This is more or less a minimal hack to make _some_ text measurement
functionality available to scripts. Since libass does not support such a
thing, this simply uses the bounding box of the rendered text.
This is far from ideal. Problems include:
- using a bitmap bounding box
- additional memory waste and/or flushing caches
- dependency on window size
- odd small deviations with different window sizes (run osd-test.lua and
resize the window after each timer update; the bounding boxes aren't
adjusted in an overly useful way)
- inability to query the size _after_ actual rendering
But I guess it's a start. Since I'm aware that it's crap, add a threat
to the manpage that this may be changed/removed again. For now, I'm
interested whether anyone will have use for it in its current form, as
it's an often requested feature.
When the demuxer cache read until the end of the stream, and was
finished and completely inactive, the cache properties were not updated
anymore via MP_EVENT_CACHE_UPDATE.
Unfortunately, many cache properties depend on the current playback
position, such as cache-duration or fw-bytes. This is especially visible
on the OSC. If everything was cached, seeking around didn't update the
displayed forward cache duration.
That means checking demuxer_reader_state.idle is not enough. You also
need to check whether the current playback position changed.
Fix this by explicitly using the current playback position, and update
the properties if it changed "enough". "Enough" is 1 second of media
time in this example, which may or may not be appropriate.
In general, this could probably be done better. There are many other
triggers that change the cache state and that are not covered. For now
I'm content with getting rid of the obvious problems.
I think the OSC problem in particular was caused by changing it from
polling to using property change notifications.
Apparently this bit-rotted a lot, and now causes more problems that it
helps. In particular picking up broadcom EGL instead of Mesa EGL will
break things. With RPI getting proper Mesa/DRM/V4L support, this problem
should solve itself as well.
Some filters may block the playloop for a longer time. For example, if a
decoder fails to decode anything and somehow just discards packets, the
filter graph would run (in a blocking manner) until all packets are
read, which could take a longer time if the demuxer thread is fast
enough.
Make it exit every 100ms. That should at least give the user a chance to
stop playback.
Filtering could run on a different thread, but I don't see much value in
doing that in the general case. It would just waste a thread. Although
being able to use mp_filter_graph_interrupt() would be slightly nicer
than such a timeout mechanism. Decoding in particular can actually use a
separate thread (--vd-queue-enable), but again, this is not enabled by
default, because it just wastes a thread.
Like the previous f_decoder_wrapper commit, this is probably a sin.
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.
p->log has a prefix set that gives some context and distinguishes audio
and video decoders. The "public" wrapper filter didn't use it, which is
a regression since commit a3823ce0e0.
Filtering is integrated into an event loop, which is something the
filter API user provides. To make interacting with the event loop
easier, and in particular to avoid filtering to block event handling,
add functions the event loop code can suspend filtering.
While we cannot actually suspend a single filter, it's pretty easy to
suspend the filter graph run loop itself, which is responsible for
selecting which filter to run next.
This commit shouldn't change behavior at all, but the functions will be
used in later commits.
This was forgotten.
Hardware decoding typically breaks immediately, because many hw decoding
APIs require allocating all surfaces in advance (and/or libavcodec was
not made flexible enough to add new surfaces later). If the queue is
large enough, it will run out of surfaces, fail decoding, and fall back
to software decoding. We consider this the user's fault.
--hwdec-extra-frames can be used to avoid this, if you have enough GPU
memory, and the needed number of frames is lower than the arbitrary
mpv-set maximum limit of that option.
I don't know what should happen when the same value is written to the
property. It seems that it would be more natural if it were ignored
(since that's also what is done with options now), but you could argue
the other way around as well. In any case, changing it silently could
leads to user scripts etc. breaking, so don't change it now.
Instead, add blabla to the manpage to put the responsibility on the
user, so when we suddenly change it one day, we can blame any breakages
on someone else.
See: #7501
This required libsmbclient, which is a heavy dependency, and as a
library, has all kinds of problems. For one, the API requires completely
unacceptable global state (in particular, leaks auth state), and is not
thread-safe (meaning concurrent reads to multiple files block each
other).
There are better replacements: you can use the Linux kernel's builtin
CIFS support, fusesmb, or contribute supoport for libdsm.
This commit fixes a bug where handle for a framebuffer gets double
freed.
It seems to happen that the same prime fd gets two framebuffers.
As the prime fd is the same the resulting prime handle is also the
same.
This means one handle but 2 framebuffers and can lead to the following
chain:
1. The first framebuffer gets deleted the handle gets also freed via
the ioctl.
2. In startup phase not all 4 dumb buffers for overlay drawing
are set up. It can happen that the last dumb buffer gets the
handle we freed above.
3. The second framebuffer gets freed and the handle will be
freed again resulting that the 4's dumb buffer handle is not
backed by a buffer.
4. Drm prime continues to assign handles to its prime fds an
will lead to have this handle which was just freed to
reassign again but to an prime buffer.
5.Now the overlay should be drawn into dumb buffer 4 which
still has the same handle but is backed by the wrong buffer.
This leads to two different behaviors:
- MPV crashes as the drm prime buffers size als calculated
by the decoder output format. The overlay output format
differs and it takes more space. SO the size check
in kernel fails.
- MPV is continuing play. This happens when the decoders
allocates a bigger buffer than needed for the overlay.
For example overlay is Full HD and decoder output is 4k.
This leads to the behavior das the overlay wil be drawn
into the wrong buffer as its a drm prime buffer and results
in a flicker every fourth step.
Instead of having f_decoder_wrapper create its own copy of the entire
mpv option tree, create a struct local to that file and move all used
options to there.
movie_aspect is used by the "video-aspect" deprecated property code. I
think it's probably better not to remove the property yet, but
fortunately it's easy to work around without needing special handling
for this option or so.
correct_pts is used to prevent use of hr-seek in playloop.c. Ignore
that, if you use --no-correct-pts you're asking for trouble anyway. This
is the only behavior change.
The demuxer cache employs a strange method to make track switching
instant with caching enabled. Normally this would mean you have to wait
until the cache has played out (and you get new packets, including
packets from the newly selected track), or you have to perform a slow
high level seek (decoding video again etc.). The strange method is that
it performs a demuxer-level seek without a high level seek so it looks
like a continuous stream to the decoder, and the newly select stream
gets packets at the current playback position. This is called a refresh
seek.
This works only if some weird heuristics work. It needs a packet "unique
ID", for which it uses either dts or pts. The value must be strictly
monotonic increasing. If this doesn't work, the referesh seek can't be
executed, and the user has to wait until the end of the cache. Sometimes
there are files that simply do not work.
In the present case, there's actually a hack that we can extend. Packets
with unset position are likely generated by the parser, and the hack
which this commit touches simply attempts to make up a new (hopefully
unique) position value, even if the value itself makes no sense. It only
ha to be deterministic.
It turns out libavcodec sometimes output packets with repeating position
values. This commit tries to handle this case too with the same hack.
Fixes: #7498
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
This is supposed to enable communication between filter graphs on
separate threads. Having multiple threads makes only sense if they can
run concurrently with each other, which requires such an asynchronous
queue as a building block. (Probably.)
The basic idea is that you have two independent filters, which can be
each part of separate filter graphs, but which communicate into one
direction with an explicit queue. This is rather similar to unix pipes.
Just like unix pipes, the queue is limited in size, so that still some
data flow control enforced, and runaway memory usage is avoided.
This implementation is pretty dumb. In theory, you could avoid avoid
waking up the filter graphs in quite a lot of situations. For example,
you don't need to wake up the consumer filter if there are already
frames queued. Also, you could add "watermarks" that set a threshold at
which producer or consumer should be woken up to produce/consume more
frames (this would generally serve to "batch" multiple frames at once,
instead of performing high-frequency wakeups). But this is hard, so the
code is dumb. (I just deleted all related code when I still got
situations where wakeups were lost.)
This is actually salvaged and modified from a much older branch I had
lying around. It will be used in the next commit.
Instead of vague ideas about making different filter graphs on different
threads interact directly, this have no direct support. Instead, helpers
are required (such as added with the next commit).
Document it. Different root filters (i.e. separate filter graphs) are
now considered to be part of separate threads, so assert() if they're
found to accidentally interact.
Previously: 5.1 > 5.2 > luajit
Now: 5.2 > luajit > 5.1
I randomly decided that this should be done, since I suspect most
environments will prefer the highest Lua version anyway. There is not
much of a point picking the older one by default.
Maybe 5.1 should be dropped fully, but considering we need to stay
compatible with luajit, there is no particular incentive for this.
Preparation for a future commit. The demuxer queues might be read from
other threads than the one to issue the seek, and passing SEEK_BLOCK
with such a seek will provide a convenient way to synchronize this.
mp_filter_mark_async_progress() can asynchronously mark a filter for
processing, without waking up the filter thread. (It's some sort of
idiotic micro-optimization I guess?) But since it sets async_pending
without doing the wakeup, a mp_filter_wakeup() after this will do
nothing, and the wakeup is lost. Fix it by checking for the needed
wakeup separately.
Fortunately, nothing used this function yet, so there is no impact.
I may (optionally) move decoding to a separate thread in a future
change. It's a bit attractive to move the entire decoder wrapper to
there, so if the demuxer has a new packet, it doesn't have to wake up
the main thread, and can directly wake up the decoder. (Although that's
bullshit, since there's a queue in between, and libavcodec's
multi-threaded decoding plays cross-threads ping pong with packets
anyway. On the other hand, the main thread would still have to shuffle
the packets around, so whatever, just seems like better design.)
As preparation, there shouldn't be any mutable state exposed by the
wrapper. But there's still a large number of corner-caseish crap, so
just use setters/getters for them. This recorder thing will inherently
not work, so it'll have to be disabled if threads are used.
This is a bit painful, but probably still the right thing. Like
speculatively pulling teeth.
This explicitly depends on the pixfmt list from FFmpeg (done so to
easily spot regression, incompatible changes, and other unexpected
things).
Some local changes in mpv change some of the output. For pal8 an alias
was added back, and the [GENERIC] markers are removed because the mpv
aliases are not dependent on the mpv config anymore (which was
unnecessary).
The other changes are due to ffmpeg adding some new formats.
This was a hack that attempted to line up external audio tracks with
video. The problem is that if you do a keyframe seek backwards, video
will usually seek much farther back than audio (due to much higher
keyframe aka seek point distances). The hack somehow made seeking a 2
step process.
This existed in 4 different forms in the history of this code base, and
it was always very cumbersome. We mostly needed this for ytdl_hook (I
think?), which uses the 4th form, which is nicely confined to
demux_timeline and is unrelated to the "external" audio tracks in the
high level player.
Since this is (probably) not really widely needed anymore, get rid of
it. Better do this now, than when somehow rewriting all the seeking code
(which might happen in this decade or the next or so) and when it
wouldn't be easily revertable anymore in case we find we "really" need
it unlike expected.
There is no issue if hr-seeks are used. Also, you can still use edl
files to "bundle" multiple streams as if it was a single stream (this is
what ytdl_hook does now).
It seems sporadic errors are possible, such as connection timeouts.
Before the recent demuxer change, the demuxer thread retried many times
even on EOF, so an error was only interpreted as EOF once the decoder
queues ran out.
Change it to use EOF only. Since this may actually lead to the demuxer
thread being "stuck" and retrying forever (depending on libavformat API
behavior), I'm also adding a heuristic to prevent this, using a random
retry counter. This should not be necessary, but libavformat cannot be
trusted. (This retrying forever could be stopped by the user, but
obviously it would still heat the CPU for a longer time if the user is
not present.)
The "seekbarkeyframes" option is now interpreted such if it's true, the
player default is used. Too lazy to make this a choice option or
whatever; the Lua option parser doesn't have support for that anyway.
Someone who cares can adjust this.
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 tries to fix#7206 (hr-seeking past EOF does not stop playback)
again. Commit 57fbc9cd76 should have fixed this, but trying it again
(using that git revision), it often did not work. Whatever the fuck.
So add another dumb special case that will break within weeks. Note that
the check in handle_eof() had no effect, since execute_queued_seek() is
called later, which cancels EOF in the same case.
Hr-seek past the last frame instantly enters EOF, which means
handle_playback_time() will not set playback_pts to the video PTS (as
all video frames are skipped), which leads to the playback time being
taken from the last seek target. This results in confusing behavior,
especially since the seek time will be clipped to the file duration for
display, but not for further relative seeks.
Obviously, the time should be set to the last video frame, so use the
last video frame as fallback if both audio and video have ended. Also,
since the same problem exists with audio-only playback, add a fallback
for audio PTS too. We don't know which was the "last" fragment of media
played (to decide whether to use the audio or video PTS as the
fallback), but it doesn't matter since the maximum works.
This could lead to some undesired effects. In particular the audio PTS
is basically a bad guess, and is for example not clipped against --end.
(But the ridiculous way audio syncing and clamping currently works, I'm
not going to touch that shit unless I rewrite it completely.) The cover
art case is slightly broken: using --keep-open with keyframe seeks will
result in 0 as playback PTS (the video PTS). OK, who cares, it got late.
Also casually get rid of last_vo_pts, since that barely made any sense
at all.
Fixes: #7487
The seeking logic saves the last video frame it has seen (for example
for being able to seek to the last frame, or backstepping).
Unfortunately, the frame was fed back to the filtering pipeline in
situations when it shouldn't have. Then it's an out of order frame,
because it really saves the last _discarded_ frame.
For example, seeking to the end of a file with --keep-open, shift+up,
shift+down => invalid video pts warning due to saved_frame being fed
back.
Explicitly discard saved_frame when it's obviously not needed anymore.
The removed accesses to "r" are strictly speaking unrelated (just
const-propagating them).
In this case the video track has seek_start == seek_end, and due to the
"seek_start >= seek_end" condition, this was considered broken, and no
seek range was created, breaking cached seeking.
Fix this by allowing the case if they're equal, and a valid timestamp.
(NB: seeking backward in this will still jump to position 0, because it
is the video timestamp. This is unfortunately how it's supposed to work.
HR-seeks will also do this, but decode and skip the entire audio until
the seek target, so it will mostly appear to work.)
Exposed by commit b56e2efd5f. demux_timeline reported a bogus EOF if
"parallel" streams were used. If a virtual source reported EOF, it was
propagated as global EOF, without serving packets of other virtual
sources that have not ended yet.
Fix this by not reporting global EOF just because a source has not
returned a packet. Instead make the reader retry by returning no packet
and no EOF state, which will call d_read_packet() again with a different
source. Rely on the eof_reached flags to signal global EOF.
Since eof_reached is now more important, set it in a certain other case
when it apparently should have been set. do_read_next_packet()'s return
value is now ignored, so get rid of 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.
Setting demux_set_stream_wakeup_cb() will make all sh_stream (i.e.
track) specific wakeups go to this callback. But the callback takes care
of only the sub_preload() case (where it tries to pre-load subtitles
from already parsed and memory-present subtitles in a blocking way).
The old code assumed that the normal demuxer wakeup callback is called.
This was disregarded when the newer code was added. (And actually, the
original plan was to make _all_ per-sh_stream wakeups go to specialized
callbacks to avoid wasted work. dec_sub really should set the callback
always, and propagate wakeups to the playloop code. But it's too far
into the night to write coherent code.)
I couldn't actually observe any manifestation of this bug. Normally, the
playloop wakes up for other reasons (such as driving audio and video
decoding), so the lost wakeups rarely matter.
wm4
Sven Kroeger
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