With the addition of fractional scaling support, wl->scaling was
converted to a double. Some compositors (Plasma) can report values under
1 for fractional scaling, so this meant wl->scaling could be some
small fractional value. This is fine except that when using the legacy
code for drawing the mouse cursor (i.e. not the cursor-shape protocol),
it still uses the old integer scaling method in core wayland. The reason
for this is simply because fractionally scaling the mouse cursor surface
is nonsense and nobody even has cursor images for anything besides a
select few sizes anyways (32x32, 48x48, etc.). The existing integer
scaling sort of works but it's pretty bad too and you can get some weird
sizes anyway. This is why cursor-shape is preferred since it fixes this.
Anyways, since buffer scaling for the cursor only takes integers, there
could be truncation to 0 in the previously mentioned fractional scale
this. This naturally causes the compositor to send us an error and mpv
quits. The fix is to always make sure that the scale value used for the
cursor is at least 1. Anything less makes no sense. Fixes#12309.
There was assumption in the code that default settings are compatible
with dumb mode and are only one that should be used in this case.
Force bilinear if dumb mode is enabled.
ac725764ec originally added these to
prevent those profiles from having auto as a value. However the auto
value was removed in 53d032374d later. So
having these lines here no longer serves any purpose since the default
value for slang is NULL once again (vlang/alang were never needed here).
These were deprecated a long time ago and apparently didn't even work
with lavfi filters. Go ahead and remove them and additionally clean up
some code related to them. m_config_from_obj_desc_and_args becomes much
simpler now and a couple of arguments can be completely removed.
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.
A bit different from the OPT_REPLACED/OPT_REMOVED ones in that the
options still possibly do something but they have a deprecation
message. Most of these are old and have no real usage. The only
potentially controversial ones are the removal of --oaffset and
--ovoffset which were deprecated years ago and seemingly have no real
replacement. There's a cryptic message about --audio-delay but who
knows. The less encoding mode code we have, the better so just chuck
it.
e9e93b4dbe added a warning about writing
the same value to the playlist-pos property that in the future it would
stop restarting playback. Instead, you should use the
playlist-play-index command for that. Well go ahead and drop the old
deprecated behavior now and do what wm4 wanted this to do: just ignore
if the same value is written again.
drop-frame-count and vo-drop-frame-count are ancient and have no reason
to exist anymore. The other change is that support for writing to
display-fps has been removed, and the property is strictly read-only
now. 3a2dc8b22e is what deprecated it with
a warning to users, so we can remove it without much trouble.
We've got an ungodly amount of OPT_REPLACED and OPT_REMOVED sitting
around in the code. This is harmless, but the vast majority of these are
ancient. 26f4f18c06 is the last commit
that touched the majority of these and of course that only changed how
options were declared so all of this stuff was deprecated even before
that. No use in keeping these, so just delete them all. As an aside,
there was actually a cocoa_opts but it had only a single option which
was replaced by something else and empty otherwise. So that entire thing
was just simply removed. OPT_REPLACED/OPT_REMOVED declarations that were
added in 0.35 or later were kept as is.
The osd support was originally written with the requirement that we have
actual frames getting delivered to the VO. This isn't always the case
though. If you force a window on a blank audio file for example, then
there will be no frame thus draw_frame did nothing. Since the previous
commit allows us to reliably detect this, we can rearrange the code
around a little bit to make this possible. A key change is to make the
osd_subsurface have wl->surface as the parent. This is seemingly
required otherwise the osd_surface buffers are never visible above the
empty video_surface when we have a black window. Also nuke the desync
call since it's completely pointless. Fixes#12429.
I didn't set file-local-options/hwdec because you have to store the
hwdec value to restore it after cropdetect anyway, and if the user
manually changes hwdec after cropdetect, the new value isn't reset when
changing file.
If you change file while cropdetect is active and you try to crop the
next video, it fails with "Already cropdetecting!". This is because
timers.detect_crop wasn't cleared correctly, only the timer variable
local to the loop was being set to nil.
The defaults were awful and horribly regressed many files while also not
fixing banding on files that actually needed it, sometimes even
*increasing* banding due to the low threshold.
Fixes: 12ffce0f22
See-Also: haasn/libplacebo@e1e43376d1
This probably makes `vo_gpu` tone mapping worse, or something, but who
cares. The status quo for a while now has been to use `vo_gpu_next` if
you care about HDR rendering at all.
See-Also: haasn/libplacebo@ec60dd156b
See-Also: haasn/libplacebo@0903cbd05d
This new filter is slightly sharper, and significantly faster, than
mitchell. It also tends to preserve detail better. All in all, there is
no reason not to use it by default, especially from a performance PoV.
(In vo_gpu_next, hermite is implemented efficiently using hardware
accelerated bilinear interpolation)
See-Also: 75b3947b2c
It can hurt people's feelings to refer to scalers as "high quality" and
"low quality", when it is so subjective.
I decided to preserve the lanczos sections at least because it's mostly
talking about the difference between EWA Lanczos and Lanczos, which is
less controversial than the difference between, say, Lanczos and
Catmull.
Avoid generating too much audio after EOF.
Note: This often has no effect, because less audio is produced than
required.
Usually this comes to effect with the userspeed filter at high speed
(4x) and going back to 1x speed to remove the filter.
After the final input packet, the filter padded with silence to allow
one more iteration. That was not enough to process the final frames.
Continue padding the end of `input_buffer` with silence until the final
frames have been processed.
Implementation: Instead of padding when adding final samples, pad before
running WSOLA iteration. Count number of added silent frames and
remaining input frames for time keeping.
This changes the emitted pts values from the start of the search block
to the center of the search block. Change initial `output_time`
accordingly. Initial `search_block_index` is irrelevant, because it's
overwritten before the first iteration.
Using the `output_time` removes the rounding of `search_block_index`,
which also fixes the <20 microsecond gaps in timestamps between output
packets.
Rationale:
The variance in audio position was in the range `0..search-interval`.
With this change, the range is
(- search-interval / 2)..(search-interval / 2)`
which ensures lower maximum offset.
Target block can be anywhere in the previous search-block, varying by
`search-interval` while the filter is active. This resulted in constant
audio offset when returning to 1x playback speed.
- Move the search block to the target block to sync up exactly.
- Drop old frames to minimize input_buffer usage.
The internal time update function involved multiple problems:
- Time was updated after WSOLA iteration. The means speed was updated
one iteration later than it could be.
- The update functions caused spikes of too many or too few samples
advanced, leading to audio glitches on speed changes.
- The inconsistent updates made it very difficult to produce gapless
audio packets.
- The `output_time` update function involved complicated feedback:
`search_block_index` influenced how many frames from `input_buffer`
are retained, which influenced how much `output_time` is changed,
which influenced `search_block_index`.
With these changes:
- Time is updated before WSOLA iterations. Speed changes are effective
instantly.
- There are no spikes in playback speed during speed changes.
- No significant gaps are introduced in output packets.
- The time update function becomes (function calls omitted for brevity)
output_time += ola_hop_size * playback_rate
Functions received a `playback_rate` parameter to check how many samples
are needed before iteration. Internal state is only updated when the
iteration is actually run, so the speed is allowed to change until
enough data is received.
The first WSOLA iteration overlapped audio with whatever was in the
`wsola_output` buffer. This was either silence (if not run before), or
old frames (if switching to 1x and back to a different speed).
Track the state of the output buffer and memcpy the whole window for the
first iteration instead.
