| Commit message (Collapse) | Author | Age | Files | Lines |
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Manual changes done:
* Merged the interface-changes under the already master'd changes.
* Moved the hwdec-related option changes to video/decode/vd_lavc.c.
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Fixes #5980
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Crashes NVIDIA, probably buggy on others. No one ever tests this shit.
See #2192
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Rather than the linear cd/m^2 units, these (relative) logarithmic units
lend themselves much better to actually detecting scene changes,
especially since the scene averaging was changed to also work
logarithmically.
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In theory our "eye adaptation" algorithm works in both ways, both
darkening bright scenes and brightening dark scenes. But I've always
just prevented the latter with a hard clamp, since I wanted to avoid
blowing up dark scenes into looking funny (and full of noise).
But allowing a tiny bit of over-exposure might be a good thing. I won't
change the default just yet (better let users test), but a moderate
value of 1.2 might be better than the current 1.0 limit. Needs testing
especially on dark scenes.
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The previous approach of using an FIR with tunable hard threshold for
scene changes had several problems:
- the FIR involved annoying hard-coded buffer sizes, high VRAM usage,
and the FIR sum was prone to numerical overflow which limited the
number of frames we could average over. We also totally redesign the
scene change detection.
- the hard scene change detection was prone to both false positives and
false negatives, each with their own (annoying) issues.
Scrap this entirely and switch to a dual approach of using a simple
single-pole IIR low pass filter to smooth out noise, while using a
softer scene change curve (with tunable low and high thresholds), based
on `smoothstep`. The IIR filter is extremely simple in its
implementation and has an arbitrarily user-tunable cutoff frequency,
while the smoothstep-based scene change curve provides a good, tunable
tradeoff between adaptation speed and stability - without exhibiting
either of the traditional issues associated with the hard cutoff.
Another way to think about the new options is that the "low threshold"
provides a margin of error within which we don't care about small
fluctuations in the scene (which will therefore be smoothed out by the
IIR filter).
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Instead of desaturating towards luma, we desaturate towards the
per-channel tone mapped version. This essentially proves a smooth
roll-off towards the "hollywood"-style (non-chromatic) tone mapping
algorithm, which works better for bright content, while continuing to
use the "linear" style (chromatic) tone mapping algorithm for primarily
in-gamut content.
We also split up the desaturation algorithm into strength and exponent,
which allows users to use less aggressive desaturation settings without
affecting the overall curve.
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Too many broken hardware decoders. Noticed wrong decoding of a video
file encoded with x262 on RX Vega when using VAAPI (Mesa 18.3.2).
Looks fine with swdec and a cheap hardware BD player.
Reverts 017f3d0674e48a587b9e6cd7a48f15519c799c3e
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List auto as an option for target-peak, and state that auto is its
default operation.
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Describe ReplayGain album-to-track fallback behavior introduced in
commits e392d6610d1e35cc0190c794c151211b0aae83e6 and
be90f2c8dd0431e252e43d5249e89446309113af.
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Adapt documentation accordingly and
also, fix an off-by-one check in the code.
closes #6371
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--record-file is nice, but only sometimes. If you watch some sort of
livestream which you want to record, it's actually much nicer not to
record what you're currently "seeing", but anything you're receiving.
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wm4-commits--merge-edition
This bumps libmpv version to 1.103
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The demuxer cache is the only cache now. Might need another change to
combat seeking failures in mp4 etc. The only bad thing is the loss of
cache-speed, which was sort of nice to have.
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This code shouldn't even exist in libavformat. If you still need it, you
can enable it via --demuxer-lavf-o.
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Often requested, trivial.
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Until now, stopping playback aborted the demuxer and I/O layer violently
by signaling mp_cancel (bound to libavformat's AVIOInterruptCB
mechanism). Change it to try closing them gracefully.
The main purpose is to silence those libavformat errors that happen when
you request termination. Most of libavformat barely cares about the
termination mechanism (AVIOInterruptCB), and essentially it's like the
network connection is abruptly severed, or file I/O suddenly returns I/O
errors. There were issues with dumb TLS warnings, parsers complaining
about incomplete data, and some special protocols that require server
communication to gracefully disconnect.
We still want to abort it forcefully if it refuses to terminate on its
own, so a timeout is required. Users can set the timeout to 0, which
should give them the old behavior.
This also removes the old mechanism that treats certain commands (like
"quit") specially, and tries to terminate the demuxers even if the core
is currently frozen. This is for situations where the core synchronized
to the demuxer or stream layer while network is unresponsive. This in
turn can only happen due to the "program" or "cache-size" properties in
the current code (see one of the previous commits). Also, the old
mechanism doesn't fit particularly well with the new one. We wouldn't
want to abort playback immediately on a "quit" command - the new code is
all about giving it a chance to end it gracefully. We'd need some sort
of watchdog thread or something equally complicated to handle this. So
just remove it.
The change in osd.c is to prevent that it clears the status line while
waiting for termination. The normal status line code doesn't output
anything useful at this point, and the code path taken clears it, both
of which is an annoying behavior change, so just let it show the old
one.
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Be a bit more detailed, and discourage disabling it.
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This option has been deprecated upstream for a long time, probably
doesn't even work anymore, and won't work moving forwards as we replace
the vulkan code by libplacebo wrappers.
I haven't removed the option completely yet since in theory we could
still add support for e.g. a native glslang wrapper in the future. But
most likely the future of this code is deletion.
As an aside, fix an issue where the man page didn't mention d3d11.
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Extra line prevents the sub-title formatting.
Removing it, the option is formatted like the others.
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Despite their place in the tree, hwdecs can be loaded and used just
fine by the vulkan GPU backend.
In this change we add Vulkan interop support to the cuda/nvdec hwdec.
The overall process is mostly straight forward, so the main observation
here is that I had to implement it using an intermediate Vulkan buffer
because the direct VkImage usage is blocked by a bug in the nvidia
driver. When that gets fixed, I will revist this.
Nevertheless, the intermediate buffer copy is very cheap as it's all
device memory from start to finish. Overall CPU utilisiation is pretty
much the same as with the OpenGL GPU backend.
Note that we cannot use a single intermediate buffer - rather there
is a pool of them. This is done because the cuda memcpys are not
explicitly synchronised with the texture uploads.
In the basic case, this doesn't matter because the hwdec is not
asked to map and copy the next frame until after the previous one
is rendered. In the interpolation case, we need extra future frames
available immediately, so we'll be asked to map/copy those frames
and vulkan will be asked to render them. So far, harmless right? No.
All the vulkan rendering, including the upload steps, are batched
together and end up running very asynchronously from the CUDA copies.
The end result is that all the copies happen one after another, and
only then do the uploads happen, which means all textures are uploaded
the same, final, frame data. Whoops. Unsurprisingly this results in
the jerky motion because every 3/4 frames are identical.
The buffer pool ensures that we do not overwrite a buffer that is
still waiting to be uploaded. The ra_buf_pool implementation
automatically checks if existing buffers are available for use and
only creates a new one if it really has to. It's hard to say for sure
what the maximum number of buffers might be but we believe it won't
be so large as to make this strategy unusable. The highest I've seen
is 12 when using interpolation with tscale=bicubic.
A future optimisation here is to synchronise the CUDA copies with
respect to the vulkan uploads. This can be done with shared semaphores
that would ensure the copy of the second frames only happens after the
upload of the first frame, and so on. This isn't trivial to implement
as I'd have to first adjust the hwdec code to use asynchronous cuda;
without that, there's no way to use the semaphore for synchronisation.
This should result in fewer intermediate buffers being required.
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Since linear downscaling makes sense to handle independently from
linear/sigmoid upscaling, we split this option up. Now,
linear-downscaling is its own option that only controls linearization
when downscaling and nothing more. Likewise, linear-upscaling /
sigmoid-upscaling are two mutually exclusive options (the latter
overriding the former) that apply only to upscaling and no longer
implicitly enable linear light downscaling as well.
The old behavior was very confusing, as evidenced by issues such
as #6213. The current behavior should make much more sense, and only
minimally breaks backwards compatibility (since using linear-scaling
directly was very uncommon - most users got this for free as part of
gpu-hq and relied only on that).
Closes #6213.
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by default the pixel format creation falls back to software renderer
when everything fails. this is mostly needed for VMs. additionally one
can directly request an sw renderer or exclude it entirely.
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According to earlier discussions, this can improve visual quality.
This only changes the preferred order of the formats, not the
formats themselves.
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This code shouldn't even exist in libavformat. If you still need it, you
can enable it via --demuxer-lavf-o.
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Often requested, trivial.
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Be a bit more detailed, and discourage disabling it.
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With the advent of actual HDR devices, my real measured ICC profile has
an "infinite" contrast, since the display is completely off on pure
black inputs. 100k:1 might not be enough, so let's just bump it up to
1m:1 to be safe.
Also, improve the logging in the case that the detected contrast is too
high by default.
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This logic has been changed throughout the years, notably in 38ac5d5 and
3bdbf6. Update the documentation to reflect the current state.
Closes #5834.
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The manpage parts were forgotten when removing the options.
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This seems to work surprisingly well, and it's enabled by default
(unlike the old text claims).
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Instead of using an internal counter to keep track of the value that was
set last, attempt to find the current value of the property/option in
the value list, and then set the next value in the list.
There are some potential problems. If a property refuses to accept a
specific value, the cycle-values command will fail, and start from the
same position again. It can't know that it's supposed to skip the next
value. The same can happen to properties which behave "strangely", such
as the "aspect" property, which will return the current aspect if you
write "-1" to it. As a consequence, cycle-values can appear to get
"stuck".
I still think the new behavior is what users expect more, and which is
generally more useful. We won't restore the ability to get the old
behavior, unless we decide to revert this commit entirely.
Fixes #5772, and hopefully other complaints.
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I had this enabled for quite a while and experienced no issues. I'm not
aware of other issues either.
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Until recently, the AO was reinitialized strictly only on decoder format
changes. But the commit for simplifying audio format negotiation removed
this. Now the AO is recreated for any format change.
This is sort of annoying if you change playback speed. The
insertion/removal of af_scaletempo can change the sample format. For
example, the acompressor filter will convert output to double, so
toggling scaletempo will force the format back to float. This recreates
the AO under the --gapless-audio=weak default. This likely affects a lot
of other filters too.
Work this around by allowing sample format changes, and keeping the
current AO format in these cases. This is probably not a big problem.
Most audio APIs force the output format to float anyway.
This means you actually have to worry about what the default gapless
mode does to your audio. If you start with a file that uses 8 bit per
sample, and then continue playing a 24 bit FLAC, it will be converted
down to 8 bit per sample. (Assuming they are played in a way that uses
the gapless logic.)
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This was left out of e3e2c79 by mistake.
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See #5670.
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Fixes #5640.
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Probably mostly useful for the libmpv render API.
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The playback start logic explicitly waits until the first frame has been
displayed. Usually this will introduce a wait of 1 vsync. For normal
playback this doesn't matter, but with respect to low latency needs,
this only leads to additional data getting queued up in the demuxer or
network buffers.
Another thin |
