--dscale= and --*scale-window= (i.e. an empty string) are respectively
valid settings for their options (and, in fact, the defaults).
This fixes the bug that it was impossible to reset e.g. tscale-window
back to the default "unset" setting after setting it once.
Credit goes to @CounterPillow for locating the cause of this bug.
For testing in VMs I guess?
This features a very broken hack that probably works. Though I didn't
test the packed format case. Again, the mismatch is essentially due to
big endian byte addresses decreasing as bit addresses increase, so you
can't represent a bit position in a byte stream with a single address,
which the mpv metadata does.
OSD is broken because repack.c doesn't support big endian. You'll have
to live with it.
The recent changes to the image format metadata broke big endian, and
that was intentional. Some things are inherent to little endian (like
the idea to coalesce bit and byte offsets into a single bit offset), and
they don't be fixed. But some obvious things can be fixed, such as
marking LE vs. BE formats the right way around on BE hosts.
The metadata is formally still in LE, except that if the LE/BE flag
matches the host endian, the host endian can be used when accessing
packed formats with bit shifts, or when computing byte aligned component
byte offsets. The former may work because formats with LE/BE variants
use the same bit offsets after byte swapping, the latter may work
because little endian is the natural concept for addressing memory. But
it will "subtly" fail to do the right thing in some cases, and code
using this can't know, so have fun.
Many things are broken, but this makes e.g. vo_gpu mostly work.
My general opinion about BE computers is that you should get a better
computer, you can get one for free from any garbage dump.
IMGFMT_RGB30 was added first; FFmpeg added AV_PIX_FMT_X2RGB10 later.
This is exactly the same, so treat them as such. For some reason,
libswscale still seems to output incompatible data - not sure what this
is about, but I'm not going to debug it.
Instead of applying this only to "regular" formats, do it with all
formats.
For some reason, some repackers still have their own endian code. These
could probably be removed, but whatever.
This change attempts to fix detection of how endian swapping is to be
performed. Details can be found in the code comments.
It should not change anything, other than fixing handling of the
X2RGB10BE ffmpeg pixel format. This format was detected incorrectly, and
the component location metadata was discarded due to an internal
consistency check. With this commit, it is handled correctly. At first I
thought the X2RGB10BE ffmpeg pixdesc metadata was wrong, but it appears
to be correct. The problem with this format is that it's the first
packed RGB format that requires bit shift to access, and where the
endian word size is larger than the (rounded up) component size, all
while pixdesc would "require" you to perform 3 memory accesses (instead
of 1), and the code tries to reverse this.
It appears that trying to use the pixdesc metadata is much, much more
work than just duplicating it in a saner form. To be fair, most problems
are with big endian, and the mpv internal format does not care much
about endian _hosts_.
The pull mode APIs were previously required to have thread-safe
ao_controls. However, locks were added in b83bdd1 for parity with push
mode. This introduced deadlocks in ao_wasapi.
Instead, only lock ao_control for the push mode APIs.
fixes#7787
See also #7832, #7811. We'll wait for feedback to see if those should
also be closed.
For cases in which the requirements of the GPU API prevent directly
uploading a texture with a given stride, we need to fix the stride
manually in host memory. This incurs an extra memcpy, but there's not
much we can do about it. (Even in `ra_gl` land, the driver will just
hide this memcpy from the user)
Note: This code could be done better. It could only copy as many texels
as needed, and it could pick a stride that's a multiple of
`gpu->limits.align_tex_xfer_stride` for better performance. Patches
welcome (tm)
Fixes#7759
Implementation copy/pasted from:
https://code.videolan.org/videolan/libplacebo/-/commit/f793fc0480f
This brings mpv's tone mapping more in line with industry standard
practices, for a hopefully more consistent result across the board.
Note that we ignore the black point adjustment of the tone mapping
entirely. In theory we could revisit this, if we ever make black point
compensation part of the mpv rendering pipeline.
This standard says we should use a value of 203 nits instead of 100 for
mapping between SDR and HDR.
Code copied from https://code.videolan.org/videolan/libplacebo/-/commit/9d9164773
In particular, that commit also includes a test case to make sure the
implementation doesn't break roundtrips.
Relevant to #4248 and #7357.
glslang accepted this, perhaps erronneously, but mesa does not. It seems
to be incorrect. A caveat is that this means *all* SSBOs are now
coherent, but since we only use SSBOs for peak detection, that's a
non-issue. (And besides, marking something as coherent when we don't
perform any synchronization commands on it should be a no-op anyway)
Fixes#7823
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
AOs which use the "push" API must set this field now. Actually, this was
sort of always required, but happened to work anyway. The future
intention is to use device_buffer as the pre-buffer amount, which has to
be available right before audio playback is started. "Pull" AOs really
need this too conceptually, just that the API is underspecified.
From what I can see, only ao_null did not do this yet.
Previously, device_buffer defaulted to 0 on pulse. This meant that
commit baa7b5c would always wait with a timeout of 0, leading to
high CPU usage for PulseAudio users.
By setting device_buffer to the number of samples per channel that
PulseAudio sets as its target, this commit fixes this behaviour.
since the title bar catches the mouse up and down events, the underlying
events view doesn't reset the isMoving state and no mouse movements are
signalled to the core. now we also reset the state in mouse up events
on the title bar.
Fixes#7807
The "screenshot window" command (ctrl+s by default) somehow broke video
colors with --gpu-api=vulkan --profile=gpu-hq when playback was paused.
I don't know the cause, but the rest of the code seems to imply
gl_video_reset_surfaces() needs to be called manually to flush some
caches, and it fixes the issue, so I assume there's no great mystery
here.
Commit 07b0c18 introduced some build breakages. Some breakages
were fixed on c1fc535 and a1adafe. This one is still remaining.
This commit fixes the following build error:
[153/521] Compiling video/out/vulkan/context_wayland.c
../video/out/vulkan/context_wayland.c:26:10: fatal error: video/out/wayland/presentation-time.h: No such file or directory
26 | #include "video/out/wayland/presentation-time.h"
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
compilation terminated.
Relevant to: #7802
The playback thread may obviously still fill the AO'S entire audio
buffer, which means it unset p->draining, which makes no sense and broke
ao_drain(). So just don't unset it here.
Not sure if this really fixes this, it was hard to reproduce. Regression
due to the recent changes. There are probably many more bugs like this.
Stupid asynchronous nightmare state machine. Give me a language that
supports formal verification (in presence of concurrency) or something.
I feel like this makes slightly more sense. At least it doesn't include
the potentially arbitrary constant latency that is generally included in
the delay value. Also, the buffer status doesn't matter - either we've
filled the entire buffer (then we can wait this long), or there's not
enough data anyway (then the core will wake up the thread if new data is
available).
But ultimately, we have to guess, unless the AO does notify us with
ao_wakeup_playthread().
Draining may now wait for no reason up to 1/4th of the total buffer
time. Shouldn't be a disimprovement in practice.
It's conceivable that ao->driver->reset() will make the audio API wait
for ao_read_data() (i.e. its audio callback) to return. Since we
recently moved the reset() call inside the same lock that ao_read_data()
acquires, this could deadlock. Whether this really happens depends on
how exactly the AO behaves. For example, ao_wasapi does not have this
problem. "Push" AOs are not affected either.
Fix by moving it outside of the lock. Assume ao->driver->start() will
not have this problem.
Could affect ao_sdl, ao_coreaudio (and similar rotten fruit AOs). I'm
unsure whether anyone experienced the problem in practice.
Instead of the relatively subtle underflow handling, simply signal
whether the stream is in a playing state. Should make it more robust.
Should affect ao_alsa and ao_pulse only (and ao_openal, but it's
broken).
For ao_pulse, I'm just guessing. How the hell do you query whether a
stream is playing? Who knows. Seems to work, judging from very
superficial testing.
Just a detail. If wrong (not unlikely because I'm just guessing my own
messy state machine), this will make the player freeze due to waiting
for something that never happens. Enjoy.
The feeder thread basically woke up the core and itself too often, and
caused some CPU overhead. This was caused by the recent buffer.c
changes.
For one, do not let ao_read_data() wake up the core, and instead rely on
the feeder thread's own buffer management. This is a bit strange, since
the change intended to unify the buffer management, but being more
consequent about it is better deferred to later, when the buffer
management changes again anyway. And also, the "more" condition in the
feeder thread seems outdated, or at least what made it make sense has
been destroyed, so do something that may or may not be better. In any
case, I'm still not getting underruns with ao_alsa, but the wakeup
hammering is gone.
Commit 6a13954d67 lowered the frequency of wakeups with this
condition. But it seems it sometimes the audio sync mode really should
get the wakeup before the frame is rendered. Normally, vo_thread is
supposed to perform this wakeup. Now the wakeup frequency is twice of
what should be needed - whatever, maybe it can be fixed properly once or
if timing is moved to the VO entirely in the future.
Fixes: #7777 (probably, untested)
This affects "pull" AOs only: ao_alsa, ao_pulse, ao_openal, ao_pcm,
ao_lavc. There are changes to the other AOs too, but that's only about
renaming ao_driver.resume to ao_driver.start.
ao_openal is broken because I didn't manage to fix it, so it exits with
an error message. If you want it, why don't _you_ put effort into it? I
see no reason to waste my own precious lifetime over this (I realize the
irony).
ao_alsa loses the poll() mechanism, but it was mostly broken and didn't
really do what it was supposed to. There doesn't seem to be anything in
the ALSA API to watch the playback status without polling (unless you
want to use raw UNIX signals).
No idea if ao_pulse is correct, or whether it's subtly broken now. There
is no documentation, so I can't tell what is correct, without reverse
engineering the whole project. I recommend using ALSA.
This was supposed to be just a simple fix, but somehow it expanded scope
like a train wreck. Very high chance of regressions, but probably only
for the AOs listed above. The rest you can figure out from reading the
diff.
wasapi/coreaudio/sdl were affected, alsa/pusle were not.
The confusion here was that resume() has different meaning with pull and
push AOs.
Fixes: #7772
This should make --term-title work in Windows 8.1 and below.
OSC sequences are defined in ECMA-48. The 'Change Window Title' command,
as far as I can tell, is a de-facto standard defined by xterm[1]. In
either case, this code is probably still not standards-compliant.
This also changes mp_write_console_ansi to convert to UTF-16 before
parsing control sequences, because that made it easier to pass the OSC
param to SetConsoleTitleW. I think it's also more correct to do it this
way, even though it doesn't really matter much for our limited terminal
parsing. As a side-effect of this, mp_write_console_ansi no longer
mutates its argument.
[1]: https://invisible-island.net/xterm/ctlseqs/ctlseqs.html#h3-Operating-System-Commands
Regression since the recent refactor. How did nobody notice?
This happened because the push code now calls the function for the pull
code. Both the former and latter apply the volume, so oops.
The recent change to the common code removed all calls to ->drain. It's
currently emulated via a timed sleep and polling ao_eof_reached(). That
is actually fallback code for AOs which lacked draining. I could just
readd the drain call, but it was a bad idea anyway. My plan to handle
this better is to require the AO to signal a underrun, even if
AOPLAY_FINAL_CHUNK is not set. Also reinstate not possibly waiting for
ao_lavc.c. ao_pcm.c did not have anything to handle this; whatever.
This simply printf()s a concatenation of the provided string and the
relevant escape sequences. No idea what exactly defines this escape
sequence (is it just a xterm thing that is now supported relatively
widely?), and this simply uses information provided on the linked github
issue.
Not much of an advantage over --term-status-msg, though at least this
can have a lower update frequency. Also I may consider setting a default
value, and then it shouldn't conflict with the status message.
Fixes: #1725
This is preparation to further cleanups (and eventually actual
improvements) of the audio output code.
AOs are split into two classes: pull and push. Pull AOs let an audio
callback of the native audio API read from a ring buffer. Push AOs
expose a function that works similar to write(), and for which we start
a "feeder" thread. It seems making this split was beneficial, because of
the different data flow, and emulating the one or other in the AOs
directly would have created code duplication (all the "pull" AOs had
their own ring buffer implementation before it was cleaned up).
Unfortunately, both types had completely separate implementations (in
pull.c and push.c). The idea was that little can be shared anyway. But
that's very annoying now, because I want to change the API between AO
and player.
This commit attempts to merge them. I've moved everything from push.c to
pull.c, the trivial entrypoints from ao.c to pull.c, and attempted to
reconcile the differences. It's a mess, but at least there's only one
ring buffer within the AO code now. Everything should work mostly the
same. Pull AOs now always copy the audio data under a lock; before this
commit, all ring buffer access was lock-free (except for the decoder
wakeup callback, which acquired a mutex). In theory, this is "bad", and
people obsessed with lock-free stuff will hate me, but in practice
probably won't matter. The planned change will probably remove this
copying-under-lock again, but who knows when this will happen.
One change for the push AOs now makes it drop audio, where before only a
warning was logged. This is only in case of AOs or drivers which exhibit
unexpected (and now unsupported) behavior.
This is a risky change. Although it's completely trivial conceptually,
there are too many special cases. In addition, I barely tested it, and
I've messed with it in a half-motivated state over a longer time, barely
making any progress, and finishing it under a rush when I already should
have been asleep. Most things seem to work, and I made superficial tests
with alsa, sdl, and encode mode. This should cover most things, but
there are a lot of tricky things that received no coverage. All this
text means you should be prepared to roll back to an older commit and
report your problem.
When the window is mapped, some ICCCM WM_HINTS are set.
The input field is set to true and state is set to NormalState.
To quote the spec, "The input field is used to communicate to the window
manager the input focus model used by the client" and "[c]lients with
the Passive and Locally Active models should set the input flag to True".
mpv falls under the Passive Input model, since it expects keyboard input,
but only listens for key events on its single, top-level window instead
of subordinate windows (Locally Active) or the root window (Globally
Active).
From the end users prospective, all EWMH/ICCCM compliant WMs (especially
the minimalistic ones) will allow the user to focus mpv, which will allow
mpv to receive key events. If the input field is not set, WMs are
allowed to assume that mpv doesn't require focus.
Commit 9d32d62b61 broke this when it changed OPT_TIME. I simply forgot
to adjust the command definition. The effect was that "no" was not
accepted as value.
