| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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Requested.
Fixes: #6303
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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
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Forgotten in one of the previous commits. Also undeprecates
display-adrop since it's out of sight now.
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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.
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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.
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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
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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
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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
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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
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Lets the application fully control the rendering onto the screen
instead of the compositor.
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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
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This was deprecated 2 releases ago. The deprecation changelog entry says
that there are no plans to remove it short-term, but I guess I lied.
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And document that it's ignored with bilinear scaler.
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Because it's confusing and useless. If nobody complains, we'll have one
weird cache configuration option less.
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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.
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Believe it or not, users were confused about this.
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Why not.
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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
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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
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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).
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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.
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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.
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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.
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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.
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Make it a proper list, instead of a paragraph soup.
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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.
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In particular, all_formats description split away the example section of
an unrelated option, so move that to its proper place.
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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.
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Might be helpful for... whatever.
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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.
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Cut the arbitrary offset, and document what unit/timesource it uses.
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options.rst to clarify the option, some more text in mpv.rst to separate
out the compatibility stuff a little.
Fixes: #7461 (options.rst change only)
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This seems stable enough to use. Change the default, and remove it from
the sw-fast profile.
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Of course nobody will read this. I'm just putting it there so I can
blame users, who run into problems, for not having read it.
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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
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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 #7396
Fixes #2547
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As requested. I guess option name and manpage text could be better and
clearer.
Closes: #7442
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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".
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mpv actually used to be able to, from what I remember, but this was
changed for simplicity and because of problems with FFmpeg.
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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.
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was forgotten in commit 3275cd0
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See manpage additions. The libarchive behavior mentioned in the last
paragraph there is technically unrelated, but makes this new option
mostly pointless.
See: #7182
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The text here has become somewhat outdated over the years, and it's
worth updating to reflect the current situation.
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