meant to give
us as much time as possible before the next repaint so calling it here
is probably optimal.
Additionally, we can make better use of the feedback_discarded event.
The wp_presentation_feedback should not be destroyed here. It will be
taken care of either when we get feedback again or when the player
quits. Instead what we can do is set a bool that tells wayland_sync_swap
to update itself based on mp_time delta. In practice, the result is not
any different than before, but it should be more understandable what is
going on now.
Of course, the segfault mentioned at the beginning is fixed with this as
well.
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lua/js utils.get_env_list() uses `environ' which was ANSI, thus
it broke any unicode names/values.
mpv already has an internal utf8_environ for win32, but it's used
only at the getenv(..) wrapper and not exposed in itself, and also it
has lazy initialization - on first getenv() call.
Now `environ' maps to a function which ensures initialization while
keeping it an l-value (like posix expects).
The cost of this fuglyness is that files should include osdep/io.h
(which now declares environ as extern) rather than declaring it
themselves, or else the build will break on mingw.
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This was a vague idea how to handle passing byte arrays from Lua to the
mpv command interface in a somewhat reasonable way. The idea was
cancelled, but leave the Lua part of it, since it might get useful
later, and prevents passing (and silently cutting off) such byte
strings.
Barely tested, let's say not tested at all.
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