Apple has decided that Mac OS X is now named macOS for the time
being. For consistency, it makes sense to use the same name for the
operating system in all places where it occurs. This commit renames
OS X to macOS in the documentation in places where it was otherwise
still using the old name.
currently we use the whole screen rectangle to calculate the window
geometry. this doesn't take the menu bar or the Dock into account.
by default use the visible screen rectangle instead. this is also a
change in behaviour, since the window can't be placed outside of this
rectangle anymore. also add an option to change to the old behaviour,
because it can still be useful in certain cases, like placing the window
directly underneath the menu bar when used a desktop background.
Fixes#8272
--alpha=yes doesn't affect only transparent videos and images, but also
the background. I spent time researching how to implement transparent
backgrounds and had no idea that they already worked at least on
Wayland.
Background transparency will work on X11 when
https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/2376 is
merged. It doesn't work on Windows. No idea about macOS. Either way,
this paragraph already says that it only works on certain environments
twice.
References #6590
With the current wording, I thought that playlist-prefetch doesn't
prefetch the next url within a playlist of m3u8 urls, but it makes a big
difference, and I would have enabled it earlier if it wasn't for this
paragraph.
This makes it clear that you can prefetch any file, but that it won't
prefill the cache with the video data. This is true for any video,
there's nothing unique about HLS streams in the behavior of this option.
Simple groundwork for adding a couple of user options that allow
selecting the screen with a string name. The next two commits implements
these options for xorg and wayland.
This reverts commit 3d17e19c2c.
The effect of turning off this setting is that mpv doesn't tell libass what
the video stream's resolution is. This happens to result in some files having
their transforms scaled in ways that give higher performance (as described
in #7435) because libass happened to guess a video resolution that resulted
in transforms yielding smaller bitmaps, but it's just as easy for the opposite
to happen depending on the resolutions and effects involved.
The option's name is also somewhat misleading: setting the storage size affects
blur, but it also affects stroke (which is far more important for the vast
majority of scripts) and 3D transforms (which look very screwy when done wrong).
Recent versions of mpv have applied security checks to --playlist
that previously only existed if playlist files were played as an
input directly. This commit documents this change and how to work
around it, in the event that playlist files are trusted.
Pointless feature that can be done with environment variables. It was
also implemented incorrectly and broke autoprobing.
This reverts commit 015b676875.
As per the client API, a client can connect to any arbitrary wayland
socket. mpv has always just passed NULL which connected to the
compositor currently in use, but one could just as easily pass the name
of a different socket (i.e. the value of WAYLAND_DISPLAY). Here, we just
expose this argument as a user configurable option. If the user passes a
socket name that does not exist, then print a warning and fall back to
NULL.
Picks up files like "cover.jpg". It's made part of normal external file
loading, so I'm adding 3 new options that are direct equivalents for the
options that control loading of external subtitle and audio files. Even
though I bet nobody wants them and they just increase confusion... I
guess the world is actually hell, so this outcome should be fine.
It prefers non-specific external files like "cover.jpg" over embedded
cover art. Not sure if that's wanted or unwanted.
There's some pain over explicitly marking such files as external
pictures. This is basically an optimization: in most cases, a heuristic
would treat an image file loaded with --external-file the same (it's a
heuristic because ffmpeg can't tell us whether something is an image or
a video). However, even with this heuristic, it would decode the cover
art picture again on each seek, which would essentially slow down
seeking in audio files. This bothered me greatly, which is why I'm
adding these additional options at all, and bothered with the previous
commit.
Fixes: #3056
on macOS 10.15 setting the activation policy behaves quite weirdly. the
call changes the current active App to a nameless process, which
probably also the reason that prevents the not focusing to work.
a workaround for that, is to refocus the previous active app.
Fixes#7725
--audio-stream-silence is a shitty feature compensating for awful
consumer garbage, that mutes PCM at first to check whether it's
compressed audio, using formats advocated and owned by malicious patent
troll companies (who spend more money on their lawyers than paying any
technicians), wrapped in a wasteful way to make it constant bitrate
using a standard whose text is not freely available, and only rude users
want it. This feature has been carelessly broken, because it's
complicated and stupid. What would Jesus do? If not getting an aneurysm,
or pushing over tables with expensive A/V receivers on top of them, he'd
probably fix the feature. So let's take inspiration from Jesus Christ
himself, and do something as dumb as wasting some of our limited
lifetime on this incredibly stupid fucking shit.
This is tricky, because state changes like end-of-audio are supposed to
be driven by the AO driver, while playing silence precludes this. But it
seems code paths for "untimed" AOs can be reused.
But there are still problems. For example, underruns will just happen
normally (and stop audio streaming), because we don't have a separate
heuristic to check whether the buffer is "low enough" (as a consequence
of a network stall, but before the audio output itself underruns).
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
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.
Seems like this is requested all the time.
It seems libass allows out of range values, but does allows the subtitle
to go out of the screen at the bottom (only when moving it to the top
it's "clamped"). Too bad, don't do that then. The bitmap sub rendering
code on the other hand is under our control, and will not move a
subtitle out of the screen.
Fixes: #7986
This is taken from a somewhat older proof-of-concept script. The basic
idea, and most of the implementation, is still the same. The way the
profiles are actually defined changed.
I still feel bad about this being a Lua script, and running user
expressions as Lua code in a vaguely defined environment, but I guess as
far as balance of effort/maintenance/results goes, this is fine.
It's a bit bloated (the Lua scripting state is at least 150KB or so in
total), so in order to enable this by default, I decided it should
unload itself by default if no auto-profiles are used. (And currently,
it does not actually rescan the profile list if a new config file is
loaded some time later, so the script would do nothing anyway if no auto
profiles were defined.)
This still requires defining inverse profiles for "unapplying" a
profile. Also this is still somewhat racy. Both will probably be
alleviated to some degree in the future.
This probably makes it much faster (I wouldn't know, I didn't run any
benchmarks ). Seems to work as well (although I'm not sure, it's not
like I'd perform rigorous tests).
The scale_zimg test seems to mysteriously treat color in fully
transparent alpha differently, which makes no sense, and isn't visible
(but makes the test fail). I can't be bothered with investigating this
more. What do you do with failing tests? Correct, you disable them. Or
rather, you disable whatever appears to cause them to fail, which is the
threading in this case.
This change follows mostly the tile_example.cpp. The slice size uses a
minimum of 64, which was suggested by the zimg author. Some of this
commit is a bit inelegant and weird, such as recomputing the scale
factor for every slice, or the way slice_h is managed. Too lazy to make
this more elegant.
zimg git had a regressio around active_region (which is needed by the
slicing), which was fixed in commit 83071706b2e6bc634. Apparently, the
bug was never released, so just add a warning to the manpage.
Mess this into the --geometry option, because I like to be
irresponsible. I considered adding a separate option, but at least this
allows me to defer the question how the hell this should work as
property (geometry simply and inherently does not).
Tested on IceWM only. Option equality test and string output not tested.
I'm tired of dealing with this frequent spawning of xdg-screensaver when
debugging and what not. xdg-screensaver was never a serious tool anyway,
it's more like some self-deprecating joke by FDO folks.
This will affect X11 on GNOME and other DEs. I'm singling out GNOME
though, because they are the ones actively sabotaging any sane
technical solutions and community cooperation.
I have been accused of taking it out on innocent GNOME users, while none
of this will reach GNOME developers. Of course that is not the
intention.
libplacebo exposes this feature already, because this particular type of
bug is unusually common in practice. Simply make use of it, by exposing
it as an option.
Could probably also bump the libplacebo minimum version to get rid of
the #if, but that would break debian oldoldstable or something.
Fixes#7867.
De-emphasize it, since a user should usually not use it. This _could_ be
used to make the cache seekable with --cache=no, but it's better and
more intuitive to use --cache=yes. As such, the only use of this is for
debugging. I'm not quite sure if this should be removed entirely, but I
still see some value in it (for example if you want the cache lookahead,
but you're using a stream where cache seeking is somehow broken).
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.
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 mode drops or repeats audio data to adapt to video speed, instead
of resampling it or such. It was added to deal with SPDIF. The
implementation was part of fill_audio_out_buffers() - the entire
function is something whose complexity exploded in my face, and which I
want to clean up, and this is hopefully a first step.
Put it in a filter, and mess with the shitty glue code. It's all sort of
roundabout and illogical, but that can be rectified later. The important
part is that it works much like the resample or scaletempo filters.
For PCM audio, this does not work on samples anymore. This makes it much
worse. But for PCM you can use saner mechanisms that sound better. Also,
something about PTS tracking is wrong. But not wasting more time on
this.
Can be useful to force it to adapt to extreme speed changes, while a
higher limit would just use a fraction closer to the original video
speed.
Probably useful for testing only.
As discussed in the referenced issue. This is quite a behavior change,
bit since this option is new, and not included in any releases yet, I
think it's OK.
Fixes: #7648
By default --sub-auto uses "exact". This was far from an "exact" match,
because it added anything that started with the video filename (without
extension), and seemed to end in something that looked like a language
code.
Make this stricter. "exact" still tolerate a language code, but the
video's filename must come before it without any unknown extra
characters. This may not load subtitles in some situations where it
previously did, and where the user might think that the naming
convention is such that it should be considered an exact match.
The subtitle priority sorting seems a bit worthless. I suppose it may
have some value in higher "fuzz" modes (like --sub-auto=fuzzy).
Also remove the mysterious "prio += prio;" line. I probably shouldn't
have checked, but it goes back to commit f16fa9d31 (2003), where someone
wanted to "refine" the priority without changing the rest of the code or
something.
Mostly untested, so have fun.
Fixes: #7702
Consider e.g. --aid=2 with a file that has only 1 track. Then it would
fall back to selecting track 1. Stop doing this. If no matching track is
found, this will not select any track now.
Note that the fingerprint stuff (track_layout_hash in the source)
prevents softens the impact of this change. Without the fingerprint,
playing a dual-audio file with the second track selected, and then a
single-audio file, would play the second file without audio. But the
fingerprint resets it due to differences in the track list.
Try to exhaustively document this and tricky interactions between the
other features. What a damn mess, I think it's simply cursed. Of course
it's still my fault.
See: #7608
Some time ago, properties and options were mostly unified. However, the
track selection properties/options semantics are incompatible to this
change. I'm still trying to handle the fallout.
There are two things that are in the way:
1. Track properties somehow return the runtime selection, not the option
value (all while properties are supposed to be aliases to options
with the same name).
2. The user's track options are not supposed to be changed without
interaction. If a track is auto-selected, the property should return
its ID, but the option value should remain at "auto". Only if the
user actually writes to the property the option should change. E.g.
playing e.g. an audio-only file and then a normal video file not play
the video file with --vid=no just because the audio file had no video
track.
In addition to each of them being in conflict with the property/option
unification, attempt to fix one of them breaks the other one.
Today, we're trying to fix parts of this and avoiding an unfortunate
case where you can get a conflicting option/property value, and where
trying to select a track does nothing if the track to select has the
same ID as the option value.
This breaks 2. from above in certain situations. See manpage additions.
See: #7608
While --input-file was removed for justified reasons, wanting to pass
down socket FDs this way is legitimate, useful, and easy to implement.
One odd thing is that
Fixes: #7592
The mp_filter_run() invocation blocks as long as the demuxer provides
packets and the queue can be filled. That means it may block quite a
long time of the decoder queue size is large (since we use libavcodec in
a blocking manner; it regrettably does not have an async. API).
This made the main thread freeze in certain situations, because it has
to wait on the decoder thread.
Other than I suspected (I wrote that code, but that doesn't mean I know
how the hell it works), this did not freeze during seeking: seek resets
flushed the queue, which also prevented the decoder thread from adding
more frames to it, thus stopping decoding and responding to the main
thread in time. But it does fix the issue that exiting the player waited
for the decoder to finish filling the queue when stopping playback.
(This happened because it called mp_decoder_wrapper_set_play_dir()
before any resets. Related to the somewhat messy way play_dir is
generally set. But it affects all "synchronous" decoder wrapper API
calls.)
This uses pretty weird mechanisms in filter.h and dispatch.h. The
resulting durr hurr interactions are probably hard to follow, and this
whole thing is a sin. On the other hand, this is a _very_ user visible
issue, and I'm happy that it can be fixed in such an unintrusive way.
Let's see how much everyone hates this. Leaving it enabled seems
problematic, because libavcodec returns an unspecific error if it
doesn't like it.
Fixes: #6300
See manpage additions. This has been a topic in MPlayer/mplayer2/mpv
since forever. But since libavcodec multi-threaded decoding was added,
I've always considered this pointless. libavcodec requires you to
"preload" it with packets, and then you can pretty much avoid blocking
on it, if decoding is fast enough.
But in some cases, a decoupled decoder thread _might_ help. Users have
for example come up with cases where decoding video in a separate
process and piping it as raw video to mpv helped. (Or my memory is
false, and it was about vapoursynth filtering, who knows.) So let's just
see whether this helps with anything.
Note that this would have been _much_ easier if libavcodec had an
asynchronous (or rather, non-blocking) API. It could probably have
easily gained that with a small change to its multi-threading code and a
small extension to its API, but I guess not.
Unfortunately, this uglifies f_decoder_wrapper quite a lot. Part of this
is due to annoying corner cases like legacy frame dropping and hardware
decoder state. These could probably be prettified later on.
There is also a change in playloop.c: this is because there is a need to
coordinate playback resets between demuxer thread, decoder thread, and
playback logic. I think this SEEK_BLOCK idea worked out reasonably well.
There are still a number of problems. For example, if the demuxer cache
is full, the decoder thread will simply block hard until the output
queue is full, which interferes with seeking. Could also be improved
later. Hardware decoding will probably die in a fire, because it will
run out of surfaces quickly. We could reduce the queue to size 1...
maybe later. We could update the queue options at runtime easily, but
currently I'm not going to bother.
I could only have put the lavc wrapper itself on a separate thread. But
there is some annoying interaction with EDL and backward playback shit,
and also you would have had to loop demuxer packets through the
playloop, so this sounded less annoying.
The food my mother made for us today was delicious.
Because audio uses the same code, also for audio (even if completely
pointless).
Fixes: #6926
Try to deal with various corner cases. But when I fix one thing, another
thing breaks. (And it's 50/50 whether I find the breakage immediately or
a few months later.) So results may vary.
The default for--hr-seek is changed to "default" (not creative enough to
find a better name). In this mode, audio seeking is exact if there is no
video, or if the video has only a single frame. This change is actually
pretty dumb, since audio frames are usually small enough that exact
seeking does not really add much. But it gets rid of some weird special
cases.
Internally, the most important change is that is_coverart and is_sparse
handling is merged. is_sparse was originally just a special case for
weird .ts streams that have the corresponding low-level flag set. The
idea is that they're pretty similar anyway, so this would reduce the
number of corner cases. But I'm not sure if this doesn't break the
original intended use case for it (I don't have a sample anyway).
This changes last-frame handling, and respects the duration of the last
frame only if audio is disabled. This is mostly "coincidental" due to
the need to make seeking past EOF trigger player exit, and is caused by
setting STATUS_EOF early. On the other hand, this might have been this
way before (see removed chunk close to it).
This is useful with live streams, and it's not much worse than the h264
first packet hack, which reads some data anyway.
For some reason, the option wasn't even documented, so do that.
In addition, print the start time even if it's negative. That should not
be possible, but for some reason, the field is an int64_t copied from an
uint64_t so... whatever. Keeping the logging slightly more straight
forward is better anyway.
Remove some redundant fields that controlled or indicated whether the
demuxer was/could/should prefetch. Redefine how the eof/reading fields
work.
The in->eof field is now always valid, instead of weirdly being reset to
false in random situations. The in->reading field now corresponds to
whether the demuxer thread is working at all, and is reset if it stops
doing anything.
Also, I always found it stupid that dequeue_packet() forced the demuxer
thread to retry reading if it was EOF. This makes little sense, but was
probably added for files that are being appended to (running downloads).
It makes no sense, because if the cache really tried to read until file
EOF, it would encounter partial packets and throw errors, so all is lost
anyway. Plus stream_file now handles this better. So stop this behavior,
but add a temporary option that enables the old behavior.
I think checking for ds->eager when enabling prefetching never really
made sense (could be debated, but no, not really). On the other hand,
the change above exposed a missing wakeup in the backward demuxing code.
Some chances of regressions that could make it stuck in certain states
or so, or incorrect demuxer cache state reporting to the player
frontend.
In all_formats mode, we've ignored what --ytdl-format did so far, since
we've converted the full format list, instead of just the formats
selected by youtube-dl.
But we can easily restore --ytdl-format behavior: just mark the selected
tracks as default tracks.
I don't think the skip_muxed option was overlay useful. While it was
nice to filter out the low quality muxed versions (as it happens on the
alphabetic site, I suspect it's compatibility stuff), it's not really
necessary, and just makes for another tricky and rarely used
configuration option. (This was different before muxed tracks were also
delay-loaded, and including the muxed versions slowed down loading.)
Add the force_all_formats option instead, which handles the HLS case.
Set it to true because they are also delay-loaded now, and don't slow
down startup as much.
See manpage additions. We would have to extend delay_open to support
multiple sub-tracks (for audio and video), and we'd still don't know (?)
whether it might contain more than one stream each (thinking of HLS
master streams). And if it's a true interleaved file (such as a "normal"
mp4 file provided as fallback for more primitive players), we'd either
have to signal such "bundled" tracks, or waste bandwidth.
This restructures a lot. The if/else tree in add_single_video for format
selection was a bit annoying, so it's split into separate if blocks,
where it checks each time whether a URL was determined yet.
Pretty worthless I guess. I only tested one site (and 2 videos), it's
somewhat likely that it will break with other sites. Even if you leave
the option disabled (the default).
Slightly related to #3548. This will allows you to use the bitrate
stream selection mechanism, that was added for HLS, with normal videos.
Works as ad-filter. I had some more plans, for example replacing
matching text with different text, but for now it's dropping matches
only. There's a big warning in the manpage that I might change
semantics. For example, I might turn it into a primitive sed.
In a sane world, you'd probably write a simple script that processes
downloaded subtitles before giving them to mpv, and avoid all this
complexity. But we don't live in a sane world, and the sooner you learn
this, the happier you will be. (But I also want to run this on muxed
subtitles.)
This is pretty straightforward. We use POSIX regexes, which are readily
available without additional pain or dependencies. This also means it's
(apparently) not available on win32 (MinGW). The regex list is because I
hate big monolithic regexes, and this makes it slightly better.
Very superficially tested.
As requested I guess. It behaves quite similar to the --loop* options.
Not quite happy with the idea that 1) the option is mutated on each
operation (but at least it's consistent with --loop* and doesn't require
more properties), and 2) the ab-loop command will do nothing once all
loop iterations are done. As a concession, the OSD shows something about
"disabled".
Fixes: #7360
this creates a default log for the last mpv run when started from the
bundle. that way one can get a log of what happened even after an issue
occurred. also add a menu entry under Help to show the current log, but
only when the bundle is used.
Fixes#7396Fixes#2547
See #7435 and related for context.
Basically, it seems that while the original vsfilter processed subtitles
like with this option set to "yes", many current players (mpc-hc
default, vlc, probably most libass users) treat them like with "no". In
the linked issue, this makes rendering severely slower, and can consume
a lot of memory (or just overflow libass memory calculations). It seems
that changing this to "no" will lead to more good than bad, especially
because newer subtitles may be authored for the "no" behavior.
Most libass users seem to use "no" exactly because they do not call
ass_set_storage_size() at all. This API was needed because the scaling
of the subtitles depends on the video size (vsfilter bugs, or
something). In addition, it's my personal opinion that rendering should
not depend on the video at all, so I like setting the default of this to
"no".
This originally existed as a hack for weston. In certain scenarios, a
frame taking too long to render would cause vo_wayland_wait_frame to
timeout which would result in a ton of dropped frames. The naive
solution was to just to add a slight delay to the time value. If a
frame took too long, it would likely to fall under the timeout value and
all was well. This was exposed to the user since the default delay
(1000) was completely arbitrary.
However with presentation time, this doesn't appear to be neccesary.
Fresh frames that take longer than the display's refresh rate (16.666 ms
in most cases) behave well in Weston. In the other two main compositors
without presentation time (GNOME and Plasma), they also do not
experience any ill effects. It's better not to overcomplicate things, so
this "feature" can be removed now.
See manpage additions. The libarchive behavior mentioned in the last
paragraph there is technically unrelated, but makes this new option
mostly pointless.
See: #7182
In the distant past, the cuviddec backed copy hwaccel could be
configured directly using lavc options. However, since that time,
we gained support for automatic hw ctx creation which ended up
bypassing the lavc options.
Rather than trying to find a way to pass those options again, a
better idea is to make the 'cuda-decode-device' option, used by
the interop hwaccels, work for the copy hwaccels too.
And that's pretty simple: we have to add a create function that
checks the option and passes it on to ffmpeg.
Note that this does require a slight re-jig to the configuration
flags, as we now have a scenario where we want to build with support
for the cuda copy hwaccels but not the interop ones. So we need
a distinct configuration flag for that combination.
Fixes#7295.
Add an "auto-safe" mode, mostly triggered by Ubuntu's nonsense to force
hwdec=vaapi in the global config file in their mpv package. But to be
honest it's probably something more people want.
This is implemented as explicit whitelist. On Windows, HEVC/Intel is
sometimes broken, but it's still whitelisted, and in theory we'd need a
detailed whitelist of device names etc. (like for example browsers tend
to do). On OSX, videotoolbox is a pretty bad choice, but unfortunately
the only one, so it's whitelisted too. There may be a larger number of
hwdec wrappers that work anyway, and I'm for example ignoring Android.
Apparently there are two different options for controlling which
screen an mpv window goes onto: --fs-screen and --screen. The former
explicitly only controls which screen a fullscreened window goes onto,
but does not appear to actually care about this option at runtime for
X11, so pressing f will always fullscreen to the screen mpv is currently
on. This means the option is of questionable usefulness for starters.
Making it worse, if you use --screen=1 --fs, mpv will actually fullscreen
on screen 0, because --fs-screen isn't set. Instead of doing that, fall
back to whatever --screen is set to.
Whenever I deal with this, I have to look at the code to make sense of
this. And beyond that, there are some strange inconsistencies. (I think
this code is cursed. It always was, and maybe always will be.)
Although the manpage claimed that using multiple items for -add etc. is
deprecated, string list options didn't warn against it. So add the
warning, and add something in the changelog (even though nobody will
ever read this).
The manpage mentioned --vf-append, but this didn't even exist. So add
it, I guess. We encourage using -append for the other option types, so
for consistency, it should work on filter options. (And I already
tricked me into believing it existed when I mentioned it in the
manpage.)
Make the "operations" table separate for all option types, and mention
the option type on every single of the top-level list options.
the Apple Remote has long been deprecated and abandoned by Apple.
current macs don't come with support for it anymore. support might be
re-added with the next commit.
Seems like this was silently changed to enabled by default on the change
to libplacebo, without adjusting the manpage. Fix the documented
default.
Also add a comment about Nvidia; see referenced issue.
Fixes: #7245
Merged from mpv-repl git repo commit 5ea2bf64f9c239f0326b02. Some
changes were made on top of it:
- Tabs were converted to 4 spaces indentation (plus some manual
indentation fixes in some places).
- All user-visible mentions of "repl" were renamed to "console".
- The README was converted to a manpage (with heavy changes, some
additions taken from stats.rst; rossy converted the key bindings
table to RST).
- The method to change the default key binding was changed.
- Change minor detail about "font" default value setting (not a
functional change).
- Integrate into the player as builtin script, including an option to
prevent loading it.
Above changes and commit message done by wm4.
Signed-off-by: wm4 <wm4@nowhere>
I often watch sporting events. On many occasions I get files with the
same filename for each session. For example, for F1 I might have the
following directory structure:
F1/
FP1.mkv
FP2.mkv
FP3.mkv
Qualification.mkv
Race.mkv
Since usually one simply watches one race after the other, I usually
just rsync the new event's files over the old ones, so, for example,
Race.mkv will be replaced from the file for the last event with the file
from the new event.
One problem with this is that I like to use --resume-playback for other
kinds of media, so I have it on by default. That works great for, say, a
movie, but doesn't work so well with this scheme, because you can
trivially forget to pass --no-resume-playback on the command line and
end up 2 hours in, watching spoilers as the race results scroll down the
screen :-)
This patch adds a new option, --resume-playback-check-mtime, which
validates that the file's mtime hasn't changed since the watch_later
configuration was saved. It does this by setting the watch_later
configuration to have the same mtime as the file after it is saved.
Switching back and forth between checking mtime and not checking mtime
works fine, as we only choose whether to compare based on it, but we
update the watch_later configuration mtime regardless of its value.
I have no idea why this still exists, since we have --input-ipc-server.
I think there was something about Windows, but the latter option is
implemented even on Windows.
It sometimes happens that HLS streams freeze because the HTTP server is
not responding for a fragment (or something similar, the exact
circumstances are unknown). The --timeout option didn't affect this,
because it's never set on HLS recursive connections (these download the
fragments, while the main connection likely nothing and just wastes a
TCP socket).
Apply an elaborate hack on top of an existing elaborate hack to somehow
get these options set. Of course this could still break easily, but hey,
it's ffmpeg, it can't not try to fuck you over. I'm so fucking sick of
ffmpeg's API bullshit, especially wrt. HLS.
Of course the change is sort of pointless. For HLS, GET requests should
just aggressively retried (because they're not "streamed", they're just
actual files on a CDN), while normal HTTP connections should probably
not be made this fragile (they could be streamed, i.e. they are backed
by some sort of real time encoder, and block if there is no data yet).
The 1 minute default timeout is too high to save playback if this
happens with HLS.
Vaguely related to #5793.
Until now, we've made FFmpeg use the default network timeout - which is
apparently infinite. I don't know if this was changed at some point,
although it seems likely, as I was sure there was a more useful default.
For most use cases, a smaller timeout is more useful (for example
recording something in the background), so force a timeout of 1 minute.
See: #5793
Until now, each .c file in test/ was built as separate, self-contained
binary. Each binary could be run to execute the tests it contained.
Change this and make them part of the normal mpv binary. Now the tests
have to be invoked via the --unittest option. Do this for two reasons:
- Tests now run within a "properly" initialized mpv instance, so all
services are available.
- Possibly simplifying the situation for future build systems.
The first point is the main motivation. The mpv code is entangled with
mp_log and the option system. It feels like a bad idea to duplicate some
of the initialization of this just so you can call code using them.
I'm also getting rid of cmocka. There wouldn't be any problem to keep it
(it's a perfectly sane set of helpers), but NIH calls. I would have had
to aggregate all tests into a CMUnitTest list, and I don't see how I'd
get different types of entry points easily. Probably easily solvable,
but since we made only pretty basic use of this library, NIH-ing this is
actually easier (I needed a list of tests with custom metadata anyway,
so all what was left was reimplement the assert_* helpers).
Unit tests now don't output anything, and if they fail, they'll simply
crash and leave a message that typically requires inspecting the test
code to figure out what went wrong (and probably editing the test code
to get more information). I even merged the various test functions into
single ones. Sucks, but here you go.
chmap_sel.c is merged into chmap.c, because I didn't see the point of
this being separate. json.c drops the print_message() to go along with
the new silent-by-default idea, also there's a memory leak fix unrelated
to the rest of this commit.
The new code is enabled with --enable-tests (--enable-test goes away).
Due to waf's option parser, --enable-test still works, because it's a
unique prefix to --enable-tests.
The use of glXGetCurrentDisplay() restricted this to the GLX backend.
But actually it works under EGL as well. Removing the GLX-specific call
and using the general mpv-internal method to get the X "Display" makes
it work in mpv.
I didn't know this. Nvidia didn't list this as extension in the EGL
context when I still used their GPUs.
Note that this might in theory break use of vdpau in some libmpv clients
using the render API. But only if MPV_RENDER_PARAM_X11_DISPLAY is not
used, and they relied on mpv using glXGetCurrentDisplay(). EGL does not
provide such an API, and hwdec_vaapi.c also uses what hwdec_vdpau.c uses
now. Considering that vaapi is preferable these days, it's not bad at
all if these clients get "broken". They can be easily fixed by passing
the display to mpv correctly.
(Only half of the buffer is actually used in a useful way, see manpage
or commit which added the option.)
Might have some advantages with broken network filesystem drivers.
See: #6802
In some corner cases (see #6802), it can be beneficial to use a larger
stream buffer size. Use this as argument to rewrite everything for no
reason.
Turn stream.c itself into a ring buffer, with configurable size. The
latter would have been easily achievable with minimal changes, and the
ring buffer is the hard part. There is no reason to have a ring buffer
at all, except possibly if ffmpeg don't fix their awful mp4 demuxer, and
some subtle issues with demux_mkv.c wanting to seek back by small
offsets (the latter was handled with small stream_peek() calls, which
are unneeded now).
In addition, this turns small forward seeks into reads (where data is
simply skipped). Before this commit, only stream_skip() did this (which
also mean that stream_skip() simply calls stream_seek() now).
Replace all stream_peek() calls with something else (usually
stream_read_peek()). The function was a problem, because it returned a
pointer to the internal buffer, which is now a ring buffer with
wrapping. The new function just copies the data into a buffer, and in
some cases requires callers to dynamically allocate memory. (The most
common case, demux_lavf.c, required a separate buffer allocation anyway
due to FFmpeg "idiosyncrasies".) This is the bulk of the demuxer_*
changes.
I'm not happy with this. There still isn't a good reason why there
should be a ring buffer, that is complex, and most of the time just
wastes half of the available memory. Maybe another rewrite soon.
It also contains bugs; you're an alpha tester now.
Not like anyone reads it. Although putting all this text before listing
the allowed option values sort of has the intention to discourage users
from using the option at all. Advertise Ctrl+h, which is a decent way of
enabling hardware decoding temporarily.
The user can raise the number of tolerated hardware decoding errors. On
the other hand, we have a static limit on packets that are "saved" for
fallback handling (and that's a good idea to avoid unbounded memory
usage). In this case, it could happen that the start of a file was fine
after a fallback, but after that buffered amount of data, it would
suddenly skip.
It's more useful to skip buffering entirely if the number of tolerated
decoding errors exceeds the fixed buffer.
(And also, I'm sure nobody gives a shit about this feature.)
The statement about the display FPS is outdated by several years.
"audio"-sync mode does not use the display FPS anymore, and that it's
X11 only also isn't true anymore.
These modes have separate implementations for audio and display video
sync. modes, so the explanations are separate.
Why the hell are users playing around with this anyway? The explanations
are probably too special to make sense for anyone who doesn't know the
code (and who knows the code doesn't need them anyway), but whatever.
This is mostly just because of the odd RGB default gamma issue, which
shouldn't have any real impact. This also sets allow_approximate_gamma,
which I hope is fine for normal use cases.
Raise swscale and zimg default parameters. This restores screenshot
quality settings (maybe) unset in the commit before. Also expose some
more libswscale and zimg options.
Since these options are also used for VOs like x11 and drm, this will
make x11/drm/etc. much slower. For compensation, provide a profile that
sets the old option values: sw-fast. I'm also enabling zimg here, just
as an experiment.
The core problem is that we have a single set of command line options
which control the settings used for most swscale/zimg uses. This was
done in the previous commit. It cannot differentiate between the VOs,
which need to be realtime and may accept/require lower quality options,
and things like screenshots or vo_image, which can be slower, but should
not sacrifice quality by default.
Should this have two sets of options or something similar to do the
right thing depending on the code which calls libswscale? Maybe. Or
should I just ignore the problem, make it someone else's problem (users
who want to use software conversion VOs), provide a sub-optimal
solution, and call it a day? Definitely, sounds good, pushing to master,
goodbye.
By default utilizes the color space of the desktop on which the
swap chain is located. If a specific value is defined, it will be
instead be utilized.
Enables configuration of the PQ color space (BT.2020 primaries,
PQ transfer function) for HDR.
Additionally, signals the swap chain color space to the renderer,
so that the render looks correct without having to specify
target-trc or target-prim manually.
Due to all of the APIs being Win10+ only, will only work starting
with Windows 10.
Enabling this by default probably causes a number of issues, such as
breaking vo_sdl, or reacting to various input devices while the window
is not focused. It's also pretty obscure, or at least new. Disable it by
default.
This doesn't take a ',' separated list. --script is just an alias for
--scripts--append. --scripts accepts a list, but uses the
mplayer-inherited platform-dependent path separator.
Fixes: #5996
Internally, vo_gpu uses NaN for some options to indicate a default value
that is different depending on the context (e.g. different scalers).
There are 2 problems with this:
1. you couldn't reset the options to their defaults
2. NaN is a damn mess and shouldn't be part of the API
The option parser already rejected NaN explicitly, which is why 1.
didn't work. Regarding 2., JSON might be a good example, and actually
caused a bug report.
Fix this by mapping NaN to the special value "default". I think I'd
prefer other mechanisms (maybe just having every scaler expose separate
options?), but for now this will do. See you in a future commit, which
painfully deprecates this and replaces it with something else.
I refrained from using "no" (my favorite magic value for "unset" etc.)
because then I'd have e.g. make --no-scale-param1 work, which in
addition to a lot of effort looks dumb and nobody will use it.
Here's also an apology for the shitty added test script.
Fixes: #6691
The code is very basic:
- only handles gamepads, could be extended for generic joysticks in the
future.
- only has button mappings for controllers natively supported by SDL2.
I heard more can be added through env vars, there's also ways to load
mappings from text files, but I'd rather not go there yet. Common ones
like Dualshock are supported natively.
- analog buttons (TRIGGER and AXIS) are mapped to discrete buttons using an
activation threshold.
- only supports one gamepad at a time. the feature is intented to use
gamepads as evolved remote controls, not play multiplayer games in mpv :)
Awful shit. I probably wouldn't accept this code from someone else, just
so you know.
The idea is that a sws_utils user can automatically use zimg without
large code changes. Basically, laziness. Since zimg support is still
very new, and I don't want that anything breaks just because zimg was
enabled at build time, an option needs to be set to enable it. (I have
especially especially obscure stuff in mind, which is all what
libswscale is used in mpv.)
This _still_ doesn't cause zimg to be used anywhere, because the
sws_utils user has to opt-in by setting allow_zimg. This is because some
users depend on certain libswscale features.
This provides a very similar API to sws_utils.h, which can be used to
convert and scale from one mp_image to another.
This commit adds only the code, but does not use it anywhere.
The code is quite preliminary and barely tested. It supports only a few
pixel formats, and will return failure for many others. (Unlike
libswscale, which tries to support anything that FFmpeg knows.)
zimg itself accepts only planar formats. Supporting other formats
requires manual packing/unpacking. (Compared to libswscale, the zimg API
is generally lower level, but allows for more flexibility.) Only BGR0
output was actually tested. It appears to work.
On a audio/video desync by more than 0.5 seconds, display-sync mode was
disabled, and not enabled again (until playback restart, e.g. a seek).
The idea was that it this only happens when this playback mode is broken
and can't perform well anyway (A/V desync is a clear indication that
something is very wrong). Instead of behaving like a god damn POS, it
should revert to the more robust audio-sync mode.
Unfortunately, this could happen sporadically due to temporary system
performance problems, such as toggling fullscreen. Users didn't like
this, and asked for a function to disable it, or to recover in some
other way.
This mechanism is questionable anyway. If an ignorant user enables
display-sync, and encounters problems with it (without being able to
determine that display-sync is messing up), the player will still behave
like a POS on every playback, and even after every seek. It might
actually be helpful to fail more consistently. Also, I've found that
it's sill relatively reliable anyway even without this mechanism.
So just remove the fallback.
Fixes: #7048
OK, so --cache-secs is useless, because the default is set to 10 hours.
And that part about the "maximum" was obviously a lie (I wonder if it
simply changed at some point).
Query information on the system output most linked to the swap chain,
and either utilize a user-configured format, or either 8bit
RGBA or 10bit RGB with 2bit alpha depending on the system output's
bit depth.
In this case, gapless will most likely not work. It will result in (very
slight) desync, or (more commonly with small buffer sizes), in an
underflow.
I think it would be legitimate to disable gapless at end of playback
completely if video is enabled at all. But this would need an exception
for cover art mode, so I guess the current solution is OK as well.
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.
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.
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.
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?
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.
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.
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.
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.
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.
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. ad9e473c55). 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.
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.
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).
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.
Clarify existing semantics for the --start/--end/--length options.
De-emphasize the difference between absolute and relative timestamps,
since they've not been different by default since mpv 0.14.
Document a bug, that also happens to work as a feature: if the option
value begins with spaces, the code for checking for relative timestamps
is inactive, and they're always considered absolute. The check is done
on the first character of the string - so even a negative timestamp will
be treated as absolute.)
Yes, this is useful in extremely rare situations, such as when you
really want send a specific timestamp (even a negative one) to the
demuxer.
This changes the behavior of the --ab-loop-a/b options. In addition, it
makes it work with backward playback mode.
The most obvious change is that the both the A and B point need to be
set now before any looping happens. Unlike before, unset points don't
implicitly use the start or end of the file. I think the old behavior
was a feature that was explicitly added/wanted. Well, it's gone now.
This is because of 2 reasons:
1. I never liked this feature, and it always got in my way (as user).
2. It's inherently annoying with backward playback mode.
In backward playback mode, the user wants to set A/B in the wrong order.
The ab-loop command will first set A, then B, so if you use this command
during backward playback, A will be set to a higher timestamps than B.
If you switch back to forward playback mode, the loop would stop
working. I want the loop to just continue to work, and the chosen
solution conflicts with the removed feature.
The order issue above _could_ be fixed by also switching the AB-loop
user option values around on direction switch. But there are no other
instances of option changes magically affecting other options, and doing
this would probably lead to unexpected misery (dying from corner cases
and such).
Another solution is sorting the A/B points by timestamps after copying
them from the user options. Then A/B options set in backward mode will
work in forward mode. This is the chosen solution. If you sort the
points, you don't know anymore whether the unset point is supposed to
signify the end or the start of the file.
The AB-loop code is slightly better abstracted now, so it should be easy
to restore the removed feature. It would still require coming up with a
solution for backwards playback, though.
A minor change is that if one point is set and the other is unset, I'm
rendering both the chapter markers and the marker for the set point.
Why? I don't know. My test file had chapters, and I guess I decided this
looked better.
This commit also fixes some subtle and obvious issues that I already
forgot about when I wrote this commit message. It cleans up some minor
code duplication and nonsense too.
Regarding backward playback, the code uses an unsanitary mix of internal
("transformed") and user timestamps. So the play_dir variable appears
more than usual.
To mention one unfixed issue: if you set an AB-loop that is completely
past the end of the file, it will get stuck in an infinite seeking loop
once playback reaches the end of the file. Fixing this reliably seemed
annoying, so the fix is "just don't do this". It's not a hard freeze
anyway.
Has been deprecated for almost 3 years. Manpage didn't mention the
deprecation, but CLI and release notes did. It wouldn't be much effort
to keep this option working, but I just don't see the damn point.
--start/--end can specify chapters using special syntax, which is
equivalent.
This commit generally fixes backward playing in wav, at least in most
PCM cases.
libavformat's wav demuxer (and actually all other raw PCM based
demuxers) have a specific behavior that breaks backward demuxing. The
same thing also breaks persistent seek ranges in the demuxer cache,
although that's less critical (it just means some cached data gets
discarded). The backward demuxing issue is fatal, will log the message
"Demuxer not cooperating.", and then typically stop doing anything.
Unlike modern media formats, these formats don't organize media data in
packets, but just wrap a monolithic byte stream that is described by a
header. This is good enough for PCM, which uses fixed frames (a single
sample for all audio channels), and for which it would be too expensive
to have per frame headers.
libavformat (and mpv) is heavily packet based, and using a single packet
for each PCM frame causes too much overhead. So they typically "bundle"
multiple frames into a single packet. This packet size is obviously
arbitrary, and in libavformat's case hardcoded in its source code.
The problem is that seeking doesn't respect this arbitrary packet
boundary. Seeking is sample accurate. You can essentially seek inside a
packet. The resulting packets will not be aligned with previously
demuxed packets. This is normally OK.
Backward seeking (and some other demuxer layer features) expect that
demuxing an earlier demuxed file position eventually results in the same
packets, regardless of the seeks that were done to get there. I like to
call this "deterministic" demuxing. Backward demuxing in particular
requires this to avoid overlaps, which would make it rather hard to get
continuous output.
Fix this issue by detecting wav and hopefully other raw audio formats
with a heuristic (even PCM needs to be detected as heuristic). Then, if
a seek is requested, align the seek timestamps on the guessed number of
samples in the audio packets returned by the demuxer.
The heuristic excludes files with multiple streams. (Except "attachment"
video streams, which could be an ID3 tag. Yes, FFmpeg allows ID3 tags on
WAV files.) Such files will inherently use the packet concept in some
way.
We don't know how the demuxer chooses the internal packet size, but we
assume that it's fixed and aligned to PCM frame sizes. The frame size is
most likely given by block_align (the native wav frame size, according
to Microsoft). We possibly need to explicitly read and discard a packet
if the seek is done without reading anything before that. We ignore any
subsequent packet sizes; we need to avoid the very last packet, which
likely has a different size.
This hack should be rather benign. In the worst case, it will "round"
the seek target a little, but the maximum rounding amount is bounded.
Maybe we _could_ round up if SEEK_FORWARD is specified, but I didn't
bother.
An earlier commit fixed the same issue for mpv's demux_raw.
An alternative, and probably much better solution would be clipping
decoded data by timestamp. demux.c could allow the type of overlap the
wav demuxer introduces, and instruct the decoder to clip the output
against the last decoded timestamp. There's already an infrastructure
for this (demux_packet.end field) used by EDL/ordered chapters.
Although this sounds like a good solution, mpv unfortunately uses floats
for timestamps. The rounding errors break sample accuracy. Even if you
used integers, you'd need a timebase that is sample accurate (not always
easy, since EDL can merge tracks with different sample rates).
As well as other filtering. I was writing this with the assumption that
timestamps go backwards (which I first planned to do). But in fact,
timestamps go forward, frame durations are positive, and adding a frame
duration to a timestamp yields the correct result. The only strange
thing is that timestamps are negative.
Also, media of course goes backwards. In other possible implementation,
filters would see normal forward playback, interrupted by seeks or
discontinuities. It turns out the current implementation of providing a
continuous backward media stream is probably better for filters.
Even deinterlacing seems to work. libavcodec always outputs fields in as
interleaved frames (i.e. fields are not reversed), and making up
timestamps for the new frames (when doubling the framerate) works
exactly like like in the forward case.
Actually the previous paragraph was a lie, and libavcodec does not
output fields as interleaved frames in rare cases. Sometimes AVFrame
contains single fields. In this case you'd need to inverse the field
dominance for deinterlacing filters to work correctly.
The way backward playback is implemented doesn't break basic assumptions
about timestamps after the decoder, so I guess all the encoding mode
needs to do is to adjust for the start offset, which it already does.
Though I might be wrong and my test was possibly flawed.
Stream recording on the other hand will fail immediately with
--record-file, and --stream-record will probably yield unexpected
results if any backstep seeks are done.
Make --audio-backward-overlap default to 2 for Opus. I have no idea why
this is needed. It seems to fix backward decoding though (going purely
by listening).
Normally, this should not be needed, since initial padding is completely
contained within the first packet (normally, and in the case I tested).
So the 2nd packet/frame should be fine, but for some unknown reason it
works only with the 3rd.
The only reasonable solution to this is probably to make discarding of
preroll frames based on timestmaps, instead of frame/packet count. But
then you get issues with video and its dumb timestamp reordering. So for
now, fuck it.
This seems more useful in general. This change also happens to fix a
miscounting of preroll packets when some of them were "rounded" away,
and which could make it stuck.
Also a simple intra-refresh encode with x264 (and muxed to mkv by it)
seems to work now. I guess I misinterpreted earlier results.
Just "mpv file.mkv --play-direction=backward" did not work, because
backward demuxing from the very end was not implemented. This is another
corner case, because the resume mechanism so far requires a packet
"position" (dts or pos) as reference. Now "EOF" is another possible
reference.
Also, the backstep mechanism could cause streams to find different
playback start positions, basically leading to random playback start
(instead of what you specified with --start). This happens only if
backstep seeks are involved (i.e. no cached data yet), but since this is
usually the case at playback start, it always happened. It was racy too,
because it depended on the order the decoders on other threads requested
new data. The comment below "resume_earlier" has some more blabla.
Some other details are changed.
I'm giving up on the "from_cache" parameter, and don't try to detect the
situation when the demuxer does not seek properly. Instead, always seek
back, hopefully some more.
Instead of trying to adjust the backstep seek target by a random value
of 1.0 seconds. Instead, always rely on the random value provided by the
user via --demuxer-backward-playback-step. If the demuxer should really
get "stuck" and somehow miss the seek target badly, or the user sets the
option value to 0, then the demuxer will not make any progress and just
eat CPU. (Although due to backward seek semantics used for backstep
seeks, even a very small seek step size will work. Just not 0.)
It seems this also fixes backstepping correctly when the initial seek
ended at the last keyframe range. (The explanation above was about the
case when it ends at EOF. These two cases are different. In the former,
you just need to step to the previous keyframe range, which was broken
because it didn't always react correctly to reaching EOF. In the latter,
you need to do a separate search for the last keyframe.)
Simple enough to do. May have mixed results. Typically, bitmap subtitles
will have a tight bounding box around the rendered text. But if for
example there is text on the top and bottom, it may be a single big
bitmap with a large transparent area between top and bottom. In
particular, DVD subtitles are really just a single screen-sized
RLE-encoded bitmap, though libavcodec will crop off transparent areas.
Like with sd_ass, you can't move subtitles _down_ if they are already in
their origin position. This could probably be improved, but I don't want
to deal with that right now.
Not specifying a --start or using --start=100% with
--play-direction=backward usually does not work. The demuxer gets no
packets and immediately enters EOF state, which then hangs because
backward playback mode neither considers this mode, nor propagates the
EOF.
As far as demuxer implementations are concerned, this behavior is OK and
even wanted. Seeking near the end with SEEK_FORWARD set is allowed not
to return any packets (so a normal relative forward seek as done by the
user would end playback). Seeking exactly to the end or past it without
SEEK_FORWARD set is probably also sane.
Another vaguely related issue is that a backward seek during playback
start does not "establish" the demux position correctly: if stream A
hits the next keyframe and seeks back, while stream B has not had a
chance to read a packet yet, then stream B will never try to read from
the old position. The effect is that stream B (and thus playback) will
effectively miss the seek target. This is "random" because it depends on
the order and number of packet read calls made by the decoders.
Fixing this is probably hard, and requires extending the already complex
state machine with more states, so turn the manpage into a TODO list for
now.
Raw audio formats can be accessed sample-wise, and logically audio
packets demuxed from it would contain only 1 sample. This is
inefficient, so raw audio demuxers typically "bundle" multiple samples
in one packet.
The problem for the demuxer cache and backward playback is that they
need properly aligned packets to make seeking "deterministic". The
requirement is that if you read some packets, and then seek back, you
eventually see the same packets again. demux_raw basically allowed to
seek into the middle of a previously returned packet, which makes it
impossible to make the transition seamless. (Unless you'd be aware of
the packet data format and cut them to make it seamless, which is too
complex for such a use case.)
Solve this by always aligning seeks to packet boundaries. This reduces
the seek accuracy to the arbitrarily chosen packet size. But you can use
hr-seek to fix this. The gain from not making raw audio an awful special
case pays in exchange for this "stupid" suggestion to use hr-seek.
It appears this also fixes that it could and did seek into the middle of
the frame (not sure if this code was ever tested - it goes back to
removing the code duplication between the former demux_rawaudio.c and
demux_rawvideo.c).
If you really cared, you could introduce a seek flag that controls
whether the seek is aligned or not. Then code which requires
"deterministic" demuxing could set it. But this isn't really useful for
us, and we'd always set the flag anyway, unless maybe the caching were
forced disabled.
libavformat's wav demuxer exhibits the same issue. We can't fix it (it
would require the unpleasant experience of contributing to FFmpeg), so
document this in otions.rst. In theory, this also affects seek range
joining, but the only bad effect should be that cached data is
discarded.
See manpage additions. This is a huge hack. You can bet there are shit
tons of bugs. It's literally forcing square pegs into round holes.
Hopefully, the manpage wall of text makes it clear enough that the whole
shit can easily crash and burn. (Although it shouldn't literally crash.
That would be a bug. It possibly _could_ start a fire by entering some
sort of endless loop, not a literal one, just something where it tries
to do work without making progress.)
(Some obvious bugs I simply ignored for this initial version, but
there's a number of potential bugs I can't even imagine. Normal playback
should remain completely unaffected, though.)
How this works is also described in the manpage. Basically, we demux in
reverse, then we decode in reverse, then we render in reverse.
The decoding part is the simplest: just reorder the decoder output. This
weirdly integrates with the timeline/ordered chapter code, which also
has special requirements on feeding the packets to the decoder in a
non-straightforward way (it doesn't conflict, although a bugmessmass
breaks correct slicing of segments, so EDL/ordered chapter playback is
broken in backward direction).
Backward demuxing is pretty involved. In theory, it could be much
easier: simply iterating the usual demuxer output backward. But this
just doesn't fit into our code, so there's a cthulhu nightmare of shit.
To be specific, each stream (audio, video) is reversed separately. At
least this means we can do backward playback within cached content (for
example, you could play backwards in a live stream; on that note, it
disables prefetching, which would lead to losing new live video, but
this could be avoided).
The fuckmess also meant that I didn't bother trying to support
subtitles. Subtitles are a problem because they're "sparse" streams.
They need to be "passively" demuxed: you don't try to read a subtitle
packet, you demux audio and video, and then look whether there was a
subtitle packet. This means to get subtitles for a time range, you need
to know that you demuxed video and audio over this range, which becomes
pretty messy when you demux audio and video backwards separately.
Backward display is the most weird (and potentially buggy) part. To
avoid that we need to touch a LOT of timing code, we negate all
timestamps. The basic idea is that due to the navigation, all
comparisons and subtractions of timestamps keep working, and you don't
need to touch every single of them to "reverse" them.
E.g.:
bool before = pts_a < pts_b;
would need to be:
bool before = forward
? pts_a < pts_b
: pts_a > pts_b;
or:
bool before = pts_a * dir < pts_b * dir;
or if you, as it's implemented now, just do this after decoding:
pts_a *= dir;
pts_b *= dir;
and then in the normal timing/renderer code:
bool before = pts_a < pts_b;
Consequently, we don't need many changes in the latter code. But some
assumptions inhererently true for forward playback may have been broken
anyway. What is mainly needed is fixing places where values are passed
between positive and negative "domains". For example, seeking and
timestamp user display always uses positive timestamps. The main mess is
that it's not obvious which domain a given variable should or does use.
Well, in my tests with a single file, it suddenly started to work when I
did this. I'm honestly surprised that it did, and that I didn't have to
change a single line in the timing code past decoder (just something
minor to make external/cached text subtitles display). I committed it
immediately while avoiding thinking about it. But there really likely
are subtle problems of all sorts.
As far as I'm aware, gstreamer also supports backward playback. When I
looked at this years ago, I couldn't find a way to actually try this,
and I didn't revisit it now. Back then I also read talk slides from the
person who implemented it, and I'm not sure if and which ideas I might
have taken from it. It's possible that the timestamp reversal is
inspired by it, but I didn't check. (I think it claimed that it could
avoid large changes by changing a sign?)
VapourSynth has some sort of reverse function, which provides a backward
view on a video. The function itself is trivial to implement, as
VapourSynth aims to provide random access to video by frame numbers (so
you just request decreasing frame numbers). From what I remember, it
wasn't exactly fluid, but it worked. It's implemented by creating an
index, and seeking to the target on demand, and a bunch of caching. mpv
could use it, but it would either require using VapourSynth as demuxer
and decoder for everything, or replacing the current file every time
something is supposed to be played backwards.
FFmpeg's libavfilter has reversal filters for audio and video. These
require buffering the entire media data of the file, and don't really
fit into mpv's architecture. It could be used by playing a libavfilter
graph that also demuxes, but that's like VapourSynth but worse.
The ytdl wrapper can resolve web links to playlists. This playlist is
passed as big memory:// blob, and will contain further quite normal web
links. When playback of one of these playlist entries starts, ytdl is
called again and will resolve the web link to a media URL again.
This didn't work if playlist entries resolved to EDL URLs. Playback was
rejected with a "potentially unsafe URL from playlist" error. This was
completely weird and unexpected: using the playlist entry directly on
the command line worked fine, and there isn't a reason why it should be
different for a playlist entry (both are resolved by the ytdl wrapper
anyway). Also, if the only EDL URL was added via audio-add or sub-add,
the URL was accessed successfully.
The reason this happened is because the playlist entries were marked as
STREAM_SAFE_ONLY, and edl:// is not marked as "safe". Playlist entries
passed via command line directly are not marked, so resolving them to
EDL worked.
Fix this by making the ytdl hook set load-unsafe-playlists while the
playlist is parsed. (After the playlist is parsed, and before the first
playlist entry is played, file-local options are reset again.) Further,
extend the load-unsafe-playlists option so that the playlist entries are
not marked while the playlist is loaded.
Since playlist entries are already verified, this should change nothing
about the actual security situation.
There are now 2 locations which check load_unsafe_playlists. The old one
is a bit redundant now. In theory, the playlist loading code might not
be the only code which sets these flags, so keeping the old code is
somewhat justified (and in any case it doesn't hurt to keep it).
In general, the security concept sucks (and always did). I can for
example not answer the question whether you can "break" this mechanism
with various combinations of archives, EDL files, playlists files,
compromised sites, and so on. You probably can, and I'm fully aware that
it's probably possible, so don't blame me.
Originally, vo_gpu/vo_opengl considered the case of Nvidia proprietary
drivers, which required vdpau/GLX, and Intel open source drivers, which
require vaapi/EGL. Since window creation and GPU context creation are
inseparable in mpv's internal API, it had to pick the correct API very
early, or hardware decoding wouldn't work. "x11probe" was introduced for
this reason. It created a GLX context (without showing the window yet),
and checked whether vdpau was available. If yes, it used GLX, if not, it
continued probing x11/EGL. (Obviously it couldn't always fail on GLX
without vdpau, which is why it was a separate "probe" backend.)
Years passed, and now the situation is different. Vdpau is dead. Nvidia
drivers and libavcodec now provide CUDA interop, which requires EGL, and
fixes some of the vdpau problems. AMD drivers now provide vaapi, which
generally works better than vdpau. Intel didn't change.
In particular, vaapi provides working HEVC Main10 support. In theory, it
should work on vdpau too, with quality reduction (no 10 bit surfaces),
but I couldn't get it to work.
So always prefer EGL. And suddenly hardware decoding works. This is
actually rather important, because HEVC is unfortunately on the rise,
despite shitty encoders and unoptimized decoders. The latter may mean
that hardware decoding works better than libavcodec.
This should have been done a long, long time ago.
Linux analog TV support (via tv://) was excessively complex, and
whenever I attempted to use it (cameras or loopback devices), it didn't
work well, or would have required some major work to update it. It's
very much stuck in the analog past (my favorite are the frequency tables
in frequencies.c for analog TV channels which don't exist anymore).
Especially cameras and such work fine with libavdevice and better than
tv://, for example:
mpv av://v4l2:/dev/video0
(adding --profile=low-latency --untimed even makes it mostly realtime)
Adding a new input layer that targets such "modern" uses would be
acceptable, if anyone is interested in it. The old TV code is just too
focused on actual analog TV.
DVB is rather obscure, but has an active maintainer, so don't remove it.
However, the demux/stream ctrl layer must go, so remove controls for
channel switching. Most of these could be reimplemented by using the
normal method for option runtime changes.
This is a straightforward parallel implementation of error diffusion
algorithms in compute shader. Basically we use single work group with
maximal possible size to process the whole image. After a shift
mapping we are able to process all pixels column by column.
A large ring buffer are allocated in shared memory to speed things up.
However the size of required shared memory depends linearly on the
height of video window (or screen height in fullscreen mode). In case
there is no enough shared memory, it will fallback to `--dither=fruit`.
The maximal allowed work group size is hardcoded as 1024. Ideally we
could query `GL_MAX_COMPUTE_WORK_GROUP_INVOCATIONS`. But for whatever
reason, it seems most high end card from nvidia and amd support only
the minimal required value, so I guess we can stick to it for now.
half of the materials we used were deprecated with macOS 10.14, broken
and not supported by run time changes of the macOS theme. furthermore
our styling names were completely inconsistent with the actually look
since macOS 10.14, eg ultradark got a lot brighter and couldn't be
considered ultradark anymore.
i decided to drop the old option --macos-title-bar-style and rework
the whole mechanism to allow more freedom. now materials and appearance
can be set separately. even if apple changes the look or semantics in
the future the new options can be easily adapted.
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.
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.
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.
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).
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.
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 017f3d0674
