| Commit message (Collapse) | Author | Age | Files | Lines |
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The justification for this is the fact that the `video-aspect` property
doesn't work well with `cycle_values` commands that include the value
"-1".
The "video-aspect" property has effectively no change in behavior, but
we may want to make it read-only in the future. I think it's probably
fine to leave as-is, though.
Fixes #6068.
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There's potential confusion about how long a process started with the
"subprocess" command is allowed to live. Add some more explanations
regarding "subprocess" specifics (such as the playback_only field), and
things that apply to asynchronous commands in general.
Partially for #7025.
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Not the prettiest way to get it done, but seems to work.
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This lets the user define an adapter name that can then be passed
further into the internals.
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In preparation for making vo_drm able to use swapchain-depth
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It turns out you can actually bind keybind to another keybind. Just be
careful with all the quotes.
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this will request a 16bit half-float framebuffer instead if a 8bit
integer framebuffer.
Fixes #3613
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libass had an API to configure this since 2013. mpv always used
ASS_FONTPROVIDER_AUTODETECT, because usually there's little reason to
use anything else. The intention of the now added option is to allow
users to disable use of system fonts.
I didn't consider it worth the trouble to add the coretext and
directwrite enum items from ASS_DefaultFontProvider. The "auto" choice
will have the same effect if they're available. Also, the part of the
code which defines the option does not necessarily have libass available
(it's still optional!), so defining all enum items as choices is icky. I
still added fontconfig, since that may be nice to emulate a nostalgic
2010 feeling of mpv freezing on fontconfig.
The option for OSD is even less useful. (But you get it for free, and
why pass up a chance to add yet another useless option?)
This is not quite what was requested in #6947, but as close as it gets.
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We default to EGL instead of GLX now, which means vdpau only works
if we explicitly specify that we want a GLX context, as vdpau lacks
interop for EGL.
Update the hwdec documentation to reflect this.
Concerns #6980.
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The question came up on how a client would figure out where
screenshot-directory saved its screenshots if it contained
mpv-specific expansions. This command should remedy the situation
by providing a way for the client to ask mpv to do an expansion.
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These properties contain the current subtitle's start and end times.
Can be useful to cut sample audio through the scripting interface.
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This flag makes mpv continue using the PulseAudio driver even if the
sink is suspended.
This can be useful if JACK is running with PulseAudio in bridge mode and
the sink-input assigned to mpv is the one JACK controls, thus being
suspended.
By forcing mpv to still use PulseAudio in this case, the user can now
adjust the sink to an unsuspended one.
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Replace the "+" with "/". The "+" was supposed to imply that the cache
is the sum of the time (demuxer cache) and the size in bytes (stream
cache). We could not provide something nicer, because we had no idea how
many seconds of media was buffered in the stream cache.
Now the stream cache is done, and both the duration and byte size show
the amount buffered in the demuxer cache. Hopefully "/" is better to
imply this properly. Update the manpage explanations too.
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skip-logo.lua is just what I wanted to have. Explanations are on the top
of that file. As usual, all documentation threatens to remove this stuff
all the time, since this stuff is just for me, and unlike a normal user
I can afford the luxuary of hacking the shit directly into the player.
vf_fingerprint is needed to support this script. It needs to scale down
video frames as part of its operation. For that, it uses zimg. zimg is
much faster than libswscale and generates more correct output. (The
filter includes a runtime fallback, but it doesn't even work because
libswscale fucks up and can't do YUV->Gray with range adjustment.)
Note on the algorithm: seems almost too simple, but was suggested to me.
It seems to be pretty effective, although long time experience with
false positives is missing. At first I wanted to use dHash [1][2], which
is also pretty simple and effective, but might actually be worse than
the implemented mechanism. dHash has the advantage that the fingerprint
is smaller. But exact matching is too unreliable, and you'd still need
to determine the number of different bits for fuzzier comparison. So
there wasn't really a reason to use it.
[1] https://pypi.org/project/dhash/
[2] http://www.hackerfactor.com/blog/index.php?/archives/529-Kind-of-Like-That.html
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It's inadequate for most uses. There are better mechanisms.
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Helper for the ab-loop-dump-cache command, see manpage additions.
This is kind of shit. Not only is this a very "special" feature, but it
also vomits more messy code into the big and already bloated demux.c,
and the implementation is sort of duplicated with the dump-cache code.
(Except it's different.) In addition, the results sort of depend what a
video player would do with the dump-cache output, or what the user wants
(for example, a user might be more interested in the range of output
audio, instead of the video).
But hey, I don't actually need to justify it. I'm only justifying it for
fun.
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But don't tell the reader which those APIs are. Hope the user will just
search for "async" in the Lua section (lua.rst). But of course, nobody
will ever care about anything related to this.
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That's right, and it's probably not the end of it. I'll just claim that
I have no idea how to create a proper user interface for this, so I'm
creating multiple partially-orthogonal, of which some may work better in
each of its special use cases.
Until now, there was --record-file. You get relatively good control
about what is muxed, and it can use the cache. But it sucks that it's
bound to playback. If you pause while it's set, muxing stops. If you
seek while it's set, the output will be sort-of trashed, and that's by
design.
Then --stream-record was added. This is a bit better (especially for
live streams), but you can't really control well when muxing stops or
ends. In particular, it can't use the cache (it just dumps whatever the
underlying demuxer returns).
Today, the idea is that the user should just be able to select a time
range to dump to a file, and it should not affected by the user seeking
around in the cache. In addition, the stream may still be running, so
there's some need to continue dumping, even if it's redundant to
--stream-record.
One notable thing is that it uses the async command shit. Not sure
whether this is a good idea. Maybe not, but whatever. Also, a user can
always use the "async" prefix to pretend it doesn't.
Much of this was barely tested (especially the reinterleaving crap),
let's just hope it mostly works. I'm sure you can tolerate the one or
other crash?
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Until now, the following could happen: if you set a 1GB forward cache,
and a 1GB backward cache, and you opened a 2GB file, it would prune away
the data cached at the start as playback progressed past the 50% mark.
With this commit, nothing gets pruned, because the total memory usage
will still be 2GB, which equals the total allowed memory usage of 1GB +
1GB.
There are no explicit buffers (every packet is malloc'ed and put into a
linked list), so it all comes down to buffer size computations. Both
reader and prune code use these sizes to decide whether a new packet
should be read / an old packet discarded. So just add the remaining free
"space" from the forward buffer to the available backward buffer. Still
respect if the back buffer is set to 0 (e.g. unseekable cache where it
doesn't make sense to keep old packets).
We need to make sure that the forward buffer can always append, as long
as the forward buffer doesn't exceed the set size, even if the back
buffer "borrows" free space from it. For this reason, always keep 1 byte
free, which is enough to allow it to read a new packet. Also, it's now
necessary to call pruning when adding a packet, to get back "borrowed"
space that may need to be free'd up after a packet has been added.
I refrained from doing the same for forward caching (making forward
cache use unused backward cache). This would work, but has a
disadvantage. Assume playback starts paused. Demuxing will stop once the
total allowed low total cache size is reached. When unpausing, the
forward buffer will slowly move to the back buffer. That alone will not
change the total buffer size, so demuxing remains stopped. Playback
would need to pass over data of the size of the back buffer until
demuxing resume; consider this unacceptable. Live playback would break
(or rather, would not resume in unintuitive ways), even normal streaming
may break if the server invalidates the URL due to inactivity. As an
alternative implementation, you could prune the back buffer immediately,
so the forward buffer can grow, but then the back buffer would never
grow. Also makes no sense.
As far as the user interface is concerned, the idea is that the limits
on their own aren't really meaningful, the purpose is merely to vaguely
restrict the cache memory usage. There could be just a single option to
set the total allowed memory usage, but the separate backward cache
controls the default ratio of backward/forward cache sizes. From that
perspective, it doesn't matter if the backward cache uses more of the
total buffer than assigned, if the forward buffer is complete.
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Make most of the demuxer options runtime-changeable. This includes the
cache options and stream recording. The manpage documents some of the
possibly weird issues related to this.
In particular, the disk cache isn't shuffled around if the setting
changes at runtime.
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I once created this because someone wanted to use vapoursynth without
the Python dependency. No idea if anyone ever actually used it. It's
sort of icky (it calls itself "lazy" to preempt complaints about how
much it sucks), and complicates the build process. Kill it.
It seems much more promising to have something like this:
https://github.com/vapoursynth/vapoursynth/issues/386
This would either solve the build distribution problem by relaxing the
Python dependency, and/or allow a Lua backend to be included without
pain.
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This filter wasn't referenced anywhere and thus was dead code. It should
have been in the audio filter list in user_filters.c. This was intended
as compatibility wrapper (to avoid breaking old command lines and config
files), and has no real use. Apparently I forgot to add it to the filter
list (did I even test this shit?), and so it was rotting around for 1.5
years doing nothing (just like myself).
Note that users can just use the libavfilter provided filter to force
resampling, just that it has a different name and different options.
There's also af_format to force inserting auto conversion through the
internal f_swsresample filter.
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Useless at this point, I don't even know if it still works, or how to
test it.
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Normally I use the OSC like this: not at all, but have a key binding
that does "cycle osc" to show it. And in that case, I don't really want
it to overlap the damn video.
I could use the zoom/pan options to move the video out of the way, but
this is also sort of annoying. Likewise, you could write a script or so
which does this automatically if the OSC appears, but that's still
annoying, and computing values for these options such that the video is
moved correctly is tricky.
So I added a bunch of options that set explicit video borders (previous
commit), and a option for the OSC to use them (this commit).
Disabled by default, since I'm afraid this is too awkward and
unpolished, especially with OSC default settings.
I'm also using "osc-visibility=always". Effectively, making the OSC
appear will box the video, and making it disappear (by unloading
osc.lua) will restore the video back to normal.
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Semantics a bit questionable. This is done for the OSC (next commit),
and a comment added the manpage explicitly states this. Meaning this is
probably garbage and needs to revisit when the OSC changes and/or
someone wants to use this margin feature for something else.
Not sure about the subtitle thing. It's imaginable that someone uses
these options to create empty borders for subtitles on the bottom, so
subtitles should be located there. On the other hand, this gives a
rather unpolished user experience when using the (later added) OSC
feature to not overlap with the video. There's not much of a point if
the OSC still overlaps the video. However, I'm too lazy to think about
this, so it stays like it is.
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Somewhat similar to the old --cache-file, except for the demuxer cache.
Instead of keeping packet data in memory, it's written to disk and read
back when needed.
The idea is to reduce main memory usage, while allowing fast seeking in
large cached network streams (especially live streams). Keeping the
packet metadata on disk would be rather hard (would use mmap or so, or
rewrite the entire demux.c packet queue handling), and since it's
relatively small, just keep it in memory.
Also for simplicity, the disk cache is append-only. If you're watching
really long livestreams, and need pruning, you're probably out of luck.
This still could be improved by trying to free unused blocks with
fallocate(), but since we're writing multiple streams in an interleaved
manner, this is slightly hard.
Some rather gross ugliness in packet.h: we want to store the file
position of the cached data somewhere, but on 32 bit architectures, we
don't have any usable 64 bit members for this, just the buf/len fields,
which add up to 64 bit - so the shitty union aliases this memory.
Error paths untested. Side data (the complicated part of trying to
serialize ffmpeg packets) untested.
Stream recording had to be adjusted. Some minor details change due to
this, but probably nothing important.
The change in attempt_range_joining() is because packets in cache
have no valid len field. It was a useful check (heuristically
finding broken cases), but not a necessary one.
Various other approaches were tried. It would be interesting to list
them and to mention the pros and cons, but I don't feel like it.
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Some OGG web radio streams use timestamp resets when a new song starts
(you can find those Xiph's directory - other streams there don't show
this behavior). Basically, the OGG stream behaves like concatenated OGG
files, and "of course" the timestamps will start at 0 again when the
song changes. This is very inconvenient, and breaks the seekable demuxer
cache. In fact, any kind of seeking will break
This is more time wasted in Xiph's bullshit. No, having timestamp resets
by design is not reasonable, and fuck you. I much prefer the awful
ICY/mp3 streaming mess, even if that's lower quality and awful. Maybe it
wouldn't be so bad if libavformat could tell us WHERE THE FUCK THE RESET
HAPPENS. But it doesn't, and the randomly changing timestamps is the
only thing we get from its API.
At this point, demux_lavf.c is like 90% hacks. But well, if libavformat
applies this strange mixture of being clever for us vs. giving us
unfiltered garbage (while pretending it abstracts everything, and hiding
_useful_ implementation/low level details), not much we can do.
This timestamp linearizing would, in general, probably be better done
after the decoder, because then we wouldn't need to deal with timestamp
resets. But the main purpose of this change is to fix seeking within the
demuxer cache, so we have to do it on the lowest level.
This can probably be applied to other containers and video streams too.
But that is untested. Some further caveats are explained in the manpage.
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Until now, this usually passed a single audio frame to the decoder, and
then did a backstep operation (cache seek + frame search) again. This is
probably not very efficient, especially considering it has to search the
packet queue from the "start" every time again.
Also, with most audio codecs, an additional "preroll" frame was passed
first. In these cases, the preroll frame would make up 50% of audio
decoding time. Also not very efficient.
Attempt to fix this by returning multiple frames at once. This reduces
the number of backstep operations and the ratio the preoll frames. In
theory, this should help efficiency. I didn't test it though, why would
I do this? It's just a pain. Set it to unscientific 10 frames.
(Actually, these are 10 keyframes, so it's much more for codecs like
TrueHD. But I don't care about TrueHD.)
This commit changes some other implementation details. Since we can
return more than 1 non-preroll keyframe to the decoder, some new state
is needed to remember how much. The resume packet search is adjusted to
find N ("total") keyframe packets in general, not just preroll frames.
I'm removing the special case for 1 preroll packet; audio used this, but
doesn't anymore, and it's premature optimization anyway.
Expose the new mechanism with 2 new options. They're almost completely
pointless, since nobody will try them, and if they do, they won't
understand what these options truly do. And if they actually do, they
most likely would be capable of editing the source code, and we could
just hardcode the parameters. Just so you know that I know that the
added options are pointless.
The following two things are truly unrelated to this commit, and more
like general refactoring, but fortunately nobody can stop me.
Don't set back_seek_pos in dequeue_packet() anymore. This was sort of
pointless, since it was set in find_backward_restart_pos() anyway (using
some of the same packets). The latter function tries to restrict this to
the first keyframe range though, which is an optimization that in theory
might break with broken files (duh), but in these cases a lot of other
things would be broken anyway.
Don't set back_restart_* in dequeue_packet(). I think this is an
artifact of the old restart code (cf. ad9e473c555). It can be done
directly in find_backward_restart_pos() now. Although this adds another
shitty packet search loop, I prefer this, because clearer what's
actually happening.
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Surprised this didn't exist before.
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Before this commit, there was a single process_decoded_frame() function.
It handled various aspects of dealing with a newly decoded frame. Move
some of these to a separate process_output_frame() function.
This new function is called in the order the frames are returned to the
playback core. Some correct_audio_pts() (was process_audio_frame())
becomes slightly less awkward due to this, and the timestamp smoothing
can actually work in backward playback mode now (thus moving p->pts out
of reset_decoder()).
Behavior for normal playback also changes subtly. This shouldn't matter
in sane cases, but if you mix broken files, --no-correct-pts, and
timeline stuff, differences in behavior might be visible.
Timeline clipping (EDL/ordered chapters) works now, because it's done
before "transforming" the timestamps. Audio timestamp smoothing happens
after it, which is a behavior change, but should be more correct. This
still runs crazy_video_pts_stuff() before everything else. On the pther
hand, --no-correct-pts or missing timestamp processing is done last. But
these things didn't really work with timeline before.
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And add simpler aliases for the modes.
I'm not sure how to name things, and the option list is in general full
of different conventions. Some names are shortened, some are explicit
and long.
I guess options that have a chance to be used normally (i.e. not obscure
tuning or debugging) should have a short and convenient names.
In this specific case, play-direction is like a mixture of both. It
should be either playback-direction or play-dir, not shorten one word
but not the other.
The convenience aliases are because I got sick of typing out "backward".
I guess "back" would also do it, but there's no proper antonym (and
maybe it's "wrong" in the strict sense of the word).
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Together with the previous commit, this seems to make backward playback
work in files with vorbis and mp3 audio codecs.
For Vorbis (with libavcodec's decoder, didn't test libvorbis), the first
packet was just always completely discarded. This happened even though
we tell libavcodec that we do discarding of padding manually. It simply
happened inside the codec, not libavcodec's general initial padding
handling. In addition, the first output decoded frame seems to contain
partial data. (Unlike the opus decoder, it doesn't report any padding at
all.)
The Opus decoder (again libavcodec only tested) reports an initial
padding, but it appears to be too small, and it sounds right only with 2
packets discarded. So its status doesn't change.
I'm not sure why I need 2 frames for mp3, but with that value I had
success on the samples I tested.
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