--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.
wm4
Guido Cella
Chris Varenhorst
rcombs
Dudemanguy
Niklas Haas
RealDolos
James Ross-Gowan
Avi Halachmi (:avih)
mg