This is for text subtitles. libavformat currently always reads text
subtitles completely on init. This means the underlying stream is
useless and will consume resources for various reasons (network
connection, file handles, cache memory).
Take care of this by closing the underlying stream if we think the
demuxer has read everything. Since libavformat doesn't export whether it
did (or whether it may access the stream again in the future), we rely
on a whitelist. Also, instead of setting the stream to NULL or so, set
it to an empty dummy stream. This way we don't have to litter the code
with NULL checks.
demux_lavf.c needs extra changes, because it tries to do clever things
for the sake of subtitle charset conversion.
The main reason we keep the demuxer etc. open is because we fell for
libavformat being so generic, and we tried to remove corresponding
special-cases in the higher-level player code. Some of this is forced
due to ass/srt mkv/mp4 demuxing being very similar to external text
files. In the future it might be better to do this in a more
straight-forward way, such as reading text subtitles into libass and
then discarding the demuxer entirely, but for aforementioned reasons
this could be more of a mess than the solution introduced by this
commit.
Probably fixes#3456.
Cleaner and makes it easier to change the underlying stream.
mp_property_stream_capture() still directly accesses it directly via
demux_run_on_thread(). This is evil, but still somewhat sane and is not
getting into the way here.
Not sure if I got all field accesses.
It's just wasted memory.
One corner case is when a file grows during playback, but this is rare
and usually happens on-disk only. The cache size was generally limited
before this change already, so no reason to care.
As an unrelated change, move the cache size info to the resize_cache()
function. There's really no reason not to do this, and it's slightly
more informative if the user changes the cache size at runtime.
Because VOCTRL_CHECK_EVENTS is processed asynchronously (as of 088a007,)
the GUI thread no longer gets regular wakeups, so the old check that
made sure the video window matched the parent window's size in --wid
embedding mode did not run very often. This made --wid embedding not
very usable.
Instead of polling for window size changes, use Windows hooks to react
to them when they happen. When the parent window is owned by the same
process as the video window, use a WH_CALLWNDPROC hook. When the parent
window is not owned by the same process, WinEvents must be used, which
are not as smooth, but still work for this purpose.
Since neither SetWindowsHookEx nor SetWinEventHook take a context
parameter to send data to the hook function, the hook functions must
find the child window by its class instead, so there are a few changes
to ensure this is fast and the class is unique.
This also fixes up the logic to handle window destruction. When a parent
window is destroyed, its children are also destroyed, so this gives us a
way to react to parent window destruction without polling.
If the video has the same size as the screen, starting with --fs and
then leaving fullscreen doesn't actually leave fullscreen.
The reason is that mpv tries to restore the previous window size if
necessary (otherwise, you'd end up with a Window of nearly the same size
as the screen with some WMs). It will typically restore with the
rectangle set exactly to the screen if no other position or size is
forced. This triggers pre-EWMH fullscreen mode, which WMs detect using
various heuristics.
Apparently we triggered this with mutter (but strangely no other WMs).
It's possible that pre-EWMH fullscreen mode actually requires removing
decorations, and mutter either ignores this. But this is speculation and
I haven't checked.
Work this around by reducing the requested size by 1 pixel if it
happens.
This was observed with mutter 3.18.2.
Fixes#2072.
If spdif is enabled, the channel layout has no meaning other than
setting the number of channels. The number of channels must be fixed to
achieve the exact bitrate required.
Fixes#3445.
It doesn't necessarily have to mean anything bad.
We're still too lazy to provide any more detailed information (e.g.
whether this happened to likely bad interleaving, excessive amount of
packets like with some ASS subs, or that the readahead user option is
limitted by the packet queue size).
This should actually be rather safe - we already check whether the
estimated value jitters less than the (possibly untrustworthy) nominal
one. Remove a "safety" check that disabled this code for small
deviations, and make it trigger sooner into playback. Also lower the log
level of messages about using the estimated display FPS down to verbose.
Normally there's another mechanism for smoothing out minor estimation
differences, but that is not good enough here.
This possibly improves behavior as reported in #3433, which can be
reproduced with --vo=null:fps=48.426 --display-fps=48 (though it doesn't
consider the jitter introduced by a real VO).
Doing this required synchronizing with the VO thread, which could lead
to audio dropouts if the VO was frozen (which can happen in practice if
e.g. an opengl_cb user is not doing what the API demands).
Add a way to send asynchronous VOCTRLs, and use that for the playback
state. In theory, it would be better to make this status update a
several function and to "merge" several queued update, but that would be
slightly more effort/code, and the update is so infrequent that the
merging would never happen anyway.
The change to vo_destroy() is to make sure all queued asynchronous
reuqests are finished before making the vo_thread exit.
Even though it's only used on MS Windows, it's run on any platform with
any VO, which makes this worse.
run_control() dereferences an uint32_t as int. Whether this is allowed
depends on what uint32_t is typedefed to (dereferencing an unsigned int
as int should be fine). Fix it by always using int. The uint32_t type
never really made sense.
There are situations where the resampler is destroyed and recreated
during playback. If recreating the resampler unexpectedly fails, the
filter function is supposed to return an error. This wasn't done
correctly, because get_out_samples() accessed the resampler before the
check. Move the check up to fix this.
When fetching the playlist property, playlist_entry_from_index would be
called for each playlist entry, which traversed a linked list to get the
entry corresponding to the specified index. This was very slow for large
playlists. Since get_playlist_entry is called for each index in order,
it can avoid a full traversal of the linked list by using the next
pointer on the previously requested entry.
Instead of passing through double float timestamps opaquely, pass real
timestamps. Do so by always setting a valid timebase on the
AVCodecContext for audio and video decoding.
Specifically try not to round timestamps to a too coarse timebase, which
could round off small adjustments to timestamps (such as for start time
rebasing or demux_timeline). If the timebase is considered too coarse,
make it finer.
This gets rid of the need to do this specifically for some hardware
decoding wrapper. The old method of passing through double timestamps
was also a bit questionable. While libavcodec is not supposed to
interpret timestamps at all if no timebase is provided, it was
needlessly tricky. Also, it actually does compare them with
AV_NOPTS_VALUE. This change will probably also reduce confusion in the
future.
This affects A-B loops and --loop-file, and audio. Instead of dropping
audio by resetting the AO, try to make it seamless by not sending data
after the loop point, and after the seek send new data without a reset.
The code actually kept going out of EOF mode into resync mode back into
EOF mode when the playloop had to wait after an audio EOF caused by the
endpts. This would break seamless looping (as added by the next commit).
Apply endpts earlier, to ensure the filter_audio() function always
returns AD_EOF in this case.
The idiotic ao_buffer makes this an amazing pain in the ass.
Instead of letting it keep decoding by trying to find a new frame,
"plug" the frame queue by not removing it. (Or actually, by putting
it back instead of discarding it.)
Matters for seamless looping (following commits), and possibly some
other corner cases.
The added function vf_unread_output_frame() is a bit of a sin, but still
reasonable, since its implementation is trivial.
The --image-display-duration option controls how long an image is
displayed. It's also possible to display the image forever (until manual
user interaction stops playback).
With this, the core drops the old method to "drain" video (i.e. waiting
for the last frame duration on end of playback). Instead, we reuse
MPContext.time_frame. The old mechanism was disabled for non-images
anyway.
Fixes#3425.
When an ogg track upodates metadata, we have to perform a complicated
runtime update due to the demux.c architecture. A detail was broken and
an array was allocated with the previous number of streams, which
usually led to invalid memory write accesses at least on the first
update.
See github commit comment on commit b9ba9a89.
The touched code is for seek resets and such - we simply want to reset
the entire resample state. But I noticed after a seek a tiny bit of
audio is missing (mpv's audio sync code inserted silence to compensate).
It turns out swr_drop_output() either does not reset some internal state
as we expect, or it's designed to drop not only buffered samples, but
also future samples.
On the other hand, libavresample's avresample_read(), does not have this
problem. (It is also pretty explicit in what it does - return/skip
buffered data, nothing else.)
Is the libswresample behavior a bug? Or a feature? Does nobody even
know? Who cares - use the hammer to unfuck the situation. Destroy and
deallocate the libswresample context and recreate it. On every seek.
Change the last parameter from a bool to an int, which is supposed to
take bit-flags. The at this point only flag is MPSEEK_FLAG_DELAY, which
replaces the previous bool parameter. The old false parameter becomes 0,
the old true parameter becomes MPSEEK_FLAG_DELAY.
Since the old "immediate" parameter is now essentially inverted, two
coalesced immediate and delayed seeks end up as delayed instead of
immediate. This change doesn't matter, since there are no relative
immediate seeks anyway.
The accepts_packet packet callback is supposed to deal with subtitle
decoders which have only a small queue of current subtitle events (i.e.
sd_lavc.c), in case feeding it too many packets would discard events
that are still needed.
Normally, the number of subtitles that need to be preserved is estimated
by the rendering pts (get_bitmaps() argument). Rendering lags behind
decoding, so normally the rendering pts is smaller than the next video
frame pts, and we simply discard all subtitle events until the rendering
pts.
This breaks down in some annoying corner cases. One of them is seeking
backwards: the VO will still try to render the old PTS during seeks,
which passes a high PTS to the subtitle renderer, which in turn would
discard more subtitles than it should. There is a similar issue with
forward seeks. Add hacks to deal with those issues.
There should be a better way to deal with the essentially unknown
"rendering position", which is made worse by screenshots or rendering
with vf_sub. At the very least, we could handle seeks better, and e.g.
either force the VO not to re-render subs after seeks (ugly), or
introduce seek sequence numbers to distinguish attempts to render
earlier subtitles when a seek is done.
If the PEAK tag is invalid, return an error.
Make the error signalling conventions more uniform by strictly returning
a negative value on error, and treating >=0 as success.
The demuxer layer usually doesn't log per-stream information, and even
the replaygain information was logged only if it came from tags.
So log it in af_volume instead.
...and ignore it. The main purpose is for retrieving per-track
replaygain tags. Other than that per-track tags are not used or accessed
by the playback core yet.
The demuxer infrastructure is still not really good with that whole
synchronization thing (at least in part due to being inherited from
mplayer's single-threaded architecture). A convoluted mechanism is
needed to transport the tags from demuxer thread to user thread. Two
factors contribute to the complexity: tags can change during playback,
and tracks (i.e. struct sh_stream) are not duplicated per thread.
In particular, we update the way replaygain tags are retrieved. We first
try to use per-track tags (common in Matroska) and global tags
(effectively formats like mp3). This part fixes#3405.
With the previous commit, ao_alsa.c now has 3 possible ways to pause
playback. Actually all 3 of them need get_delay() to fake its return
value, so don't duplicate that code.
Also much of the code looks a bit questionable when considering
inconsistent pause/resume calls from outside, so ignore redundant calls.
push.c does not handle this automatically, and AOs using push.c have to
handle it themselves. Also, ALSA is low-level enough that it needs
explicit support in user code. At least I haven't found any option that
does this.
We still can get away relatively cheaply by abusing underflow-handling
for this. ao_alsa.c already configures ALSA to handle underflows by
playing silence. So we purposely induce an underflow when opening the
device, as well as when pausing or resetting the device.
This introduces minor misbehavior: it doesn't account for the additional
delay the initial silence adds, unless the device has fully played the
fragment of silence when the player starts sending data to it. But
nobody cares.
The current stdatomic check verifies the availability of the function by
calling atomic_load(). It also uses this test to check if linking
against libatomic is needed or not.
Unfortunately, on specific architectures (namely SPARC), using
atomic_load() does *not* require linking against libatomic, while other
atomic operations do. Due to this, mpv's wscript concludes that
stdatomic is available, and that linking against libatomic is not
needed, causing the following link failure:
[190/190] Linking build/mpv
audio/out/ao.c.13.o: In function `ao_query_and_reset_events':
/home/peko/autobuild/instance-0/output/build/mpv-0.18.1/build/../audio/out/ao.c:399: undefined reference to `__atomic_fetch_and_4'
In order to fix this, the stdatomic check is adjusted to call
atomic_fetch_add() instead, which does require libatomic. Thanks to
this, the wscript realizes that linking against libatomic is needed, and
the build works fine.
Signed-off-by: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
If the normal stream cache init fails, and a file cache was initialized
before, we free the file cache as well. But since the file cache is
chained to the real stream, the real stream will also be freed. This has
to be prevented by clearing the pointer to the original stream in the
uncached_stream field.
This could in particular be triggered by using --cache-initial=1000 and
aborting playback during loading. (Without that option, stream cache
init failure is far less likely.)
Forgotten in previous commit.
Also minor semi-related change: remove the extra "," from the
mpv_sub_api enum, which I accidentally added in the previous commit.
(C99 is fine with trailing ",", C89 strictly speaking not. So do
this for maximum compatibility.)
Relative seeks backwards with external audio tracks does not always work
well: it tends to happen that video seek back further than audio, so
audio will remain silent until the audio's after-seek position is
reached. This happens because we strictly seek both video and audio
demuxer to the approximate desirted target PTS, and then start decoding
from that.
Commit 81358380 removes an older method that was supposed to deal with
this. It was sort of bad, because it could lead to the playback core
freezing by waiting on network.
Ideally, the demuxer layer would probably somehow deal with such seeks,
and do them in a way the audio is seeked after video. Currently this is
infeasible, because the demuxer layer assumes a single demuxer, and
external tracks simply use separate demuxer layers. (MPlayer actually
had a pseudo-demuxer that joined external tracks into a single demuxer,
but this is not flexible enough - and also, the demuxer layer as it
currently exists can't deal with dynamically removing external tracks
either. Maybe some time in the future.)
Instead, add a gross hack, that essentially reseeks the audio if it
detects that it's too far off. The result is actually not too bad,
because we can reuse the mechanism that is used for instant track
switching. This way we can make sure of the right position, without
having to care about certain other issues.
It should be noted that if the audio demuxer is used for other tracks
too, and the demuxer does not support refresh seeking, audio will
probably be off by even a higher amount. But this should be rare.
This code is for resyncing audio-only streams (e.g. switching between
audio tracks if no video track is active). This must not be run if the
video PTS just isn't known yet. (Although the case in which this changes
anything is probably very obscure, if it can even happen. Still, it's a
bit more correct.)
This is a correction to commit 91a3bda6.
In display-sync mode, the very first video frame is idiotically fully
timed, even though audio has not been synced yet at this point, and the
video frame is more like a "preview" frame. But since it's fully timed,
an underflow is detected if audio takes longer than the display time of
the frame (we send the second frame only after audio is done).
The timing code will try to compensate for the determined desync, but it
really shouldn't. So explicitly discard the timing info in this specific
case. On the other hand, if the first frame still hasn't finished
display, we can pretend everything is ok.
This is a hack - ideally, we either would send a frame without timing
info (and then send it again or so when playback starts properly), or we
would add real pause support to the VO, and pause it during syncing.
Play a trick to make the packet pos field monotonically increasing over
segment boundaries if the source demuxers return monotonically
increasing pos values. This allows the demuxer to uniquely identify
packets with the pos field, and can do refresh seeks using that.
Normally, the packet pos field is used as a fallback for determining the
playback position if the demuxer returns no proper duration. But
demux_timeline.c always will, and the packet pos fields usually make no
sense in relation to the returned file size anyway if the timeline
source demuxers originate from separate streams.
Remove the explicit whitelisting of formats for refresh seeks. Instead,
check whether the packet position is somewhat reliable during demuxing.
If there are packets without position, or the packet position is not
monotonically increasing, then do not use them for refresh seeks.
This does not make sure of some requirements, such as deterministic
seeks. If that happens, mpv will mess up a bit on stream switching.
Also, add another method that uses DTS to identify packets, and prefer
it to the packet position method. Even if there's a demuxer which
randomizes packet positions, it hardly can do that with DTS. The DTS
method is not always available either, though. Some formats do not have
a DTS, and others are not always strictly monotonic (possibly due to
libavformat codec parsing and timestamp determination issues).
If an audio track is enabled during playback, then make it resume at the
exact "current position", instead of playing audio before that position.
This was already done for video.
If the packet read function returns, and EOF was detected, and a seek
was issued in the meantime, then don't use the EOF result. The seek will
be processed later, and reset the EOF state anyway.
The main effect is probably that we don't return EOF to the decoders
(which the playback core resets before issuing the seek), and that we
won't log an EOF message.
Not important, but slightly more correct.
When switching tracks, we normally have the problem that data gets lost
due to readahead buffering. (Which in turn is because we're stubborn and
instruct the demuxers to discard data on unselected streams.) The
demuxer layer has a hack that re-reads discarded buffered data if a
stream is enabled mid-stream, so track switching will seem instant.
A somewhat similar problem is when all tracks of an external files were
disabled - when enabling the first track, we have to seek to the target
position.
Handle these with the same mechanism. Pass the "current time" to the
demuxer's stream switch function, and let the demuxer figure out what to
do. The demuxer will issue a refresh seek (if possible) to update the
new stream, or will issue a "normal" seek if there was no active stream
yet.
One case that changes is when a video/audio stream is enabled on an
external file with only a subtitle stream active, and the demuxer does
not support rrefresh seeks. This is a fuzzy case, because subtitles are
sparse, and the demuxer might have skipped large amounts of data. We
used to seek (and send the subtitle decoder some subtitle packets
twice). This case is sort of obscure and insane, and the fix would be
questionable, so we simply don't care.
Should mostly fix#3392.
Always require them, instead of just for some components which have hard
requirements on correct atomic semantics. They should be widely
available, and are supported by all recent gcc and clang compiler
versions. We even have the fallbacks builtins, which should keep this
working on very old gcc releases.
In particular, w32_common.c recently added a hard requirement on
atomics, but checking this properly in the build system would have been
messy. This commit makes sure it always works.
The fallback where weak atomic semantics are always fine is in theory
rather questionable as well.
Exactly the same situation as with ao_alsa in commit 0b144eac (except
that we can detect the situation better under wasapi).
Essentially, wasapi will allow us to output any sample format, and not
just the one configured by the user in the audio system settings.
VOCTRL_CHECK_EVENTS is called on every frame. This is by design, and is
supposed to check the event queue of the windowing API.
With the decoupled GUI thread in w32_common.c this doesn't make too much
sense, and the purpose of VOCTRL_CHECK_EVENTS is really reduced to
checking event flags. Even worse, waiting on the GUI thread can
interfere with playback, since win32 sometimes blocks the event loop
(e.g. clicking the window title bar).
Change the code such that we really only query the event flags. Use
atomics to avoid having to add a new mutex. (We assume we always have
real atomics available. The build system doesn't check this properly,
and it could fall back to dummy atomics, which are not atomic.)
Should help with #3393. Doesn't help if the core happens to send a
synchronous request, most commonly via VOCTRL_SET_CURSOR_VISIBILITY or
VOCTRL_UPDATE_PLAYBACK_STATE.
This commit adds an --audio-channel=auto-safe mode, and makes it the
default. This mode behaves like "auto" with most AOs, except with
ao_alsa. The intention is to allow multichannel output by default on
sane APIs. ALSA is not sane as in it's so low level that it will e.g.
configure any layout over HDMI, even if the connected A/V receiver does
not support it. The HDMI fuckup is of course not ALSA's fault, but other
audio APIs normally isolate applications from dealing with this and
require the user to globally configure the correct output layout.
This will help with other AOs too. ao_lavc (encoding) is changed to the
new semantics as well, because it used to force stereo (perhaps because
encoding mode is supposed to produce safe files for crap devices?).
Exclusive mode output on Windows might need to be adjusted accordingly,
as it grants the same kind of low level access as ALSA (requires more
research).
In addition to the things mentioned above, the --audio-channels option
is extended to accept a set of channel layouts. This is supposed to be
the correct way to configure mpv ALSA multichannel output. You need to
put a list of channel layouts that your A/V receiver supports.
Pointless anyway. With superficial checking I couldn't find any decoder
which actually outputs this, and AO chmap negotiation would properly
ignore them anyway in most cases.
Assume you use a large value like --audio-delay=20. Then until now the
player would just have seeked normally to a "too late" position, and
played silence for about 20 seconds until audio in the correct time
range is coming again.
Change this by offsetting seeks by the right amount. This works for both
external and muxed files. If a seek isn't precise, then it works only
for external files.
This might cause issues with very large delay options. Hr-seek skipping
could take a lot of time (especially because it affects video too), the
demuxer queue could overflow, and other weird corner cases could appear.
But we just try this on best-effort basis, and if the user uses extreme
values we don't guarantee good behavior.
Prevents segfaults when a fullscreen switch is issued before fully
initializing the VO.
Doesn't change anything since the schedule_resize is only there to
resize in case the image size switches, which happens long after init.
We send a refcounted frame to the encoder, but then disrespect
refcounting rules and write to the frame data without making sure the
buffer is really writeable.
In theory this can lead to reallocation on every frame is the encoder
really keeps a reference. If we really cared, we could fix this by
providing a buffer pool. But then again, we don't care.
This is logical: the function makes sense only in situations where you
are going to write to the audio data. To make it worse,
av_buffer_realloc() also handles this situation, but only if the buffer
size changes (simply because it can't realloc memory in use), so we have
to explicitly force reallocation by unreffing the buffers first.
The problem was that when in fullscreen, switching between images did
not issue a resize event, causing none of the images to be rendered
correctly.
This fixes the problem by issuing a resize event with the screen width
and height.
This commit also moves the zeroing of the events field to when it gets
retrieved by mpv rather than randomly after a resize in the vo/backend
code.
ssurface_handle_configure()'s width and height are just hints given by
the compositor, the application's free to not respect those strictly and
to compensate for e.g. aspect ratio.
This prevents crazy scenarios in which pictures with portrait aspect
ratios have a huge black area to make them 16:9 or whatever the
compositor feels like.
With X11 it was usually left up to the window manager to prevent huge
windows from being out of range, but no Wayland compositor will do
this right now.
Hugely improves usability when using mpv as an image viewer.
Missed during the recent changes.
Also simplify error checking code and check for POLLNVAL
as well (the display fd was never actually checked to be valid).
This requires changing the pixel upload alignment because the odd sizes
might not be aligned to multiples of 4.
Anyway, the restriction has no real benefit and the sizes in between 32
and 64 might be worth using, so just drop it.
Following testing after ebe798a, this is a more than sufficient size to
cover our use case.
The old default was a drop of about 58 dB PSNR using the old code, and
this new default is about 65 dB PSNR, so it's actually an improvement
despite resulting in a smaller size.
There was no outlier whatsoever when comparing sizes around the 64
neighbourhood (with every step corresponding to a PSNR drop of about
0.07 dB), so I picked this since it's a power of two and requires no
change to the current 3dlut-size parsing logic.
I also tested smaller sizes such as 32x32x32 which performed almost as
well on colorful samples, but this results in noticeable black boost in
the dark regions, which is pretty undesirable. Therefore, we should
avoid going much further below 64x64x64.
Either way, this new size is so fast to compute that the 3dlut cache is
almost useless on my end. In fact, it might even be slower to load the
profile from the cache than to recompute it from scratch. (For caches on
a disk. For cache on a tmpfs, it makes no difference)
It seems vo_x11_check_events() was supposed to return the currently
flagged events and reset them. But there are many places where
vo_x11_check_events() is called without checking its return value. This
could lead to forgotten events.
Change the code such that they can't get lost.
This code had the exact same texture indexing bug that the original
scaler code had before the introduction of the LUT_POS macro to fix it.
We can re-use this same macro here, and the performance drop is
virtually entirely negligible. The benefit is greatly improved LUT
accuracy as the 3DLUT size decreases - in particular, the old LUT
started introducing more and more black crush the lower your LUT size is
(because the error was essentially an over-contrast bias, with a
magnitude linearly related to the lut size).
The new code improves black stability as the LUT size decreases, and
only at very low values (16 and below) do black levels start noticeably
getting affected (due to crude linearization of the nonlinear response
curve).
The default value of 3dlut-size is definitely generous enough for this
to make no difference out of the box, but it also causes no performance
drop at all on my machine so I see no harm in improving the logic.
Furthermore, this means we could easily decrease the default 3dlut size
in a future commit, perhaps even down to 64x64x64 as a default. (But
more testing is warranted here)
The FFmpeg API is incredibly weird and inconsistent about this. This is
also a FFmpeg-only issue and nothing like this is in Libav - which
doesn't really show FFmpeg in a very positive light.
(To make it even worse: this is a full-blown Libav API incompatibility,
even though this crap was added for Libav ABI-compatibility. It's
absurd.)
Quoting the FFmpeg header for the AVFrame.channels field:
/**
* number of audio channels, only used for audio.
* Code outside libavutil should access this field using:
* av_frame_get_channels(frame)
* - encoding: unused
* - decoding: Read by user.
*/
int channels;
It says "should" not must, and it doesn't even mention
av_frame_set_channels(). It's also in the section for public fields (not
below a marker that indicates private fields in a public struct, like
it's done e.g. in AVCodecContext).
But not using the accessor will cause silent failures on ABI changes.
The failure that happened due to this code didn't even make it apparent
what was wrong. So just use the idiotic accessor.
Also harmonize the FFmpeg-cursing in the code. (It's fully justified.)
Fixes#3295.
Note that mpv will still check the exact library version numbers, and
reject mismatches - to protect itself from such issues in the future.
Otherwise it behaves dumb. (Although you could argue it shouldn't try to
guess whether speed changes work, but instead simply disable DS if they
don't work.)
It used not to work - but now it apparently does. Not sure when that got
fixed in FFmpeg, but there's no longer a reason to keep this hack.
This also gets rid of the check for the read_seek2 field, which is not
part of the public API.
Both backends have code to close each FD of their wakeup_pipe array.
This array is default-initialized with 0, which means if the backends
exit before the wakeup pipe is created (e.g. when probing), they would
close FD 0.
Initialize the FDs with -1. Then we call close(-1) in these situations,
which is perfectly allowed and has no bad consequences.
This fits natively into the vo/backend and allows to simplify the
polling code.
One new change is the fact that surface_handle_enter flags VO_EVENT_WIN_STATE
and VO_EVENT_RESIZE instead of only VO_EVENT_WIN_STATE. Before this, the code
hackily relied on the timeout and the loop in the wait_frame function to track
and set the scaling factor. Instead, this triggers mpv to run a schedule_resize
and adjust the new VO output dimensions immediately. This is also more accurate
since surface_handle_enter() gets called when a surface is created, moved and
resized, which is exactly what the rest of the player might be interested in.
This uses GLSL mix() instead of going through an indirect texture
access. Easy to implement and might require less resources on some
devices, since the oversample code was already essentially just a
special case of this.
Could be made the new default (as per issue #2685), but that should be
done in a separate commit.
Until now, this has been either handled over vo.event_fd (which should
go away), or by putting event handling on a separate thread. The
backends which do the latter do it for a reason and won't need this, but
X11 and Wayland will, in order to get rid of event_fd.
There's no need to call wl_display_flush() since all the client-side
buffered data has already been flushed prior to polling the fd.
Instead only check for POLLIN and the usual ERR+HUP.
Don't just cause vo_opengl to update the ICC profile every time the
window is moved. Instead, explicitly check if the screen was changed.
Mostly untested.
Some client API users simply don't like such filenames. For their sake,
don't return them, but return a dummy filename instead. (Returning a
latin1-ized version would work too, but is slightly more work.)
Also remove the "\n" from the replacement dummy filename. This was
accidental.
mixer.c didn't really deserve to be separate anymore, as half of its
contents were unnecessary glue code after recent changes. It also
created a weird split between audio.c and af.c due to the fact that
mixer.c could insert audio filters. With the code being in audio.c
directly, together with other code that unserts filters during runtime,
it will be possible to cleanup this code a bit and make it work like the
video filter code.
As part of this change, make the balance code work like the volume code,
and add an option to back the current balance value. Also, since the
balance semantics are unexpected for most users (panning between the
audio channels, instead of just changing the relative volume), and there
are some other volumes, formally deprecate both the old property and the
new option.
Since mixer->ao is always NULL now (it was really just forgotten to be
removed), the uninit call never actually cleared the af field, leaving
a dangling pointer that could be accessed by volume control.
This makes the difference between passing VA_FRAME_PICTURE or
VA_BOTTOM_FIELD for progressive frames (that should be force-
deinterlaced) to VAProcPipelineParameterBuffer.flags. VA-VPP doesn't
really seem to care, and we can get rid of mp_refqueue_is_interlaced()
entirely. It could be argued it's better to pass field flags instead of
the progressive flag.
"Real" frame flag vs. what we pretend it to be. It always used the real
flag, and thus never deinterlaced unflagged frames, even if the
suboption was set to "no".
The hw_subfmt field roughly corresponds to the field
AVHWFramesContext.sw_format in ffmpeg. The ffmpeg one is of the type
AVPixelFormat (instead of the underlying hardware format), so it's a
good idea to switch to this too for preparation.
Now the hw_subfmt field is an mp_imgfmt instead of an opaque/API-
specific number. VDPAU and Direct3D11 already used mp_imgfmt, but
Videotoolbox and VAAPI had to be switched.
One somewhat user-visible change is that the verbose log will now always
show the hw_subfmt as image format, instead of as nonsensical number.
(In the end it would be good if we could switch to AVHWFramesContext
completely, but the upstream API is incomplete and doesn't cover
Direct3D11 and Videotoolbox.)
Old-style commands using _ as separator (e.g. show_progress) were still
used in some places, including documentation and configuration files.
This commit updates all such instances to the new style (show-progress)
so that commands are easier to find in the manual.
Since it turns out that knowing what exactly a file was tagged with can
be useful for debugging purposes, expose this as a property so I can
check it more easily.
This is mostly useful for sig-peak (since nom-peak is currently entirely
calculated by us), but I added both for consistency.
--deinterlace=auto is the default, and has the obscure semantics that
deinterlacing is disabled, unless the user has manually inserted a
deinterlacing filter.
While in software decoding this doesn't matter, and we will happily
insert 2 yadif filters (if the user has already added one), or not
remove the yadif filter (if deinterlacing is disabled, but the user has
added the filter manually), this is different with hardware deinterlacer
filters. These support VFCTRL_SET_DEINTERLACE for toggling deinterlacing
filtering at runtime. It exists mainly for legacy reasons, and possibly
because it makes switching deinterlacing modes more efficient. It might
also gives us an entry-point for VO deinterlacing, maybe. For whatever
reasons this mechanism exists, we still support and use it.
This commit fixes that video.c always used VFCTRL_SET_DEINTERLACE to
disable deinterlacing, even if --deinterlace=auto was set. Fix this by
checking the value of the option directly.
For some reason, the lack of version info was preventing mpv from
appearing in the Default Programs dialog. Re-add it, but don't set the
string version numbers from version.h, because that's what was causing
trouble when the version info was removed. Like the binary version
numbers, these are now hardcoded to 2.0.0.0, which probably doesn't
matter.
The new version info block is also slightly different to the old one. It
fills out all the binary VERSIONINFO fields and makes better use of
macros. It also removes the \000 line terminators from the string
version info, since as far as I can tell, this was just cargo-culting
for an old broken version of the Microsoft resource compiler, and
binutils' windres terminates the strings properly without them.
This should get mpv working on Windows 7 machines without hardware
accelerated graphics adapters. It already worked on Windows 8 and up
because those systems would silently fall back to WARP if there was no
graphics hardware installed.
The normal MPGL_CAP_SW flag is not set, so unlike other opengl backends,
this will choose a software adapter even if opengl:sw is not specified.
The reason for this is, unlike on Linux, where vo_xv and vo_x11 can be
used, mpv on Windows does not have any VO to fall back on when hardware
acceleration isn't available, so if software adapters are rejected, the
user won't see any video output when using the default settings. WARP
seems to perform quite well, so it should be used in this case.
The cache reader thread actually unlocks the mutex protecting the
underlying stream while reading from it. That's why other code goes out
of its way to run certain stream operations on the cache thread. Do the
same.
We could have this simpler by creating a mechanism that would "park" the
cache thread and make it wait for the lock (while we have it) in order
to gain exclusive access. This could be done in the future.
Eagerly execute seeks to the underlying stream in the cache seek
entrypoint itself. While asynchronous execution is a goal of the cache,
it doesn't matter too much for seeks. They always were executed within
the lock, so the reader was blocked anyway. It's not necessary to ensure
async. execution here either, because seeks are relatively rare, and the
demuxer can just stay blocked for a while.
Fixes: mpv http://samples.mplayerhq.hu/V-codecs/DIV5/ayaneshk-test.avi
This code was supposed to adjust existing conversion filters (to make
them output a different format). But the code was just broken,
apparently a refactoring accident. It accessed af instead of af->prev.
The bug tended to add new conversion filters, even if an existing one
could have been used. (Can be tested by inserting a dummy lavrresample
filter followed by a format filter which forces conversion.)
In addition, it's probably better to return the actual error code if
reinitializing the filter fails. It would then respect an AF_FALSE
return value, which means format negotiation failed, instead of a
generic error.
af_volume has a volumedb sub-option, which allows the user to set an
explicit volume. Until recently, the player read back this value and
used it as initial softvol volume. But now it just overwrites it.
Instead of overwriting it, multiply the different gain values. Above
all, this will do the right thing if only softvol is used, or if the
user only adds the af_volume filter manually.
Normally, you want downmixing to happen first thing in the filter chain.
This is reflected in codec downmixing, which feeds the filter chain
downmixed audio in the first place. Doing this has the advantage of
needing less data to process. But the main motivation is that if there
is a drc filter in the chain, you want to process it the downmixed
audio.
Add an idiotic heuristic to achieve this. It tries to detect whether the
audio was indeed automatically downmixed (or upmixed). To detect what
the output format is going to be, it builds the filter chain normally,
and then retries with the heuristic applied (and for extra paranoia,
retries without the heuristic again if it fails to successfully rebuild
the filter chain for unknown reasons). This is simple and will work in
almost all cases.
Doing it in a more complete way is rather hard, because filters are so
generic. For example, we know absolutely nothing about the behavior of
af_lavfi, which creates an opaque filter graph with libavfilter. We
don't know why a filter would e.g. change the channel layout on its
output. (Our heuristic bails out in this case.) We're also slave to the
lowest common denominator of how our format negotiation works, and how
libavfilter's works.
In theory, we could make this mechanism explicit by introducing a
special dummy filter. The filter chain would then try to convert between
input and output formats at the dummy filter, which would give the user
more control over how downmix happens. On the other hand, the user could
just insert explicit conversion filters instead, so this would probably
have questionable value.
The algorithm and functionality is the same, but the code becomes much
simpler and easier to follow.
The assumption that there is only 1 conversion filter (lavrresample)
helps with the simplification, but the main change is to use the same
code for format/channels/rate. Get rid of the different AF_CONTROL_SET_*
controls, and change the af->data parameters directly. (af->data is
badly named, but essentially is a placeholder for the output format.)
Also, instead of trying to use the af_reinit() loop to init inserted
conversion filters or filters with changed output formats, do it inline,
and move the common code to a filter_reinit() function. This gets rid of
the awful retry variable.
In general, this should not change any runtime behavior.
This happens to be better for the af_volume filter (for softvol), and
saves some code too. It's "better" because you want to affect the
final filtered audio, such as after a manually inserted drc filter.
Drop the code for switching the volume options and properties between
af_volume and AO volume controls. interface-changes.rst mentions the
changes in detail.
Do this because this was exceedingly complex and had other problems as
well. It was also very hard to test. It's just not worth the trouble.
Some leftovers like AOCONTROL_HAS_PER_APP_VOLUME will be removed at a
later point.
Fixes#3322.
I made this call up because I sort of thought this makes senssssse. I'm
now convinced that it does not, and even is actively harmful. I'm still
quite in the dark how the DirectD 11 video API is supposed to work with
synchronization, but at least for normal graphics this call would not
make much sense.
When selecting a device that simply doesn't exist with --audio-device,
AudioUnit will still initialize and start playback without complaining.
But it will never call the audio render callback, which leads to audio
playback simply not progressing.
I couldn't find a way to get AudioUnit to report an error at all, so
here's a crappy hack that takes care of this in most cases. We assume
that all devices have a kAudioDevicePropertyDeviceIsAlive property.
Invalid devices will error when querying the property (with 'obj!' as
status code).
This is not the correct fix, because we try to double-guess AudioUnit's
behavior by accessing a lower label API. Suggestions welcome.
Although it appears to be accepted by the function, MSGL_STATUS messages
are never passed to the client API. Consequently "status" has the same
meaning as "v" and is useless.
Commit 771a8bf5 added code to avoid unnecessary vf_reconfig() calls for
unrelated reasons, but forget to consider that it has to be called at
least once if the input format changes. As a consequence it got "stuck"
due to not being able to decode more frames.
vo_frame can have more than 1 frame - the extra frames are future
references, which are sometimes useful for filtering (vo_opengl
interpolation). There's no harm in reducing the number of frames sent to
the VO requested amount of future frames, so do that.
Doesn't actually reduce the number of concurrently in use frames in
practice.
If the status line is wider than the reported terminal size, then cut it
off instead of causing the terminal to scroll down for the next line.
This is done in the most primitive way possible, assuming ASCII.
This was actually done in the past as far as I'm aware; do it again.
(Probably differently.)
Some filters support VFCTRL_SET_DEINTERLACE. This affects most hardware
deinterlace filters. They can be inserted by the user manually, or auto-
inserted by vf.c itself as conversion filter (vf_d3d11vpp). In these
cases, we shouldn't insert or remove filters outselves, and instead
VFCTRL_SET_DEINTERLACE should be invoked to switch the mode.
This wasn't done correctly in the recently refactored code and could
have broken with --deinterlace. (The refactor only considered switching
via property in this case.) Fix it by making it a proper part of the
filter_reconfig() function, and making set_deinterlacing() (which is
called by the property handler) merely call filter_reconfig() in all
cases to do the real work.
We can even avoid rebuilding the filter chain - though only if no other
auto-filters are inserted. It probably also provides a slightly cleaner
way to implement functionality in the VO while still inserting video
filter fallbacks correctly if required.
The test scenario at hand was hardware decoding a file with d3d11 and
with deinterlacing enabled. The file switches to a non-hardware
dedocdeable format mid-stream. This failed, because it tried to call
vf_reconfig() with the old filters inserted, with was fatal due to
vf_d3d11vpp accepting only hardware input formats.
Fix this by always strictly removing all auto-inserted filters
(including the deinterlacing one), and reconfiguring only after that.
Note that this change is good for other situations too, because we
generally don't want to use a hardware deinterlacer for software
decoding by default. They're not necessarily optimal, and VAAPI VPP even
has incomprehensible deinterlacer bugs specifically with software frames
not coming from a hardware decoder.
Instead of using the "vf" command code (which changes filters at runtime
on user input), use the general filter-insertion code. The latter was
added later, and is more suitable for automatically inserted filters.
The old code failed in particular when using watch-later saving, which
stored the filter list in the resume config file. If a user changed the
hardware decoding mode via command line, the stored filter chain was out
of date and could cause failure due to not working with hardware or
software decoding mode. Storing the deinterlace filter in the filter
list was unavoidable, because it was part of the user state. (The new
code only edits the actually instantiated filters.)
These mostly happen in situations where the correct behavior is
relatively new and not found in the wild (therefore not worth
implementing) and/or extremely complicated (and thus not worth worrying
about the potential edge cases and UI changes).
Still, it's best to document these where they happen to guide the poor
souls maintaining these files in the future.
This uses eglPostSubBufferNV to trigger ANGLE to check the window size
and update the size of the swapchain to match, which is recommended
here: https://groups.google.com/d/msg/angleproject/RvyVkjRCQGU/gfKfT64IAgAJ
With the D3D11 backend, using eglPostSubBufferNV with a 0-sized update
region will even skip the Present() call, meaning it won't block for a
vsync period. Hopefully ANGLE will have a less hacky way of doing this
in future. See the relevant ANGLE issue: http://anglebug.com/1438Fixes#3301
This can for example happen with vo_opengl_cb, if it is used with a GL
implementation that does not supports FBOs. (mpv itself should never
attempt to use FBOs if they're not available.)
Without this check it would trigger an assert() in our dummy
glBindFramebuffer wrapper.
Suspected cause of #3308, although it's still unlikely.
For example it should be set to IMGFMT_D3D11NV12 if it isn't already.
Otherwise, an assertion in vf.c could trigger.
This probably couldn't be provoked yet.
This moves some of the bulky user-shader specific logic into the file
dedicated to it. Rather than expose video.c state, variable lookup is
now done via a simulated closure.
This greatly improves the result when decoding typical (ST.2084) HDR
content, since the job of tone mapping gets significantly easier when
you're only mapping from 1000 to 250, rather than 10000 to 250.
The difference is so drastic that we can now even reasonably use
`hdr-tone-mapping=linear` and get a very perceptually uniform result
that is only slightly darker than normal. (To compensate for the extra
dynamic range)
Due to weird implementation details, this only seems to be present on
keyframes (or something like that), so we have to cache the last seen
value for the frames in between.
Also, in some files the metadata is just completely broken /
nonsensical, so I decided to apply a simple heuristic to detect
completely broken metadata.
This involves multiple changes:
1. Brightness metadata is split into nominal peak and signal peak.
For a quick and dirty explanation: nominal peak is the brightest value
that your color space can represent (i.e. the brightness of an encoded
1.0), and signal peak is the brightest value that actually occurs in
the video (i.e. the brightest thing that's displayed).
2. vo_opengl uses a new decision logic to figure out the right nom_peak
and sig_peak for all situations. It also does a better job of picking
the right target gamut/colorspace to use for the OSD. (Which still is
and still should be treated as sRGB). This change in logic also
fixes#3293 en passant.
3. Since it was growing rapidly, the logic for auto-guessing / inferring
the right colorimetry configuration (in pass_colormanage) was split from
the logic for actually performing the adaptation (now pass_color_map).
Right now, the new logic doesn't do a whole lot since HDR metadata is
still ignored (but not for long).
This has two reasons:
1. I tend to add new fields to this metadata, and every time I've done
so I've consistently forgotten to update all of the dozens of places in
which this colorimetry metadata might end up getting used. While most
usages don't really care about most of the metadata, sometimes the
intend was simply to “copy” the colorimetry metadata from one struct to
another. With this being inside a substruct, those lines of code can now
simply read a.color = b.color without having to care about added or
removed fields.
2. It makes the type definitions nicer for upcoming refactors.
In going through all of the usages, I also expanded a few where I felt
that omitting the “young” fields was a bug.
Commit 883d3114 seems to have (accidentally?) dropped the FBOTEX_FUZZY
from the output_fbo resize, which means that current master will keep
resizing and resizing the FBO as you change the window size, introducing
severe memory leaking after a while. (Not sure why that would cause
memory leaks, but I blame nvidia)
Either way, it's bad for performance too, so it's worth fixing.
vo_vaapi is the only thing which can't scale RGBA on the GPU. (Other
cases of RGBA scaling are handled in draw_bmp.c for some reason.)
Move this code and get rid of the osd_conv_cache thing.
Functionally, nothing changes.
This affects VOs (or other code which render OSD) which does not support
the LIBASS format, but only RGBA. Instead of having a converter stage in
osd.c, make mp_ass_packer_pack() output directly in RGBA.
In general, this is work towards refcounted subtitle images.
Although we could keep the "converter" design, doing it this way seems
simpler, at least considering the current situation with only 2 OSD
formats. It also prevents copying & packing the data twice, which will
lead to better performance. (Although I guess this case is not important
at all.)
It also fixes --force-rgba-osd-rendering when used with vo_opengl,
vo_vdpau, and vo_direct3d.
The intention is to let mp_ass_packer_pack() produce different output
for the RGBA and LIBASS formats. VOs (or whatever generates the OSD)
currently do not signal a preferred format, and this mechanism just
exists to switch between RGBA and LIBASS formats correctly, preferring
LIBASS if the VO supports it.
This is how PBOs are normally supposed to be used.
Unfortunately I can't see an any absolute improvement on nVidia binary
drivers and playing 4K material. Compared to the "old" PBO path with 1
buffer, the measured GL time decreases significantly, though.
GL generally does not support flipping the image on upload, meaning
negative strides are not supported. vo_opengl handles this by flipping
rendering if the stride is inverted, and gl_pbo_upload() "ignores"
negative strides by uploading without flipping the image.
If individual planes had strides with different signs, this broke. The
flipping affected the entire image, and only the sign of the first plane
was respected.
This is just a crazy corner case that will never happen, but it turns
out this is quite simple to support, and actually improves the code
somewhat.
This introduces a gl_pbo_upload_tex() function, which works almost like
our gl_upload_tex() glTexSubImage2D() wrapper, except it takes a struct
which caches the PBO handles. It also takes the full texture size (to
make allocating an ideal buffer size easier), and a parameter to disable
PBOs (so that the caller doesn't have to duplicate the gl_upload_tex()
call if PBOs are disabled or unavailable).
This also removes warnings and fallbacks on PBO failure. We just
silently try using PBOs on every frame, and if that fails at some point,
revert to normal texture uploads. Probably doesn't matter.
This makes the geometry of the sizing borders more like the ones in
Windows 10. It also fixes an off-by-one error that made the right and
bottom borders thinner than the left and top borders, which made it
difficult to resize the window when using the Windows 7 classic theme
(because it has pretty thin sizing borders to begin with.)
We don't support this anymore.
This tries to exit in a controlled way after command line options are
applied in order to honor logging options and, in case of libmpv, not to
kill the host. Not sure if it would be better to just vomit text to
stderr and call abort().
Working towards refcounted sub images, and also for removing bitmap
packers from VOs.
I'm not sure why we even have this overlay-add command. It was sort of
"needed" before opengl-cb was introduced, and before Lua scripts could
put ASS drawings on OSD without conflicting with the OSC. But now trying
to use it doesn't make too much sense anymore.
Still keep it because we're trying to be nice, but throw performance out
of the window. Now image data is copied 2 more times before displaying
it. This also makes using the command a bit simpler.
The libavcodec wmapro decoder will skip some bytes at the start of the
first packet and return each time. It will not return any audio data in
this state.
Our own code as well as libavcodec's new API handling
(avcodec_send_packet() etc.) discard the PTS on the first return, which
means the PTS is never known for the first packet. This results in a
"Failed audio resync." message.
Fixy it by remember the PTS in next_pts. This field is used only if the
decoder outputs no PTS, and is updated after each frame - and thus
should be safe to set.
(Possibly this should be fixed in libavcodec new API handling by not
setting the PTS to NOPTS as long as no real data has been output. It
could even interpolate the PTS if the timebase is known.)
Fixes the failure message seen in #3297.
Change all producer of libass images to packing the bitmaps into a
single larger bitmap directly when they're output. This is supposed to
help working towards refcounted sub bitmaps.
This will reduce performance for VOs like vo_xv, but not for vo_opengl.
vo_opengl simply will pick up the pre-packed sub bitmaps, and skip
packing them again. vo_xv will copy and pack the sub bitmaps
unnecessarily - but if we want sub bitmap refcounting, they'd have to be
copied anyway.
The packing code cannot be removed yet from vo_opengl, because there are
certain corner cases that still produce unpackad other sub bitmaps.
Actual refcounting will also require more work.
We have a warning mechanism for removed and for replaced options, but
none yet for options which have been simply deprecated.
For the following commit.
(Fun fact: just adding the m_option field increases binary size by
14KB.)
No method of taking a screenshot was implemented at all. vo_opengl
lacked window screenshotting, because ANGLE doesn't allow reading the
frontbuffer. There was no way to read back from a D3D11 texture either.
Implement reading image data from D3D11 textures. This is a low-quality
effort to get basic screenshots done. Eventually there will be a better
implementation: once we use AVHWFramesContext natively, the readback
implementation will be in libavcodec, and will be able to cache the
staging texture correctly. Hopefully. (For now it doesn't even have a
AVHWFramesContext for D3D11 yet. But the abstraction is more appropriate
for this purpose.)
OK, this was dumb. The file didn't have much to do with ANGLE, and the
functionality can simply be moved to d3d.c. That file contains helpers
for decoding, but can always be present (on Windows) since it doesn't
access any D3D specific libavcodec APIs. Thus it doesn't need to be
conditionally built like the actual hwaccel wrappers.
Instead of hard-coding a big list, move some of the functionality
to csputils. Affects both the auto-guess blacklist and the peak
estimation.
Also update the comments.
Too many "exceptions" these days, it's easier to just hard-code a
whitelist instead of a blacklist. And besides, it only really makes
sense to avoid adaptation for BT.601 specifically, since that's the one
we auto-guess based on the resolution.
I'm not even sure why we ever consulted *_src to begin with, since that
just describes the current image format - and not the original metadata.
(And in fact, we specifically had logic to work around the impliciations
this had on linear scaling)
image_params is *the* authoritative source on the intended (i.e.
reference) image metadata, whereas *_src may be changed by previous
passes already. So only consult image_params for picking auto-generated
values.
Also, add some more missing "wide gamut" and "non-gamma" curves to the
autoconfig blacklist. (Maybe it would make sense to move this list to
csputils in the future? Or perhaps even auto-detect it based on the
associated primaries)
User request and not that hard. Closes#3157.
Note that FFmpeg doesn't support this and there's no signalling in HEVC
etc., so the only way users can access it is by using vf_format
manually.
Mind: This encoding uses full range values, not TV range.
This is actually not entirely trivial since it involves negative Yxy
coordinates, so the CMM has to be capable of full floating point
operation. Fortunately, LittleCMS is, so we can just blindly implement
it.
Most devices seems to require special signalling (e.g. via HDMI
metadata) to actually decode HDR signals and treat them as such, so it's
probably worth warning the potential user about the fact that mpv pretty
definitely does *not* set any of this metadata signalling.
This HDR function is unique in that it's still display-referred, it just
allows for values above the reference peak (super-highlights). The
official standard doesn't actually document this very well, but the
nominal peak turns out to be exactly 12.0 - so we normalize to this
value internally in mpv. (This lets us preserve the property that the
textures are encoded in the range [0,1], preventing clipping and making
the best use of an integer texture's range)
This was grouped together with SMPTE ST2084 when checking libavutil
compatibility since they were added in the same release window, in a
similar timeframe.
Some bugs in this code are exposed by e.g. playing lossless audio files
with --ad-lavc-threads=16. (libavcodec doesn't really support threaded
audio decoding, except for lossless files.) In these cases, a major
amount of audio can be buffered, which makes incorrect handling of this
buffering obvious.
For one, draining the decoder can take a while, so if there's a new
segment, we shouldn't read audio.
The segment end check was completely wrong, and used the start value.
The main framebuffer is not the default framebuffer for the dxinterop
backend. Bind the main framebuffer and use the appropriate attachment
when reading the window content.
Fix#3284
The size check introduced in commit d941a57b did not consider that Xv
can round up the image size to the next chroma boundary. Doing that
makes sense, so it can't certainly be considered server misbehavior.
Do 2 things against this: allow if the server returns a larger image (we
just crop it then), and also allocate a properly aligned image in the
first place.
Instead of hardcoding what the libavcodec ac3 encoder expects, configure
it based on the AVCodec fields.
Unfortunately, it doesn't export the list of sample rates, so that is
done manually. This commit actually fixes the rate always to 48Khz. I
don't even know whether the other rates worked. (Possibly did, but
they'd still change the spdif parameters, and would work differently
from ad_spdif.c.)
Workaround for an awful corner-case. The new decode API "locks" the
decoder into the EOF state once a drain packet has been sent. The
problem starts with a file containing a 0-sized packet, which is
interpreted as drain packet.
This should probably be changed in libavcodec (not treating 0-sized
packets as drain packets with the new API) or in libavformat (discard
0-sized packets as invalid), but efforts to do so have been fruitless.
Note that vd_lavc.c already does something similar, but originally for
other reasons.
Fixes#3106.
See previous commit. (The mixing case was never supported, so this has
equivalent functionality.)
This also implicitly fixes a bug: the old code allocated image data for
the cropped surface size only, which means the
video_surface_get_bits_y_cb_cr call could write beyond the allocated
image memory.
For now, this affects taking screenshots only.
If the video mixer is actually needed, we want to go through the video
mixer for screenshots too - which is why this function simply always
used the video mixer, and then copied the surface to RAM as RGBA.
Add reading the surface as nv12 if possible. If the format is correct,
and no special video processing (like deinterlacing) is used, then it's
read directly without going through the video mixer.
There's no particular reason for doing this, other than avoiding the
video mixer (like vo_opengl can do it now). Also, the next commit will
make use of this in vf_vdpaurb.c.
The GL_ARB_timer_query extension and thus the GL_TIME_ELAPSED constant
don't exist for GLES.
For ES the EXT_disjoint_timer_query is used so take the constant from
that else provide the constant manually.
See pr #3216 which introduced this error.
Until now, we've always converted vdpau video surfaces to RGB, and then
mapped the resulting RGB texture. Change this so that the surface is
mapped as NV12 plane textures.
The reason this wasn't done until now is because vdpau surfaces are
mapped in an "interlaced" way as separate fields, even for progressive
video. This requires messy reinterleraving. It turns out that even
though it's an extra processing step, the result can be faster than
going through the video mixer for RGB conversion.
Other than some potential speed-gain, doing this has multiple other
advantages. We can apply our own color conversion, which is important in
more complex cases. We can correctly apply debanding and potentially
other processing that requires chroma-specific or in-YUV handling.
If deinterlacing is enabled, this switches back to the old RGB
conversion method. Until we have at least a primitive deinterlacer in
vo_opengl, this will stay this way. The d3d11 and vaapi code paths are
similar. (Of course these don't require any crazy field reinterleaving.)
Otherwise stale references will survive forever. Could leak hardware
video surfaces.
In particular, the mpv vdpau code crashed with an assertion when exiting
after toggling deinterlacing, because not all references were released.
1. this basically reverts commit de4c74e5a4.
even with CVDisplayLinkCreateWithActiveCGDisplays and
CVDisplayLinkSetCurrentCGDisplayFromOpenGLContext we still have to
explicitly set the current display ID, otherwise it will just always
choose the display with the lowest refresh rate. another weird thing is,
we still have to set the display ID another time with
CVDisplayLinkSetCurrentCGDisplay after the link was started. otherwise
the display period is 0 and the fallback will be used.
if we ever use the callback method for something useful it's probably
better to use CVDisplayLinkCreateWithActiveCGDisplays since we will need
to keep the display link around instead of releasing it at the end.
in that case we have to call CVDisplayLinkSetCurrentCGDisplay two times,
once before and once after LinkStart.
2. add windowDidChangeScreen delegate to update the display refresh rate
when mpv is moved to a different screen.
We have two problems here.
1. CVDisplayLinkGetActualOutputVideoRefreshPeriod, like the name suggests,
returns a frame period and not a refresh rate. using this as screen_fps
just leads to a slideshow. why didn't this break video playback on OS X
completely? the answer to this leads us to the second problem.
2. it seems that CVDisplayLinkGetActualOutputVideoRefreshPeriod always
returns 0 if used without CVDisplayLinkSetOutputCallback and hence always
fell back to CVDisplayLinkGetNominalOutputVideoRefreshPeriod. adding a
callback to CVDisplayLink solves this problem. the callback function at
this moment doesn't do anything but could possibly used in the future.
This can also remove all the stuff for lazily attaching the texture. It
doesn't matter if the dxinterop backend changes the bound framebuffer
during a VOCTRL, since the renderer does not rely on the GL state being
preserved.
Since there are not many sub-rectangles, this doesn't cost too much. On
the other hand, it avoids frequent warnings with vo_xv.
Also, the second copy in mp_blur_rgba_sub_bitmap() can be dropped.
The previous few commits changed sd_lavc.c's output to packed RGB sub-
images. In particular, this means all sub-bitmaps are part of a larger,
single bitmap. Change the vo_opengl OSD code such that it can make use
of this, and upload the pre-packed image, instead of packing and copying
them again.
This complicates the upload code a bit (4 code paths due to messy PBO
handling). The plan is to make sub-bitmaps always packed, but some more
work is required to reach this point. The plan is to pack libass images
as well. Since this implies a copy, this will make it easy to refcount
the result.
(This is all targeted towards vo_opengl. Other VOs, vo_xv, vo_x11, and
vo_wayland in particular, will become less efficient. Although at least
vo_vdpau and vo_direct3d could be switched to the new method as well.)
The sub-bitmaps get extended by --sub-gauss, so we have to compute the
bounding box on the original subs. Not sure if this is really
eqauivalent to what the code did before, and I don't have the sample
anymore. (But this approach sure is a _shitty_ hack.)
Implement it directly in sd_lavc.c as well. Blurring requires extending
the size of the sub-images by the blur radius. Since we now want
sub_bitmaps to be packed into a single image, and we don't want to
repack for blurring, we add some extra padding to each sub-bitmap in the
initial packing, and then extend their size later. This relies on the
previous bitmap_packer commit, which always adds the padding in all
cases.
Since blurring is now done on parts of a large bitmap, the data pointers
can become unaligned, depending on their position. To avoid shitty
libswscale printing a dumb warning, allocate an extra image, so that the
blurring pass is done on two newly allocated images. (I don't find this
feature important enough to waste more time on it.)
The previous refactor accidentally broke this feature due to a logic bug
in osd.c. It didn't matter before it happened to break, and doesn't
matter now since the code paths are different.
Until now, subtitle renderers could export SUBBITMAP_INDEXED, which is a
8 bit per pixel with palette format. sd_lavc.c was the only renderer
doing this, and the result was converted to RGBA in every use-case
(except maybe when the subtitles were hidden.)
Change it so that sd_lavc.c converts to RGBA on its own. This simplifies
everything a bit, and the palette handling can be removed from the
common code.
This is also preparation for making subtitle images refcounted. The
"caching" in img_convert.c is a PITA in this respect, and needs to be
redone. So getting rid of some img_convert.c code is a positive side-
effect. Also related to refcounted subtitles is packing them into a
single mp_image. Fewer objects to refcount is easier, and for the libass
format the same will be done. The plan is to remove manual packing from
the VOs which need single images entirely.
Apply the padding internally to each input bitmap, instead of requiring
this for the semi-public API.
Right now, everything still uses packer_pack_from_subbitmaps() to fill
the input bitmap sizes, but that's going to change with the following
commit. Since bitmap_packer.in is mutated during packing anyway, it's
more convenient to add the padding automatically.
Also, guarantee that every sub-bitmap has a padding border around it.
Don't let the padding overlap. Add padding even on the containing
borders.
This is simpler, doesn't cost much in memory usage, and is convenient
for one of the following commits.
I do not understand why github requires adding this crap to source code
repositories themselves, instead of making them part of the repository
configuration. Remove CONTRIBUTING.md to compensate for github crap
accumulating.
This is the ES equivalent to ARB_timer_query. It enables the performance
timers on ANGLE. All the added functions should be identical in
semantics to their desktop GL equivalents.
The OpenGL 3.0+ and ES specs are quite clear on what values are
accepted for the attachment object name parameter. And there's no
overlap for the default framebuffer. Sigh.
Probably fixes Mesa raising an error in this case and might fix#3251.
Regression by the previous vo_opengl change.
Until now, we've used system-specific API (GLX, EGL, etc.) to retrieve
the depth of the default framebuffer. (We equal this to display depth
and use the determined depth for dithering.)
We can actually retrieve this value through standard GL API, and it
works everywhere (except GLES 2 of course). This simplifies everything a
great deal.
egl_helpers.c is empty now. But I expect that some EGL boilerplate will
be moved to it, so don't remove it yet.
This was somehow done under the assumption that ANGLE would somehow
always use RG in ES2 mode. But there's no basis for this. Even if ANGLE
supports NV12 textures with drivers that do not allow for texture_rg,
this cas eis too obscure to worry about. So do the robust and correct
thing instead, and disable this code if texture_rg is not available.
Commit 74e3d11 resulted in the background overlay not getting destroyed
when mpv quits. Add back a piece of code that was removed in that commit
to restore correct functionality.
Fixes issue #3100
This helps with shitty APIs and even shittier drivers (I'm looking at
you, ALSA). Sometimes they won't send proper wakeups. This can be fine
during playback, when for example playing video, because mpv still will
wakeup the AO outside of its own wakeup mechanisms when sending new data
to it. But when draining, it entirely relies on the driver's wakeup
mechanism. So when the driver wakeup mechanism didn't work, it could
hard freeze while waiting for the audio thread to play the rest of the
data.
Avoid this by waiting for an upper bound. We set this upper bound at the
total mpv audio buffer size plus 1 second. We don't use the get_delay
value, because the audio API could return crap for it, and we're being
paranoid here. I couldn't confirm whether this works correctly, because
my driver issue fixed itself.
(In the case that happened to me, the driver somehow stopped getting
interrupts. aplay froze instead of playing audio, and playing audio-only
files resulted in a chop party. Video worked, for reasons mentioned
above, but drainign froze hard. The driver problem was solved when
closing all audio output streams in the system. Might have been a dmix
related problem too.)
When we're draining, don't wakeup the core on every buffer fill, since
unlike during normal playback, we won't actually get more data. The
wakeup here conceptually works like wakeups with condition variables, so
redundant wakeups do not hurt, so this is just a minor change and
nothing of consequence.
(Final EOF also requires waking up the core, but there is separate code
to send this notification.)
Also dump the p->still_playing field in trace logging.
Of course we can't just skip updating the OSD if the playloop was woken
up for the purpose of removing OSD after an OSD timer expired.
Fixes e.g. OSD bars sometimes sticking along when seeking while paused.
Normally, OSD is updated every time the playloop is run. This has to be
done, because the OSD may implicitly reference various properties,
without knowing whether they really need to be updated or not. (There's
a property update mechanism, but it's mostly unavailable, because OSD is
special-cased and can not use the client API mechanism properly.)
Normally, these updates are no problem, because the OSD is only actually
printed when the OSD text actually changes.
But commit d23ffd24 added a rate-limiting mechanism, which tries to
limit OSD updates at most every 50ms (or the next video frame). Since it
can't know in advance whether the OSD is going to change or not, this
simply waked up the player every 50ms.
Change this so that the player is updated only as part of general
updates determined through mp_notify(). (This function also notifies the
client API of changed properties.) The desired result is that the player
will not wake up at all in normal idle mode, but still update properties
that can change when paused, such as the cache.
This is mostly a cosmetic change (in the sense of making runtime
behavior just slightly better). It has the slightly more negative
consequence that properties which update implicitly (such as "clock")
will not update periodically anymore.
This is a common idiom used in MSDN docs and Raymond Chen's example
programs to get a HINSTANCE for the current module, regardless of
whether it's an .exe or a .dll. Using GetModuleHandle(NULL) for this is
technically incorrect, since it always gets a handle to the .exe, even
when the executing code (in libmpv) is running in a .dll. In this case,
using the wrong HINSTANCE could cause namespace issues with window
classes, since CreateWindowEx uses the HINSTANCE to search for the
matching window class name.
See:
https://blogs.msdn.microsoft.com/oldnewthing/20050418-59/?p=35873https://blogs.msdn.microsoft.com/oldnewthing/20041025-00/?p=37483
There were two mistakes:
- SDL's RGB565 is always equivalent to ffmpeg's RGB565 (both are packed 16bit
native-endian integers in RGB=565 form) - this was wrongly reversed on
big endian platforms.
- SDL's RGB888 doesn't actually mean RGB24, but XRGB8888 (i.e. 32bit
packed integer, top 8 bits unused).
- Use RGB0 not RGBA when there is no alpha.
This is mainly for the nnedi3 user shader. With all whose NN weights
hardcoded into the shader source code, the shader file could be as
large as 300 kB.
This reverts commit b51957fab5.
Breaks big time. It appears to ignore explicitly configured paths within
the libav* .pc files, which for example breaks mpv-build.
Although D3D11 video decoding is unuspported on Windows 7, the
associated APIs almost work. Where they fail is texture creation, where
we try to create D3D11_BIND_DECODER surfaces. So specifically try to
detect this situation.
One issue is that once the hwdec interop is created, the damage is done,
and it can't use another backend (because currently only 1 hwdec backend
is supported). So that's where we prevent attempts to use it.
It still can fail when trying to use d3d11va-copy (since that doesn't
require an interop backend), but at that point we don't care anymore -
dxva2(-copy) is tried before that anyway.
Distros and users alike should be made aware of the fact that old
FFmpeg versions are bad. When users come to us with FFmpeg-related
trouble, the answer is “update FFmpeg” more often than not
(and no further support will be provided until they have done so),
so instead we just nag them about it here.
Instead of having 9 different properties, requiring 18 different
VOCTRLs to read them all, they are now exposed as a single property.
This is not only cleaner (since they're all together) but also allows
querying all 9 of them with only a single VOCTRL (by using
mp.get_property_native).
(The extra factor of 2 was due to an extra query being needed to get the
type, which is now also unnecessary)
This makes it much easier to access performance metrics from within a
lua script, and also makes it easier to just show a readable, formatted
version via show-text.
User hooks can now use an extra WHEN expression to specify when the
shader should be run. For example, this can be used to only run a chroma
scaling shader `WHEN CHROMA.w LUMA.w <`.
There's a slight semantics change to user shaders: When trying to bind a
texture that does not exist, a shader will now be silently skipped
(similar to when the condition is false) instead of generating an error.
This allows shader stages to depend on an optional earlier stage without
having to copy/paste the same condition everywhere.
(In other words: there's an implicit condition on all of the bound
textures existing)
When using --hwdec=auto, systems that don't provide
D3D11_CREATE_DEVICE_VIDEO_SUPPORT, which probably includes all Windows
Vista and 7 systems, will print an error message. Reduce the log level
to verbose when probing and skip the error message entirely if d3d11.dll
is not present.
This commit is in a similar spirit to 991af7d.
This comes up often, see e.g. #3220. The issue is that if the stream
input is not seekable, the demuxer is marked as not seekable. But if the
stream cache is enabled, the file still _might_ be seekable to a degree.
We recently disabled seeking in this mode because it can cause very
weird issues, mostly because if stream-layer seeking fails, the demuxers
will arbitrarily misbehave. On the other hand, it can work if the seek
is within the cached range, which is why the user can still enable it
with --force-seeking. There is a weird trade-off between allowing this
and not crapping up too easily, so just informing the user about the
possibility seems best.
Since the libavformat API is crap, we have to apply tons of heuristics
to check whether seeking will work. (No, checking it at seek time isn't
going to work either, because if a seek fails, the demuxer will be in an
undefined state. Because the libavformat API is crap.)
For clang, it's enough to just put (void) around usages we are
intentionally ignoring the result of.
Since GCC does not seem to want to respect this decision, we are forced
to disable the warning globally.
The default behavior of vo_opengl has pretty much always been 'show the
source colors as-is, without caring to adapt it to the target device'.
This decision is mostly based on the fact that if we do anything else,
lots of people will complain.
With the rise of content like BT.2020, however, it turns out more people
complain about this content being very desaturated than people complain
about this content not matching VLC - so let's just map ultra-wide gamut
content back down to standard gamut by default.
Instead of measuring the actual upload time, this instead measures the
time needed to render + map the texture via vdpau. These numbers are
still useful, since they're part of the critical path.
This is plumbed through a new VOCTRL, VOCTRL_PERFORMANCE_DATA, and
exposed as properties render-time-last, render-time-avg etc.
All of these numbers are in microseconds, which gives a good precision
range when just outputting them via show-text. (Lua scripts can
obviously still do their own formatting etc.)
Signed-off-by: wm4 <wm4@nowhere>
To avoid blocking the CPU, we use 8 time objects and rotate through
them, only blocking until the last possible moment (before we need
access to them on the next iteration through the ring buffer). I tested
it out on my machine and 4 query objects were enough to guarantee
block-free querying, but the extra margin shouldn't hurt.
Frame render times are just output at the end of each frame, via MP_DBG.
This might be improved in the future. (In particular, I want to expose
these numbers as properties so that users get some more visible feedback
about render times)
Currently, we measure pass_render_frame and pass_draw_to_screen
separately because the former might be called multiple times due to
interpolation. Doing it this way gives more faithful numbers. Same goes
for frame upload times.
When ANGLE is using D3D11 and not running in DirectComposition mode,
DXGI will hook the video window's message loop and override Alt+Enter to
trigger a transition to exclusive fullscreen mode (which doesn't even
work with mpv's renderer for some reason.) This behaviour can be
disabled by getting a pointer to the IDXGIFactory associated with the
D3D11 device and calling MakeWindowAssociation with the appropriate
flags.
Since the main thread is shared by other things in the player, using STA (single
threaded aparement) may have caused problems. Instead initialize in MTA
(multithreaded apartment).
Instead of implicitly resetting the options to defaults and then
applying the options, they're always applied on top of the current
options (in the same way adding new options to the CLI command line
will).
This does not apply to vo_opengl_cb, because that has an even worse mess
which I refuse to deal with.
Enable m_sub_options_copy() to copy nested sub-options, and also enable
it to create an option struct from defaults. We can get rid of most of
the crap in assign_options() now.
Calling handle_scaler_opt() to get a static allocation for scaler name
is still needed. It's moved to reinit_scaler(), which seems to be a
better place for it. Without it, dangling pointers could be created when
options are changed. (And in fact, this fixes possible dangling pointers
for window.name.) In theory we could create a dynamic copy, but that
seemed even more messy.
Chance of regressions.
Commit 026b75e7 actually enabled changing icc options at runtime (via
vo_cmdline), but it didn't quite work. In particular, changing the icc-
profile option just kept the old profile, because it was cached
accordingly.
As part of this, change gl_lcms.opts from a struct to a pointer to a
struct. We properly copy it, instead of allowing possibly dangling
strings, like it was done in a working but unclean way before.
Also, reinit the whole rendering chain when the auto icc profile
changes, just like it's done when icc options are changed.
Passing the bstr thing as pointer makes no sense. Everywhere else bstr
structs are passed by value because they're so small. Only when it's
supposed to receive a return value they're not.
Originally, video.c did not access any CMS things (other than lut3d
being set on it), but this has changed. In practice, almost all accesses
to it have moved to video.c. vo_opengl only created it, and set the auto
icc profile path.
Complete the move.
Some things wrt. option handling are a bit fishy. (But when is this not
the case.)
icc-profile-auto was not tested, but the distributed human CI will take
care of it.
It gets printed on every alt+tab or desktop switch under mutter and
weston, and offers no useful information since it's handled by
destroying the previous entry.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit will cause the wayland backend and vo to correctly report
the display frame rate. This didn't work as VOCTRL_GET_DISPLAY_FPS was
received way too early, before the window was created (and thus
current_output set).
The VO will now signal VO_EVENT_WIN_STATE after window initialization
and upon a resize.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This algorithm works really well. Setting it is a much better
"out-of-the-box" experience than just clipping, which will always look
ugly.
In other words, with this default, users of mpv will just be able to
play HDR content without even realizing it's HDR (pretty much).
Instead of doing HDR tone mapping on an ad-hoc basis inside
pass_colormanage, the reference peak of an image is now part of the
image params (alongside colorspace, gamma, etc.) and tone mapping is
done whenever peak_src != peak_dst.
To get sensible behavior when mixing HDR and SDR content and displays,
target-brightness is a generic filler for "the assumed brightness of SDR
content".
This gets rid of the weird display_scaled hack, sets the framework
for multiple HDR functions with difference reference peaks, and allows
us to (in a future commit) autodetect the right source peak from
the HDR metadata.
(Apart from metadata, the source peak can also be controlled via
vf_format. For HDR content this adjusts the overall image brightness,
for SDR content it's like simulating a different exposure)
The wayland protocol exposes scaling done by the compositor to
compensate for small window sizes on small high DPI displays. If the
program ignores the scaling done, what'll happen is the compositor is
going to ask the program to be scaled down by N times the window size and
then it'll upscale the program's surface by N times. The scaling
algorithm seems to be bilinear so the scaling is quite obvious.
This commit sets up callbacks to listen for the scaling factor of each
output and, on rescale events, notifies the compositor that the
surface's scale is what the compositor asked for and changes the
player's surface to the appropriate size, causing no scaling to be done
by the compositor.
Compositors not supporting this interface will ignore the callbacks and do
nothing, keeping program behaviour the same. For compositors supporting
and using this interface (mutter), this will fix the rendering to be pixel
precise as it should be.
Both the opengl wayland backend and the wayland vo have been fixed to support
this. Verified to not break either on weston and mutter.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Commit 0348cd08 was too naive/simple, and always inserted the d3d11vpp
filter if any d3d11 output image formats were supported, even if it
makes no sense. For example --vf=format=rgb8 already breaks it.
It needs to take the set of supported input formats into account. the
weird format negotiation makes this hard. As a simple and cheap
solution, make some assumptions about the supported formats of a filter.
I hope to simplify this one day by using another format negotiation
algorithm, but this can probably wait.
No reason to do so. See also commit 240ba92b.
Since now many mp_images will never have a pixel aspect ratio set,
redefine a 0/0 aspect ratio to "undefined" instead invalid. This also
brings it more in line with how decoder vs. container aspect ratios are
handled.
Most callers seem to be fine with the new behavior.
mp_image_params_valid() in particular has to be adjusted, or some things
stop working due to mp_images not becoming valid after setting size and
format.
Position the window around the original window center on video size change
(when switching to the next file with different resolution, for example)
instead of keeping the position of its top-left corner fixed.
This is quite unexpected. It's caused by mp_image_set_size(), which is
used to update certain fields which can be format-dependent, but which
is actually also supposed to reset the pixel aspect ratio.
We now have a video filter that uses the d3d11 video processor, so it
makes no sense to have one in the VO interop code. The VO uses it for
formats not directly supported by ANGLE (so the video data is converted
to a RGB texture, which ANGLE can take in).
Change this so that the video filter is automatically inserted if
needed. Move the code that maps RGB surfaces to its own inteorp backend.
Add a bunch of new image formats, which are used to enforce the new
constraints, and to automatically insert the filter only when needed.
The added vf mechanism to auto-insert the d3d11vpp filter is very dumb
and primitive, and will work only for this specific purpose. The format
negotiation mechanism in the filter chain is generally not very pretty,
and mostly broken as well. (libavfilter has a different mechanism, and
these mechanisms don't match well, so vf_lavfi uses some sort of hack.
It only works because hwaccel and non-hwaccel formats are strictly
separated.)
The RGB interop is now only used with older ANGLE versions. The only
reason I'm keeping it is because it's relatively isolated (uses only
existing mechanisms and adds no new concepts), and because I want to be
able to compare the behavior of the old code with the new one for
testing. It will be removed eventually.
If ANGLE has NV12 interop, P010 is now handled by converting to NV12
with the video processor, instead of converting it to RGB and using the
old mechanism to import that as a texture.
Main use: deinterlacing.
I'm not sure how to select the deinterlacing mode at all. You can
enumate the available video processors, but at least on Intel, all of
them either signal support for all deinterlacers, or none (the latter is
apparently used for IVTC). I haven't found anything that actually tells
the processor _which_ algorithm to use.
Another strange detail is how to select top/bottom fields and field
dominance. At least I'm getting quite similar results to vavpp on Linux,
so I'm content with it for now.
Future plans include removing the D3D11 video processor use from the
ANGLE interop code.
This avoids a copy of the video image and lowers vsync jitter. Since
there are now two options to add to the window_attribs list, it has been
made dynamic.
This makes vf_vdpaupp use the deinterlacer helper code already used by
vf_vavpp. I nice side-effect is that this also removes some traces of
code originating from vo_vdpau.c, so we can switch it to LGPL.
Extend the refqueue helper with a deint setting. If not set,
mp_refqueue_should_deint() always returns false, which slightly
simplifies vf_vdpaupp. It's of no consequence to vf_vavpp (other than it
has to set it to get expected behavior).
A lot of real-world shaders start off with comments explaining the usage
or license, generating lots of "empty" passes. This simply change allows
us to skip them, which silences the warning spam and prevents us from
having to store and copy around these empty passes.
It also adds a more useful failure check: Attempting to use a user
shader that doesn't define any passes at all.
This requires the GL_EXT_texture_norm16 extension and works in ANGLE.
A default precision had to be set for sampler3Ds, otherwise the shaders
would fail to compile.
This has often been requested for use on OSD. I don't really like having
such "special" properties, but whatever. Hopefully this will be the only
case.
Untested because I'm too damn lazy.
Fixes#2828.
Remove the opengl-hq option default that caused it not to autoselect
ANGLE (unlike --vo=opengl). Details see commit d5df90a2.
Back then the intention was to use ANGLE by default, since it integrates
much nicer with the Windows compositor (instead of native OpenGL, which
tends to cause crazy glitches). On the other hand, many opengl-hq
capabilities are not available with older ANGLE builds, so it didn't
make any sense to autoselect ANGLE for it.
With the GL_EXT_texture_norm16 extension recently added to ANGLE, it has
essentially reached feature parity to desktop GL for the subset we are
using. (Even the integer texture hack for high bit depth input could be
dropped now.)
It (probably) still does not support nnedi3, due to the weird way the NN
coefficients are imported. Also, it uses half-floats instead of 16 bit
fixed-point textures for technical reasons, which implies about 5 bits
of precision loss. If anyone actually manages to distinguish the two
dithering texture formats in a double-blind test, I will fix it.
Abstracts the annoying framerate-doubling behavior.
Same deal as with refqueue introduction: the code size blows up, but at
least it can be reused for other filters.
Calling this right at start of filter_ext() also fixes a small
regression from previous commit. The change in reference surfaces due to
the first update_pipeline() with deinterlacing enabled changed behavior
of mp_refqueue_next() and mp_refqueue_has_output(). Since
update_pipeline() was called between those, the frame output logic got
inconsistent, and the first deinterlaced frame was duplicated from the
previous non-deinterlaced frame.
Also reset the number of ref-frames when switching back to non-deint
mode.
If the deinterlacer separates fields, the framerate must be doubled.
Since we have no stable and reliably framerate anywhere, we've been
calculating it by taking the time halfway to the next frame.
vf_vavpp actually used the past frame to calculate the frame duration,
which is sort of ok, but will skip the 2nd field in a stream (since the
first frame has no past PTS). This is annoying for testing, so use the
future frame PTS instead, which means the last field of the stream will
be dropped instead.
Move the handling of the future/past frames and the associated dataflow
rules to a separate source file.
While this on its own seems rather questionable and just inflates the
code, I intend to reuse it for other filters. The logic is annoying
enough that it shouldn't be duplicated a bunch of times.
(I considered other ways of sharing this logic, such as an uber-
deinterlace filter, which would access the hardware deinterlacer via a
different API. Although that sounds like kind of the right approach,
this would have other problems, so let's not, at least for now.)
SRW locks are available since Windows Vista. They work essentially like
Linux futexes. In particular, they can be statically initialized, and do
not require deinitialization. This makes them ideal for implementing
PTHREAD_MUTEX_INITIALIZER.
We still need CRITICAL_SECTION for recursive mutexes.
This must be called if a texture shared between D3D devices is updated.
Often enough, the shared devices will be the same device, but ANGLE
forces using shared surfaces. I suppose there is no guarantee the driver
will do the expected thing. Internally, the driver could for example not
insert the required barriers before the shared texture is used.
Fixes#320 (which is closed as 'not our problem' but eh)
Relevant xorg bug: https://bugs.freedesktop.org/show_bug.cgi?id=70931
For me this happened when (accidentally) trying to play a 8460x2812 jpg
file with mpv. Like the referenced bug, xvinfo reports "maximum XvImage
size: 8192 x 8192". So the returned XvImage is 8192x2812 and memory
corruption happens.
Only after handling this BadShmSeg X11 errors are shown.
Rename it to get out of OpenGL's namespace. The gl_ prefix is used by
other mpv functions, but no OpenGL ones.
The "slice" parameter was never actually used, and all callers passed 0
for it.
The main change is actually that e first copy to a "staging" memory
frame, and then upload this at once. The old non-PBO code called
glTexsubImage2D for each OSD sub-bitmap.
The new non-PBO code path is a bit faster now if there are many small
sub-bitmaps (on Linux/nVidia). It's also a bit simpler, so this is a
win.
(Although I don't particularly appreciate the mixed normal/PBO texture
code.)
Some of these checks became pointless after dropping ES 2.0 support for
extended filtering.
GL_EXT_texture_rg is part of core in ES 3.0, and we already check for
this version, so testing for the extension is redundant.
GL_OES_texture_half_float_linear is also always available, at least as
far as our needs go.
The functionality we need from GL_EXT_color_buffer_half_float is always
available in ES 3.2, and we explicitly check for ES 3.2, so reject this
extension if the ES version is new enough.
For some reason, GLES has no glMapBuffer, only glMapBufferRange.
GLES 2 has no buffer mapping at all, and GL 2.1 does not always have
glMapBufferRange. On those PBOs remain unsupported (there's no reason to
care about GL 2.1 without the extension).
This doesn't actually work on ANGLE, and I have no idea why. (There are
artifacts on OSD, as if parts of the OSD data weren't copied.) It works
on desktop OpenGL and at least 1 other ES 3 implementation. Don't enable
it on ANGLE, I guess.
Not sure how much can be gained with this, as we can't use it properly
yet. For now, this is used only before rendering, which probably does
overwhelmingly nothing.
In the future, this should be used after temporary passes, which could
possibly reduce memory usage and even memory bandwidth usage, depending
on the drivers.
Center window position after applying W and H parameters of the --geometry
option. Passing valid X and Y values will still override the position.
Fixes#2397.
Before that position of the window top-left corner was remaining the same
when the window was scaled.
Right now VOCTRL_SET_UNFS_WINDOW_SIZE is called only by window-scale. This
change will not affect resizes made by the user (dragging window edge).
Fixes#3164.
Center the window on the original window center instead of the screen center
when the window has been resized due to requested window size exceeding the
size of the screen.
If user moved the window, he probably did it for the reason and he probably
don't want it to get back to the center of the screen when he is resizing it
(with window-scale for example).
Properly update stored client area size when the window is resized in
reinit_window_state due to window size exceeding the size of the screen.
This was causing wrong behavior with window-scale - when window size was
becoming too big the window was resized but the video was not.
I've got a broken webm that fails to seek correctly with "--start=0".
The problem is that every index entry points to 1 byte before cluster
start (!!!). demux_mkv tries to resync to the next cluster, but since it
already has read 2 bytes with ebml_read_id(), it doesn't get the first
cluster, but the following one. Actually, it can be any amount of bytes
from 1-4, whatever happens to look valid at this essentially random byte
position.
Improve this by resyncing from the original position, instead of the one
after the EBML element ID has been attempted to be read.
The file shows the following headers:
| + Muxing application: google at 177
| + Writing application: google at 186
Indeed, the file was downloaded with youtube-dl. I can only guess that
Google got it completely wrong.
Following commit 84ccebd9, the internal helpers don't allow GL_RGB and
GL_RGBA as internal formats for FBO attachments anymore.
While OpenGL itself is perfectly fine with it, I don't see much of a
reason to bother, and mixing sized and unsized internal formats is
confusing anyway.
Just remove these formats.
This code evolved into an ifdef mess as support for cancellation on
Windows was added. Make the Windows-specific code completely separate.
It looks cleaner, and it also means that some of the posix code is not
uselessly enabled on Windows. The latter made msvcrt.dll output warnings
because it does not like -1 passed as FD to read/write. (The same would
be harmless on POSIX.)
Cropping usually happens by adjusting the plane start pointers and the
image size. The former is obviously not possible for opaque hwaccel
formats, but the latter must work.
Since the code already takes care of aligning the top/left crop origin
to chroma alignment, simply set the crop origin to 0/0 in the hwaccel
case. Also add a message if such an adjustment happens.
Supporting this isn't worth much; the main usefulness is with debugging.
ES 2.0 has this weird rule that not the internalformat parameter
determines the internal format, but the combination of all texture
parameters. GL_OES_texture_half_float thus does not specify e.g. a
GL_RGBA16F format, but requires passing GL_RGBA as format and
GL_HALF_FLOAT_OES as type. We won't bother with this, since ES 2.0 is a
lost cause anyway.
This also removes the OpenGL error when the code is trying to create a
f16 FBO for testing whether FBOs work.
gl_video_upload_image() can fail in the hardware decoding case. In this
case rendering continued "normally", which meant that pass_get_img_tex()
would kill the process with an assertion failure.
Fix this by allowing gl_video_upload_image() to fail, and exit rendering
early enough to skip code which requires an image to be present. (Maybe
this is still a bit too subtle, but better than before.)
Set an error flag, and render the blue screen we introduced for shader
errors. (For this purpose also move the rendering of it to final output,
to ensure it's visible at all.) The error flag is temporary, because the
associated failure might also be temporary, unlike shader compilation
errors.
ANGLE doesn't handle this very strictly. But if they change this in the
future, it shouldn't brick us.
Not quite happy with this glsl_extensions fields, but it is quite
unintrusive after all.
No reason not to, and makes the following commit slightly simpler.
In fact, this makes the shaders more correct too. Normally, "#extension"
must come before any normal shader text, including the "precision"
directive. Not sure why this worked before. (Probably didn't.)
ANGLE was missing texture() overloads in the shader compiler for
GL_TEXTURE_EXTERNAL_OES textures. Support has been added upstream,
so we can use it now.
With the new hooks mechanism, user shaders and such are actually loaded
before rendering starts, instead of being loaded during rendering. This
is used to cache them (instead of e.g. reparsing them every frame).
The cached state wasn't cleared correctly in some situations. Namely,
resizing didn't correctly enable/disable prescale hooks.
Reorganize how these reinitializations are handled. Get rid of
reinit_rendering(), whose meaning was pretty unclear. Call the required
functions to reset or recreate state directly wherever they are needed.
For some reason, the d3d9/dxva2/d3d11 DLLs are still optional. But we
don't need to try so hard to keep exact references. In fact, there's no
reason to unload them at all.
So load them once in a central place. For simplicity, the d3d9/d3d11
backends both load all DLLs. (They will error out only if the required
DLLs could not be loaded.)
In theory, we could just call LoadLibrary multiple times (without
calling FreeLibrary), but I'm slightly worried that this could be
detected as a "bug", or that the reference count could even have a low
static limit that could be hit soon.
wscript builds hwdec_dxva2gldx.c if gl-dxinterop is enabled, while
video/dxva2.c depends on d3d-hwaccel. If d3d-hwaccel is disabled, then
hwdec_dxva2gldx.c will fail to link, because it uses
d3d9_surface_in_mp_image(), defined in dxva2.c.
Fix this by removing the use of this function. It has barely any value
at this point anyway. Just use the libavcodec documented way to get the
surface directly.
Fixes#3150.
Use dynamic memory allocation, as the static allocation is starting to
get annoying.
Currently, SC_MAX_ENTRIES is essentially still a static upper limit on
the number of shaders. But in future we could try a more clever cache
replacement strategy, which does not keep stale entries forever if the
maximum happens not to be reached.
The new uniforms introduced by 362015c have exceeded the uniform limit
when using high-radius tscale. In addition, the SC limit of 32 entries
might be pushing it with user shaders.
Just make these value a bigger to delay the onset of this same failure
mode. Maybe in the future it should be reworked to grow dynamically?
Either way, we *can* always predict a static upper bound on the number
of uniforms and shader cache entries, it's just that we forgot to do so.
Fixes#3151
This makes it so that users with actual HDR displays can just set their
config to target-trc=st2084 and get native HDR output. This will look a
bit silly for SDR content (everything will be really bright), but for
lack of a better tone mapping situation (including reverse tone mapping)
this is the easiest thing to do for now.
Ideally the brightness metadata should be part of the colorspace struct
or something (with mpv always adapting where necessary), but it depends
on the TRC and not the primaries so it's a bit more complicated than
that.
Since dumb mode is affected by tone mapping (which I'll call a feature,
not a bug), we need to copy over the configuration - in particular, the
defaults. (To prevent a render failure)
Since HDR content is now auto-detected as such, we should probably do
something smarter in the "no configuration" case, such as outputting
gamma 2.2 instead.
This decision will affect the majority of users of stock configurations
who just play back appropriately tagged HDR files, so having a good
default behavior is important. "Output the HDR content as-is" is
definitely not likely to give the user a good result.
This now lets us auto-detect appropriately tagged HDR content using
FFmpeg's new TRC entries (when available).
Hidden behind an #if because Libav stable doesn't have it yet.
Make it dynamic and never remove entries from it.
For now, this is better than possibly creating dangling pointers all
over the place in the gl_user_shader struct.
Untested.
This is now a configurable option, with tunable parameters.
I got inspiration for these algorithms off wikipedia. "simple" seems to
work pretty well, but not well enough to make it a reasonable default.
Some other notable candidates:
- Local functions (e.g. based on local contrast or gradient)
- Clamp with soft knee (linear up to a point)
- Mapping in CIE L*Ch. Map L smoothly, clamp C and h.
- Color appearance models
These will have to be implemented some other time.
Note that the parameter "peak_src" to pass_tone_map should, in
principle, be auto-detected from the SEI information of the source file
where available. This will also have to be implemented in a later
commit.
Due to the way color management in mpv worked historically, the subtitle
blending function was written to preserve the linearity of the input.
(In the past, the 3DLUT function required linear inputs)
Since the 3DLUT was refactored to accept the video color directly, the
re-linearization after blending is now virtually always redundant.
(Notably, it's also redundant when CMS is turned off, so this way of
writing the code stopped making sense a long time ago. It is a remnant
from before the pass_colormanage function was as flexible as it is now)
Currently, this relies on the user manually entering their display
brightness (since we have no way to detect this at runtime or from ICC
metadata). The default value of 250 was picked by looking at ~10 reviews
on tftcentral.co.uk and realizing they all come with around 250 cd/m^2
out of the box. (In addition, ITU-R Rec. BT.2022 supports this)
Since there is no metadata in FFmpeg to indicate usage of this TRC, the
only way to actually play HDR content currently is to set
``--vf=format=gamma=st2084``. (It could be guessed based on SEI, but
this is not implemented yet)
Incidentally, since SEI is ignored, it's currently assumed that all
content is scaled to 10,000 cd/m^2 (and hard-clipped where out of
range). I don't see this assumption changing much, though.
As an unfortunate consequence of the fact that we don't know the display
brightness, mixed with the fact that LittleCMS' parametric tone curves
are not flexible enough to support PQ, we have to build the 3DLUT
against gamma 2.2 if it's used. This might be a good thing, though,
consdering the PQ source space is probably not fantastic for
interpolation either way.
Partially addresses #2572.
This is much more readable than hard-coding magic IDs all over the file,
and removes the need for all the explanatory comments that were a direct
result of this.
This macro takes care of rotation, swizzling, integer conversion and
normalization automatically. I found the performance impact to be
nonexistant for superxbr and debanding, although rotation *did* have an
impact due to the extra matrix multiplication. (So it gets skipped where
possible)
All of the internal hooks have been rewritten to use this new mechanism,
and the prescaler hooks have finally been separated from each other.
This also means the prescale FBO kludge is no longer required.
This fixes image corruption for image formats like 0bgr, and also fixes
prescaling under rotation. (As well as other user hooks that have
orientation-dependent access)
The "raw" attributes (tex, tex_pos, pixel_size) are still un-rotated, in
case something needs them, but ideally the hooks should be rewritten to
use the new API as much as possible. The hooked texture has been renamed
from just NAME to NAME_raw to make script authors notice the change (and
also deemphasize direct texture access).
This is also a step towards getting rid of the use_integer pass.
This replaces the previous TRANSFORM by WIDTH, HEIGHT and OFFSET where
WIDTH and HEIGHT are RPN expressions. This allows for more fine-grained
control over the output size, and also makes sure that overwriting
existing textures works more cleanly.
(Also add some more useful bstr functions)
This allows users to add their own near-arbitrary hooks to the vo_opengl
processing pipeline, greatly enhancing the flexibility of user shaders.
This enables, among other things, user shaders such as CrossBilateral,
SuperRes, LumaSharpen and many more.
To make parsing the user shaders easier, shaders are now loaded as
bstrs, and the hooks are set up during video reconfig instead of on
every single frame.
These are "sequence points" where the image could be rendered out to an
FBO, hooked, and re-loaded if any such hook exists. This is perfect for
things like the current user shaders system, as well as optional effects
like unsharp masking.
Note that since we have to pick *some* FBO to store the optionally
hooked texture, we just store it in an array indexed by an increasing
counter. Since we only ever store as many as MAX_TEXTURE_HOOKS + all
internal hook points entries, this is guaranteed to be enough space.
This commit also removes some of the now unused FBOs.
The hook mechanism allows arbitrary processing stages to get dispatched
whenever certain named textures have been "finalized" by the code.
This is mostly meant to serve as a change that opens up the internal
processing in pass_read_video to user scripts, but as a side benefit all
of the code dealing with offsets and plane alignment and other such
confusing things has been rewritten.
This hook mechanism is powerful enough to cover the needs of both
debanding and prescaling (and more), so as a result they can be removed
from pass_read_video entirely and implemented through hooks.
Some avenues for optimization:
- The prescale hook is currently somewhat distributed code-wise. It might be
cleaner to split it into superxbr and NNEDI3 hooks which can each be
self-contained.
- It might be possible to move a large part of the hook code out to an
external file (including the hook definitions for debanding and
prescaling), which would be very much desired.
- Currently, some stages (chroma merging, integer conversion) will
*always* run even if unnecessary. I'm planning another series of
refactors (deferred img_tex) to allow dropping unnecessary shader
stages like these, but that's probably some ways away. In the meantime
it would be doable to re-add some of the logic to skip these stages if
we know we don't need them.
- More hook locations could be added (?)
Instead of rounding down, we round to the nearest float. This reduces
the maximum possible error introduced by this rounding operation. Also
clarify the comment.
Remove non-texture_rg compatibility from LUT sampling. OpenGL without
texture_rg support will always trigger dumb-mode, and dumb-mode does not
use LUTs. It used not to, and that was when this made sense.
Fixes broken colors with --vf=format=0bgr (but only if deband is
disabled).
0bgr means the first byte is padding, while the following three bytes
are bgr. From the vo_opengl perspective, it has 4 physical components
with 3 logical components. copy_img_tex() simply copied 3 components
from the physical representation, which means the last component (r) was
sliced off.
Fix this by not using p->color_swizzle for packed formats, and instead
let packed formats set the per-plane swizzle in texplane.swizzle. The
latter applies the swizzle as part of operation in copy_img_tex(), which
essentially moves physical to logical representations.
Unfortunately, debanding (and thus with opengl-hq defaults) is still
broken.
Make the find_plane_format function take a bit count.
This also makes the function's comment true for the first time the
function and its comment exist. (It was commented as taking bits, but
always took bytes.)
Only a few very low bit depth internal formats can be rendered to in
pure ES2 (GL_RGB565 is the "best" one).
Seems like the only potentially reasonable renderable formats in ES2
could be provided via GL_OES_rgb8_rgba8, or half-floats, so don't
bother with this at all.
Remove the attempted cleverness; keep it dumb.
This strictly calculates the average speed over an at least 1 second
window (longer if I/O blocks it).
Since this doesn't reset the speed anymore when reading stops by going
idle, the results might actually be more accurate now.
Now that we know in advance whether an implementation should support a
specific format, we have more flexibility when determining which format
to use.
In particular, we can drop the roundabout ES logic.
I'm not sure if actually trying to create the FBO for probing still has
any value. But it might, so leave it for now.
Even if everything else is available, the need for first class arrays
breaks it. In theory we could fix this since we don't strictly need
them, but I guess it's not worth bothering.
Also give the misnamed have_mix variable a slightly better name.
This merges all knowledge about texture format into a central table.
Most of the work done here is actually identifying which formats exactly
are supported by OpenGL(ES) under which circumstances, and keeping this
information in the format table in a somewhat declarative way. (Although
only to the extend needed by mpv.) In particular, ES and float formats
are a horrible mess.
Again this is a big refactor that might cause regression on "obscure"
configurations.
This uses the normal autoprobing rules like "auto", but rejects anything
that isn't flagged as copying data back to system memory.
The chunk in command.c was dead code, so remove it instead of updating
it.
MSDN documents this as "Introduced in Windows 8.1.". I assume on Windows
7 this field will simply be ignored. Too bad for Windows 7 users.
Also, I'm not using D3D11_VIDEO_PROCESSOR_NOMINAL_RANGE_16_235 and
D3D11_VIDEO_PROCESSOR_NOMINAL_RANGE_0_255, because these are apparently
completely missing from the MinGW headers. (Such a damn pain.)
We don't have any reason to disable either. Both are loaded dynamically
at runtime anyway. There is also no reason why dxva2 would disappear
from libavcodec any time soon.
ANGLE is _really_ annoying to build. (Requires special toolchain and a
recent MSVC version.) This results in various issues with people
having trouble to build mpv against ANGLE (apparently linking it
against a prebuilt binary doesn't count, or using binaries from
potentially untrusted sources is not wanted).
Dynamically loading ANGLE is going to be a huge convenience. This commit
implements this, with special focus on keeping it source compatible to
a normal build with ANGLE linked at build-time.
In theory this was needed for the previous commit (but wasn't in
practice, since for hwdec the LUMINANCE_ALPHA mangling is not applied
anymore, and ANGLE uses RG textures in absence of GL_ARB_texture_rg for
whatever crazy reasons).
In practice this caused funky colors on OSX with the uyvy422 format,
which is also fixed in this commit.
This uses EGL_ANGLE_stream_producer_d3d_texture_nv12 and related
extensions to map the D3D textures coming from the hardware decoder
directly in GL.
In theory this would be trivial to achieve, but unfortunately ANGLE does
not have a mechanism to "import" D3D textures as GL textures. Instead,
an awkward mechanism via EGL_KHR_stream was implemented, which involves
at least 5 extensions and a lot of glue code. (Even worse than VAAPI EGL
interop, and very far from the simplicity you get on OSX.)
The ANGLE mechanism so far supports only the NV12 texture format, which
means 10 bit won't work. It also does not work in ES3 mode yet. For
these reasons, the "old" ID3D11VideoProcessor code is kept and used as a
fallback.
It forces es2 mode on ANGLE. Only useful for testing. Since the normal
"angle" backend already falls back to es2 if es3 does not work, this new
backend always exit when autoprobing it.
Rename gl_hwdec_driver.map_image to map_frame, and let it fill out a
struct gl_hwdec_frame describing the exact texture layout. This gives
more flexibility to what the hwdec interop can export. In particular, it
can export strange component orders/permutations and textures with
padded size. (The latter originating from cropped video.)
The way gl_hwdec_frame works is in the spirit of the rest of the
vo_opengl video processing code, which tends to put as much information
in immediate state (as part of the dataflow), instead of declaring it
globally. To some degree this duplicates the texplane and img_tex
structs, but until we somehow unify those, it's better to give the hwdec
state its own struct. The fact that changing the hwdec struct would
require changes and testing on at least 4 platform/GPU combinations
makes duplicating it almost a requirement to avoid pain later.
Make gl_hwdec_driver.reinit set the new image format and remove the
gl_hwdec.converted_imgfmt field.
Likewise, gl_hwdec.gl_texture_target is replaced with
gl_hwdec_plane.gl_target.
Split out a init_image_desc function from init_format. The latter is not
called in the hwdec case at all anymore. Setting up most of struct
texplane is also completely separate in the hwdec and normal cases.
video.c does not check whether the hwdec "mapped" image format is
supported. This should not really happen anyway, and if it does, the
hwdec interop backend must fail at creation time, so this is not an
issue.
Older ffmpeg releases don't have ffmpeg git commit
50401f5fb7d778583b03a13bc4440f71063d319d, which fixes ffmpeg's
pkt_timebase check to reject its default "unset" timebase as invalid.
The consequence was that all non-PGS bitmap subtitle timestamps were
forced to 0.
Of course this hit _only_ shitty distros using outdated/badly maintained
ffmpeg releases, so this is not worth working around. I've already
wasted a lot of time on analyzing this dumb issue, and it could be
useful for bisecting, so don't drop pre-3.0 ffmpeg just yet.
Fixes#3109.
libass's ass_process_chunk expects long long int for the timecode and
durations arguments, thus should use llrint instead of lrint.
This does not cause any problems on most platforms, but on cygwin, it
causes strange subtitle behaviour, like subtitles not showing, getting
stuck or old subtitles showing at the same time as new subtitles.
Commit 786f37ae accidentally changed seeking behavior such that
continuous seeking (holding the seek button down) would use the previous
seek target timestamp, instead of the new video timestamp. (This is for
the default mode, seeking to keyframes.)
The result is that the movement on the seekbar is smooth, but the way
the video updates is awkward. Some might actually prefer the new
behavior (and some players effectively show similar bahavior), but I
don't. So restore the old behavior.
This is done in two steps:
First: strictly wait for the entire seek process to finish, which will
effectively make the seeking code pick up the new video timestamp
correctly.
This would play audio immediately, which would result in noise during
continuous seeking, which leads to second: explicitly abort the playback
restarting process if this case is detected, and never play audio.
The main change is with video/hwdec.h. mp_hwdec_info is made opaque (and
renamed to mp_hwdec_devices). Its accessors are mainly thread-safe (or
documented where not), which makes the whole thing saner and cleaner. In
particular, thread-safety rules become less subtle and more obvious.
The new internal API makes it easier to support multiple OpenGL interop
backends. (Although this is not done yet, and it's not clear whether it
ever will.)
This also removes all the API-specific fields from mp_hwdec_ctx and
replaces them with a "ctx" field. For d3d in particular, we drop the
mp_d3d_ctx struct completely, and pass the interfaces directly.
Remove the emulation checks from vaapi.c and vdpau.c; they are
pointless, and the checks that matter are done on the VO layer.
The d3d hardware decoders might slightly change behavior: dxva2-copy
will not use the VO device anymore if the VO supports proper interop.
This pretty much assumes that any in such cases the VO will not use any
form of exclusive mode, which makes using the VO device in copy mode
unnecessary.
This is a big refactor. Some things may be untested and could be broken.
When we receive the wl_shell_surface::configure event, it makes sense
to respect the aspect ratio of the video in windowed mode, but in
fullscreen it forces compositing and wastes resources (until atomic
modesetting is available everywhere and we can stop having
desynchronised planes).
Weston mitigates a resolution mismatch by creating black surfaces and
compositing them around the fullscreen surface, placed at the middle,
while GNOME puts it at the top-left and leaves the rest of the desktop
composited below, both of them producing a subpar experience.
Fixes#3021, #2657.
Add --taskbar-progress command line option and property which controls taskbar
progress indication rendering in Windows 7+. This option is on by default and
can be toggled during playback.
This option does not affect the creation process of ITaskbarList3. When the
option is turned off the progress bar is just hidden with TBPF_NOPROGRESS.
Closes#2535
The X11 error handler is global, and not per-display. If another Xlib
user exists in the process, they can conflict. In theory, it might
happen that e.g. another library sets an error handler (overwriting the
mpv one), and some time after mpv closes its display, restores the error
handler to mpv's one. To mitigate this, check if the error log instance
is actually set, instead of possibly crashing.
The change in vo_x11_uninit() is mostly cosmetic.
Some of this documentation was left woefully inaccurate as color
management in mpv evolved. This commit updates all of the wording and
adds notes and comments where appropriate.
This reportedly makes it work on ODROID-C2. The idea for this hack is
taken from kodi; they unconditionally set some or all of those flags.
I don't trust ALSA enough to hope that setting these flags couldn't
break something else, so we try without them first.
It's not clear whether this is a driver bug or a bug in the ALSA libs.
There is no ALSA bug tracker (the ALSA website has had a dead link to
a deleted bug tracker fo years). There's not much we can do other than
piling up ridiculous hacks. At least I think that at this point invalid
API usage by mpv can be excluded as a cause.
ALSA might be the worst audio API ever.
In order to honor the differences between OpenGL and Direct3D coordinate
systems, ANGLE uses a full FBO copy merely to flip the final frame
vertically. This can be avoided with the EGL_ANGLE_surface_orientation
extension.
I hope that this does what we expect it does: destroy the EGLDisplay
specific to our HDC. (Some implementations will terminate all EGL
contexts in the whole process.)
eglReleaseThread() merely calls eglMakeCurrent(0, 0, 0, 0), which is
not enough.
This commit also fixes the problem fixed with the previous commit,
but I think both changes are needed to make our API usage clean.
If ANGLE was probed before (but rejected), the ANGLE API can remain
"initialized", and eglGetCurrentDisplay() will return a non-NULL
EGLDisplay. Then if a native GL context is used, the ANGLE/EGL API will
then (apparently) keep working alongside native OpenGL API. Since GL
objects are just numbers, they'll simply fail to interact, and OpenGL
will get invalid textures. For some reason this will result in black
textures.
With VAAPI-EGL, something similar could happen in theory, but didn't in
practice.
Options/properties that are choices, and which include "yes" or "no"
values (or both) can now be read and written as MPV_FORMAT_FLAG.
For write access, rejecting flags in these cases was obnoxiously
unintuitive and inconvenient.
For read access, the value of this is less convincing, and actually it's
a major API change. At this point I probably have to admit that the
finer details of the client API are very unstable.
Introduce hwdec-current and hwdec-interop properties.
Deprecate hwdec-detected, which never made a lot of sense, and which is
replaced by the new properties. hwdec-active also becomes useless, as
hwdec-current is a superset, so it's deprecated too (for now).
Cache misses are a normal and expected part of the operation of a cache.
It doesn't really make sense to show a user-visible warning for them.
To work-around this, just skip trying to open the cache if it doesn't
exist yet.
First of all, black point compensation is now on by default. This is
really rather harmless and only improves the result (where "improvement"
means "less black clipping").
Second, this adds an option to limit the ICC profile's contrast, which
helps for untagged matrix profiles that are implicitly black scaled even
in colorimetric intent. (Note that this relies on BPC being enabled to
work properly, which is why the two changes are tied together)
Third, this uses the LittleCMS built in black point estimator instead of
relying on the presence of accurate A2B tables. This also checks tags
and does some amounts of noise elimination.
If the option is unspecified and the profile is missing black point
information, print a warning instructing the user to set the option, and
fall back to 1000 otherwise.
For "current" markers on OSD properties like chapter-list. The marker is
now an actual arrow instead of "> ", and non-current entries will have
the same indentation as the current entry.
While I'm not entirely sure about the new look of those lists, it's a
bit better than the visual mess that was before.
The vdpau_mixer could fail to be recreated properly if preemption
occured at some point before playback initialization (like when using
--hwdec-preload and the opengl-cb API).
Normally, the vdpau_mixer was supposed to be marked invalid when the
components using it detect a preemption, e.g. in hwdec_vdpau.c. This one
didn't mark the vdpau_mixer as invalid if preemption was detected in
reinit(), only in map_image().
It's cleaner to detect preemption directly in the vdpau_mixer, which
ensures it's always recreated correctly.
The "fs-only" choice sets the _NET_WM_BYPASS_COMPOSITOR to 1 if the
window is fullscreened, and 0 otherwise. (0 is specified to be the
implicit default - i.e. no change is requested in windowed mode.)
In particular, change the default to "fs-only".
Fixes#2582.
Including initguid.h at the top of a file that uses references to GUIDs
causes the GUIDs to be declared globally with __declspec(selectany). The
'selectany' attribute tells the linker to consolidate multiple
definitions of each GUID, which would be great except that, in Cygwin
and MinGW GCC 6.1, this method of linking makes the GUIDs conflict with
the ones declared in libuuid.a.
Since initguid.h obsoletes libuuid.a in modern compilers that support
__declspec(selectany), add initguid.h to all files that use GUIDs and
remove libuuid.a from the build.
Fixes#3097
Flag that is set by default. Reseting it will result in mpv trying to fit
client area with video instead of the whole window with border and
decorations on the screen.
Marked as (Windows only) for now until it's implemented on other platforms.
Substraction of 1 is not necessary due to .right and .bottom values of RECT
struct describing a point one pixel outside of the rectangle.
Fixes#2935. (2.b)
--sub-ass=no / --ass=no still work, but --ass-style-override=strip is
preferred now. With this change, --ass-style-override can control all
the types of style overriding.
In particular, this moves the depth test to common code.
Should be functionally equivalent, except that for DXVA2, the
IDirectXVideoDecoderService_GetDecoderRenderTargets API is called
more often potentially.
Because it's annoying and feels unnatural.
If the B point is set while paused, don't seek. If not paused, it should
properly loop immediately.
In theory there's a chance that it will show at least 1 frame after the
loop point when setting the B point. But let's not care about that.
This gives us 16 bit fixed-point integer texture formats, including
ability to sample from them with linear filtering, and using them as FBO
attachments.
The integer texture format path is still there for the sake of ANGLE,
which does not support GL_EXT_texture_norm16 yet.
The change to pass_dither() is needed, because the code path using
GL_R16 for the dither texture relies on glTexImage2D being able to
convert from GL_FLOAT to GL_R16. GLES does not allow this. This could be
trivially fixed by doing the conversion ourselves, but I'm too lazy to
do this now.
This fixes backstepping getting "stuck" when e.g. holding down a key
bound to the backstep command. The reason is that even if the backstep
itself is finished, the next backstep might not take the new video PTS
as reference if the hr-seek itself isn't finished yet.
The intention of not waiting for the hr-seek to finish was faster
backstepping by possibly skipping audio decoding. But it probably
doesn't matter enough to make the rest of the code more complex.
Basically this gets rid of the need for the accessors in d3d11va.h, and
the code can be cleaned up a little bit.
Note that libavcodec only defines a ID3D11VideoDecoderOutputView pointer
in the last plane pointers, but it tolerates/passes through the other
plane pointers we set.
This uses ID3D11VideoProcessor to convert the video to a RGBA surface,
which is then bound to ANGLE. Currently ANGLE does not provide any way
to bind nv12 surfaces directly, so this will have to do.
ID3D11VideoContext1 would give us slightly more control about the
colorspace conversion, though it's still not good, and not available
in MinGW headers yet.
The video processor is created lazily, because we need to have the coded
frame size, of which AVFrame and mp_image have no concept of. Doing the
creation lazily is less of a pain than somehow hacking the coded frame
size into mp_image.
I'm not really sure how ID3D11VideoProcessorInputView is supposed to
work. We recreate it on every frame, which is simple and hopefully
doesn't affect performance.
As a positive side-effect, this also errors out gracefully for the
extremely unlikely but possible case certain builtin filters are not
available. (This could happen only with crippled libavfilter builds that
can't be used by anything using its public API.)
Setting this here is a race condition. It's called from a CoreAudio
callbacks, and there are no locks. It's a string, so this can be
potentially severe.
It's hard to fix and only CoreAudio supported it, so remove it.
This causes the "audio-out-detected-device" property to return nothing
on all platforms.
Sucks, but better than freezing forever given the (to me) unpredictable
RPI behavior. This will be good enough to drop out of vsync timing mode,
or to abort playback.
Recreate all dispmanx objects after mode changes signalled by the TV
callback. This is needed since dispmanx objects are marked as invalid
and cease working.
One important point is that the vsync callbacks will stop coming when
this happens, so restoring the callback is important.
Note that the MMAL renderer itself does not get trashed by the firmware
on such events, but we completely reconfigure it anyway when it happens.
For Mediacodec in particular we don't care about the format. It can just
decode to whatever it wants. The only case we would care about is it not
returning an opaque format if we don't have proper interop, but
libavcodec always returns non-opaque formats by default.
Use the recently added lavc_suffix mechanism to select the wrapper
decoder.
With all hwdec callbacks being optional, and RPI/Mediacodec having only
dummy callbacks, all the callbacks can be removed as well.
The result is that the vd_lavc_hwdec struct for both of them is tiny.
It's better to move them to vd_lavc.c directly, because they are so
trivial and small.
This is intended for cases when --hwdec needs to override the decoder
implementation in use, like for example on the RPI.
It does two things:
1. Allow the hwdec to indicate a decoder suffix. libavcodec by
convention adds a suffix to all wrapper decoders, and here we start
relying on it. While not necessarily the best idea, it's the only
thing we got. libavcodec's hwaccel list is useless, because it only
has the codec ID, not the associated decoder's name.
2. Make --hwdec=auto work properly. It shouldn't fail anymore, and hwdec
probing should reliably work, even if a different decoder is selected
with --vd. The semantics of --hwdec should dictate that it overrides
the default decoder.
A minor simplification. Most callers don't need this, and there's no
good reason why the caller should provide an "initializer" like this.
(This function calls mp_image_new_dummy_ref(), which has no reason
for an initializer either.)
Another crappy fix for timestamp reset issues. This time, we try to fix
files which have very weird but legitimate frame durations, such as
cdgraphics. It can have many short frames, but once in a while there are
potentially very long frames.
Fixes#3027.
The active texture and some pixelstore parameters are now always reset
to defaults when entering and leaving the renderer. Could be important
for libmpv.
Commit 382bafcb changed the behavior for ab-loop-a. This commit changes
ab-loop-b so that the behavior is symmetric.
Adjust the OSD rendering accordingly to the two changes.
Also fix mentions of the "ab_loop" command to the now preferred
"ab-loop".
The check whether video is ready yet was done only in STATUS_FILLING.
But it also switched to STATUS_READY, which means the next time
fill_audio_out_buffers() was called, audio would actually be started
before video.
In most situations, this bug didn't show up, because it was only
triggered if the demuxer didn't provide video packets quickly enough,
but did for audio packets.
Also log when audio is started.
(I hate fill_audio_out_buffers(), why did I write it?)
Strictly schedule an update in regular intervals as long as either
stream cache or demuxer are prefetching. Don't update just always
because the stream cache is enabled ("idle != -1") or cache-related
properties are observed (mp_client_event_is_registered()).
Also, the "idle" variable was awkard; get rid of it with equivalent
code.
Calculate the buffering percentage in the same code which determines
whether the player is or should be buffering. In particular it can't
happen that percentage and buffering state are slightly out of sync due
to calling DEMUXER_CTRL_GET_READER_STATE and reusing it with the
previously determined buffering state.
Now it's also easier to guarantee that the buffering state is updated
properly.
Add some more verbose output as well.
(Damn I hate this code, why did I write it?)
demux_playlist.c recognizes if the source stream points to a directory,
and adds its directory entries. Until now, only 1 level was added.
During playback, further directory entries could be resolved as
directory paths were "played".
While this worked fine, it lead to frequent user confusion, because
playlist resuming and other things didn't work as expected. So just
recursively scan everything.
I'm unsure whether it's a good fix, but at least it gets rid of the
complaints. (And probably will add others.)
Commit a9bd4535 generally changed properties are set to string values.
This actually broke the fallback for non-string properties, because the
set string action was redirected directly to the property, instead of
the generic handler and its fallback code.
As of ffmpeg git master, only the libavdevice decklink wrapper supports
this. Everything else has dropped support.
You're now supposed to use AV_CODEC_ID_WRAPPED_AVFRAME, which works a
bit differently. Normal AVFrames should still work for these encoders.
And remove the same thing from the client API code.
The command.c code has to deal with many specialized M_PROPERTY_SET_*
actions, and we bother with a subset only.
If a mpv_node wrapped a string, the behavior was different from calling
mpv_set_property() with MPV_FORMAT_STRING directly. Change this.
The original intention was to be strict about types if MPV_FORMAT_NODE
is used. But I think the result was less than ideal, and the same change
towards less strict behavior was made to mpv_set_option() ages ago.
Since what we're doing is a linear blend of the four colors, we can just
do it for free by using GPU sampling.
This requires significantly fewer texture fetches and calculations to
compute the final color, making it much more efficient. The code is also
much shorter and simpler.
Now it will always be able to seek back to the start, even if the index
is sparse or misses the first entry.
This can be achieved by reusing the logic for incremental index
generation (for files with no index), and start time probing (for making
sure the first block is always indexed).
Until now, we have made the assumption that a driver will use only 1
hardware surface format. the format is dictated by the driver (you
don't create surfaces with a specific format - you just pass a
rt_format and get a surface that will be in a specific driver-chosen
format).
In particular, the renderer created a dummy surface to probe the format,
and hoped the decoder would produce the same format. Due to a driver
bug this required a workaround to actually get the same format as the
driver did.
Change this so that the format is determined in the decoder. The format
is then passed down as hw_subfmt, which allows the renderer to configure
itself with the correct format. If the hardware surface changes its
format midstream, the renderer can be reconfigured using the normal
mechanisms.
This calls va_surface_init_subformat() each time after the decoder
returns a surface. Since libavcodec/AVFrame has no concept of sub-
formats, this is unavoidable. It creates and destroys a derived
VAImage, but this shouldn't have any bad performance effects (at
least I didn't notice any measurable effects).
Note that vaDeriveImage() failures are silently ignored as some
drivers (the vdpau wrapper) support neither vaDeriveImage, nor EGL
interop. In addition, we still probe whether we can map an image
in the EGL interop code. This is important as it's the only way
to determine whether EGL interop is supported at all. With respect
to the driver bug mentioned above, it doesn't matter which format
the test surface has.
In vf_vavpp, also remove the rt_format guessing business. I think the
existing logic was a bit meaningless anyway. It's not even a given
that vavpp produces the same rt_format for output.
In the past, --video-unscaled also disabled zooming and aspect ratio
corrections. But this didn't make much sense in terms of being a useful
option. The new behavior just sets the initial video size to be
unscaled, but it's still affected by zoom commands and aspect ratio
corrections.
To get the old behavior back, --video-aspect=0 --video-zoom=0 need to be
added as well (in the general case). Most of the time it should not make
a difference though.
Also, there seems to have been some additional dst_rect clamping code
inside src_dst_split_scaling that didn't seem to either be necessary nor
ever get triggered. (The code immediately above it already makes sure to
crop the video if it's larger than the dst_rect)
No idea why it was there, but I just removed it.
Apply basic transformations like rotation by 90° and mirroring when
sampling from the source textures. The original idea was making this
part of img_tex.transform, but this didn't work: lots of code plays
tricks on the transform, so manipulating it is not necessarily
transparent, especially when width/height are switched. So add a new
pre_transform field, which is strictly applied before the normal
transform.
This fixes most glitches involved with rotating the image.
Cropping and rotation are now weirdly separated, even though they could
be done in the same step. I think this is not much of a problem, and
has the advantage that changing panscan does not trigger FBO
reallocations (I think...).
Typically happens with some implementations if no context is currrent,
or is otherwise broken. This is particularly relevant to the opengl_cb
API, because the API user will have no other indication what went wrong.
The underlying intention of this code is to make changing
--videotoolbox-format at runtime work. For this reason, the format can't
just be statically setup, but must be read from the option at runtime.
This means the format is not fixed anymore, and we have to make sure the
renderer is property reinitialized if the format changes. There is
currently no way to trigger reinit on this level, which is why the
mp_image_params.hw_subfmt field was introduced.
One sketchy thing remains: normally, the renderer is supposed to be
involved with VO format negotiation, which would ensure that the VO
can take the format at all. Since the hw_subfmt is not part of this
format negotiation, it's implied the get_vt_fmt() callback only
returns formats supported by the renderer. This is not necessarily
clear because vo_opengl checks this with converted_imgfmt separately.
None of this matters in practice though, because we know all formats
are always supported.
(This still requires somehow triggering decoder reinit to make the
change effective.)
For hwaccel formats, mp_image will merely point to a hardware surface
handle. In these cases, the mp_image_params.imgfmt field describes the
format insufficiently, because it mostly only describes the type of the
hardware format, not its underlying format.
Introduce hw_subfmt to describe the underlying format. It makes sense to
use it with most hwaccels, though for now it will be used with the
following commit only.
Until now, the presence of the process_image() callback was used to set
a delay queue with a hardcoded size. Change this to a vd_lavc_hwdec
field instead, so the decoder can explicitly set this if it's really
needed.
Do this so process_image() can be used in the VideoToolbox glue code for
something entirely unrelated.
Some functions which expected a codec name (i.e. the name of the video
format itself) were passed a decoder name. Most "native" libavcodec
decoders have the same name as the codec, so this was never an issue.
This should mean that e.g. using "--vd=lavc:h264_mmal --hwdec=mmal"
should now actually enable native surface mode (instead of doing copy-
back).
The sync-by-display mode relies on using the vsync statistics for
timing. As a consequence discontinuities must be handled somehow. Until
now we have done this by completely resetting these statistics.
This can be somewhat annoying, especially if the GL driver's vsync
timing is not ideal. So after e.g. a seek it could take a second until
it locked back to the proper values.
Change it not to reset all statistics. Some state obviously has to be
reset, because it's a discontinuity. To make it worse, the driver's
vsync behavior will also change on such discontinuities. To compensate,
we discard the timings for the first 2 vsyncs after each discontinuity
(via num_successive_vsyncs). This is probably not fully ideal, and
num_total_vsync_samples handling in particular becomes a bit
questionable.
Tuning it in a way to be actually useful is too much effort.
As alternative, there's the "buffering" detection, which operates on a
much higher level. The only disadvantage is that it's harder to guess
for the user whether this is a network problem, or if e.g. libavformat
is probing too much data when opening a stream. Maybe the cache-speed
property is helpful here.
For now, do not remove the associated code, but just silence the
warning.
Fixes#3019.
It's pretty "unfriendly" and causes too many issues. (Probably. At least
they're more obvious to a user than e.g. broken frame timing.)
Potentially we could apply heuristics like applying this only on
fullscreen, but let's not. It's up to the user to configure this to
get best results.
Fixes#2997.
The past behavior was a bit weird, especially when zooming out. There
was no simple way to zoom in or out in consistent increments using
keybindings alone.
The new behavior preserves most of the old behavior's semantics but
scales out to infinity better. It coincidentally also makes it
really easy to get clean power of 2 ratios (e.g. 2x, 4x, 8x and their
inverses).
Fixes#3004.
I got a report that the build on a recent aarch64 Linux kernel failed.
DVB support was detected, but errored on compilation:
In file included from ../stream/stream_dvb.c:57:0:
../stream/dvbin.h:72:5: error: unknown type name 'fe_bandwidth_t'
fe_bandwidth_t bw;
Make the test stricter, which should take care of this. (I couldn't find
out what exactly triggered the failure, nor could I attempt to reproduce
it.)
The change in stream/dvbin.h is to make sure that this isn't caused by
incorrect header inclusion. It now includes the same files as the
configure test.
There is an obscure feature which requires essentially reordering PTS
from different packets.
Unfortunately, libavcodec introduced a ridiculously shitty API for
this, which works very much unlike the audio/video API. Instead of
simply passing through the PTS, it wants to fuck with it for no reason,
and even worse, fucks with other fields and changes their semantivcs
(??????). This affects AVSubtitle.end_display_time. This probably will
cause issues for us, and I have no desire to find out whether it will.
Since only PGS requires this, and it happens not to use
end_display_time, do it for PGS only.
Fixes#3016.
Basically, this information is useless, because some muxers (hurr
libavformat) write bogus information anyway. This means if we e.g. see
PGS packets in mkv with duration explicitly set to 0, we must not trust
that value anyway. (The FFmpeg API problem is leaking into files, how
nice.)
This makes the black point closer (chromatically) to the white point, by
ensuring channels keep their consistent brightness ratios as they go
down to zero.
I also raised the 3DLUT version as this changes semantics and is a
separate commit from the previous one.
This commit refactors the 3DLUT loading mechanism to build the 3DLUT
against the original source characteristics of the file. This allows us,
among other things, to use a real BT.1886 profile for the source. This
also allows us to actually use perceptual mappings. Finally, this
reduces errors on standard gamut displays (where the previous 3DLUT
target of BT.2020 was unreasonably wide).
This also improves the overall accuracy of the 3DLUT due to eliminating
rounding errors where possible, and allows for more accurate use of
LUT-based ICC profiles.
The current code is somewhat more ugly than necessary, because the idea
was to implement this commit in a working state first, and then maybe
refactor the profile loading mechanism in a later commit.
Fixes#2815.
FFmpeg partially merged the API change. It added the AVCodecParameters
definition, but not the AVCodecContext.codecpar field. The new code
compiles only with the API fully merged, so adjust the check.
Encoding mode uses deprecated API. See previous commit. Encoding mode
will stop working/compiling at some point in the future, so unless
someone fixes the encoding code, it will stay disabled by default.
(Note that the deprecations are not merged in FFmpeg yet, but they will
soon. They've been deprecated in Libav for a while now.)
AVFormatContext.codec is deprecated now, and you're supposed to use
AVFormatContext.codecpar instead.
Handle this for all of the normal playback code.
Encoding mode isn't touched.
This was changed in 2014, so I suppose users will usually have a FFmpeg
release which includes the corresponding upstream change. If not, well
too bad for those MicroDVD-obsessed users.
Also don't try to retrieve the default framerate as exported by the
demuxer, and instead hardcode it and trust it won't ever change. this
avoids that we have to deal with a larger mess in the codecpar commit.
I don't trust it one bit, and it's a bother with the codecpar change.
If it turns out to be important for some file formats, it could be
added back (or FFmpeg fixed).
This commit adds the d3d11va-copy hwdec mode using the ffmpeg d3d11va
api. Functions in common with dxva2 are handled in a separate decode/d3d.c
file. A future commit will rewrite decode/dxva2.c to share this code.
Commit 57506b27 accidentally broke this. The status (including the
usually always active demuxer cache) should be shown only if the stream
cache is actually enabled.
This reverts commit 503c6f7fd6.
There are situations where some decoders (MF apparently) always require
a timestamp. Also, this makes bitrate estimation more granular than
necessary. It seems it's better to try to detect fiels with broken
default durations explicitly instead. Or maybe something should be
added to smooth audio timestamps after filters.
This also draws it after color management etc. In a nutshell, this
change makes the transparency checkerboard independent of upscaling,
panning, cropping etc. It will always be the same apparent size and
position (relative to the window).
It will also be independent of the video colorspace and such things.
(Note: This might cause white imbalance issues if playing a file with a
white point that does not match the display, in absolute colorimetric
mode. But that's uncommon, especially in conjunction with transparent
image files, so it's not a primary concern here)
Until now, we've let the windowing backend decide. But since they
usually require premultiplied alpha, and premultiplied alpha is easier
to handle, hardcode it.
The recent changes fixed rotation handling, but reversed the rotation
direction. The direction is expected to be counter-clockwise, because
demuxers export video rotation metadata as such.
Don't assume EOF if we didn't try to read anything in the first place.
Fixes regressions in particular with low cache sizes, which triggered
the other code paths more often.
Instead of having a separate for each, which also requires separate
additional caching in the demuxer. (The demuxer adds an indirection,
since STREAM_CTRLs are not thread-safe.)
Since this includes the cache speed, this should fix#3003.
This would get stuck in reconfiguring the filter chain forever, because
params was mutated ("params.rotate = 0;"). This was used as input for
vf_reconfig(), but the filter chain input must always be equivalent to
the decoder output, or filter chain reconfiguration will be triggered.
The line of code to reset the rotation is from a time when this used to
work differently.
Also remove the unnecessary try_filter() parameter.
This has been completely broken since commit 93546f0c. But even before,
rotation handling did not make too much sense. In particular, it rotated
the contents of the cropped image, instead of adjusting the crop
rectangle as well. The result was that things like panscan or zooming
did not behave as expected with rotation applied.
The same is true for vertical flipping. Flipping is triggered by
negative image stride. OpenGL does not support flipping the image on
upload, so it's done as part of the rendering. It can be triggered with
--vf=flip, but other filters and even decoders could setup negative
stride to flip the image.
Fix these issues by applying transforms to texture coordinates properly,
and by making rotation and flipping part of these transforms.
This still doesn't work properly for separated scaling. The issue is
that we'd have to adjust how the passes are done. For now, pick a very
stupid solution by rotating the image to a FBO, and then scaling from
that. This has the avantage that the scale logic doesn't have to be
complicated for such a rare case. It could be improved later.
Prescaling is apparently still broken. I don't know if chroma
positioning works properly either. None of this should affect the case
with no rotation.
gl_transform_vec() assumed column-major, while everything else seemed to
assumed row-major memory organization for gl_transform.m. Also,
gl_transform_trans() seems to contain additional confusion.
This didn't matter until now, as everything has been orthogonal, this
the swapped matrix entries were always 0.
If the texture count is lower than 4, entries in va.textcoord[] will
remain uninitialized. While this is unlikely to be a problem (since
these values are unused on the shader side too), it's not nice and might
explain some things which have shown up in valgrind.
Fix by always initializing the whole thing.
E.g. "mouse 100 100 1 double" did not actually process the double-click,
because double-click emulation is on by default. So the user would have
to send two successive clicks instead. This is probably not expected, so
disable this weird logic for artificial input.
Fixes#2899.
In particular, this prevents subtitle packets from building up in the
subtitle queue if e.g. --vo=null is used. In this situation,
sub_get_bitmaps() is never called, and thus the segment never switched.
This also seems to help with flickering at segment switch boundaries (if
subs are supposed to be visible at the transition points).
In theory, this could trigger a switch too early, but the way VO and
subtitle renderer interact wrt. timing is a bit iffy anyway.
SEEK_HR is interpreted by demux_mkv.c, and enables subtitle preroll by
prefetching additional subtitle pakcets which might overlap with the
seek destination. This should make the case work when segment boundaries
fall into the middle of subtitle events.
This still usually leaves a flicker of at least 1 frame on start,
because dec_sub.c does not ensure that enough subtitles are read before
rendering after a segment switch. (Probably a WONTFIX.)
The default security descriptor for named pipes in Windows allows the
pipe to be opened for read access by the Everyone group and Anonymous
account, as well as low-integrity processes (like web browser renderer
processes.) This does not allow commands to be ran, but it does allow
events to be received.
I don't think any sensitive data is exposed by events, but that may not
always be the case and Lua plugins might change this, since they can
broadcast their own events with script-message. To be safe, this commit
sets a custom security descriptor on the named pipe which only allows
access from processes running under the same user account with an
integrity level greater than or equal to the one used by mpv.
Instead of reallocating almost all of the shader string several times
per pass, build it into a fixed buffer that will be reallocated as
needed.
While this still uses a linear search and full comparison of the shader
text, this will compare the shader's string length first before doing a
full comparison as a nice side effect. (That's also why the fragment
shader is compared first - it's more likely to be different for
different cache entries than the vertex shader stub.)
The mp_set_av_packet()/mp_pts_from_av() functions check whether the
timebase is set at all (i.e. AVRational.num!=0), so there's no need to
fiddle with pointers.
Until now, bstr_xappend_vasprintf() called vsnprintf() always twice:
once to determine how much output the call would produce, and a second
time to actually output the data to the (possibly resized) target
memory.
Change this so that it tries to output to the already allocated memory
first, and repeat the call only if allocation is required.
This is especially helpful, as bstr_xappend_vasprintf() is designed to
avoid reallocation when building strings. Usually, the second
vsnprintf() will happen only at the beginning, when the buffer hasn't
been extended to his largest needed size yet.
Not sure if there is a need to optimize this; but see the next commit.
JSON IPC works on Windows now, and although the transports for each
plaform have similar characteristics, they unfortunately have different
names (Unix domain sockets on Linux/Unix vs. named pipes on Windows.)
Hopefully this change better reflects the purpose of the option too,
since with --input-ipc-server, mpv acts as an IPC server that can
service many simultaneous clients (as opposed to --input-file, which can
only do one-to-one IPC.)
This implements the JSON IPC protocol with named pipes, which are
probably the closest Windows equivalent to Unix domain sockets in terms
of functionality. Like with Unix sockets, this will allow mpv to listen
for IPC connections and handle multiple IPC clients at once. A few cross
platform libraries and frameworks (Qt, node.js) use named pipes for IPC
on Windows and Unix sockets on Linux and Unix, so hopefully this will
ease the creation of portable JSON IPC clients.
Unlike the Unix implementation, this doesn't share code with
--input-file, meaning --input-file on Windows won't understand JSON
commands (yet.) Sharing code and removing the separate implementation in
pipe-win32.c is definitely a possible future improvement.
Glitches when resizing are still possible, but are reduced. Other VOs
could support this too, but don't need to do so.
(Totally avoiding glitches would be much more effort, and probably not
worth the trouble. How about you just watch the video the player is
playing, instead of spending your time resizing the window.)
Should reflect I/O speed.
This could go into the terminal status line. But I'm not sure how to put
it there, since it already uses too much space, so it's not there yet.
The old algorithm produced results which were not uniformly distributed,
i.e. some particular shuffles were preferred over others.
The new algorithm is an implementation of the Fisher-Yates shuffle which
is guaranteed to shuffle uniformly given a sufficiently uniform rand()
and ignoring potential floating-point errors.
Signed-off-by: wm4 <wm4@nowhere>
Until now, we have tried to create a GL 3.0 context. The main reason for
this is that many Mesa-based drivers did not support anything better.
But some drivers (Mesa AMD) will not report a higher OpenGL version,
because their compatibility mode is restricted. While later GL features
are reported as extensions just fine, there doesn't seem to be a way to
determine or enable higher GLSL versions.
Add some more shitty hacks to try to deal with this messed up situation,
and try to probe each interesting GL version separately (starting with
3.3, then 3.2 etc.). Other backends might suffer from similar problems,
but these will have to deal with it on their own.
Probably fixes#2938, or maybe not.
Prevents an infinite loop of configure event and set_fullscreen
request on Weston and other compositors respecting the protocol.
Fixes#2817
This reverts commit eb6b2b6e50.
This colorspace has been historically used as a calibration target for
most digital projectors and sees some involvement in the UltraHD
standards, so it's a useful addition to mpv.
This changes behavior somewhat. The old behavior can be restored by
running "mp.use_suspend=true". It was originally introduced for the OSC,
but I can't reproduce whatever misbehavior I was seeing.
(See mp.suspend()/resume() for explanations what the suspend mechanism
does.)
converted_imgfmt will be used by the renderer logic to build an
appropriate shader chain. It doesn't influence the format of any
textures. Thus it doesn't matter whether the hw video surface is mapped
as RGB or RGBA. What matters is if the video actually contains alpha or
not. Since virtually all hardware decoder do not support alpha in any
way, this can be hardcoded as "no alpha".
This avoids unnecessary GPU work.
This also gets rid of the kind of hard to read texture swizzle setup and
turns it into something dumber.
Assumes that we don't create any FBOs with 2 channel formats. (Only the
video source textures are handled by this commit.)
This is particularly useful for opus which allows only a fairly restrictive set
of samplerates. If the codec doesn't provide a list of samplerates, just
continue to try the requsted one and hope for the best.
fixes#2957
This function chooses the best match to a given samplerate from a provided
list. This can be used, for example, by the ao to decide what samplerate to use
for output.
* Use the update-core command
* Add --check-c-compiler=gcc to be safe
* Add warning about potential pitfalls of adding C:\msys2\mingw64\bin to %PATH%
* Recommend winpty
* Add note about ANGLE
Previously, gl->DXOpenDeviceNV was called twice using dxva2 with dxinterop. AMD
drivers refused to allow this. With this commit, context_dxinterop sets its own
implementation of MPGetNativeDisplay, which can return either a
IDirect3DDevice9Ex or a dxinterop_device_HANDLE depending on the "name" request
string. hwdec_dxva2gldx then requests both of these avoiding the need to call
gl->DXOpenDeviceNV a second time.
Drag&drop mechanisms typically support multiple types for the drop data.
Move most of the logic which types are accepted and preferred to
event.c, where the data is also interpreted.
(Maybe sorting the types by assigning scores is over-engineered, since
they're already sorted by preference, but it's actually not much more
code.)
Not very interesting/meaningful yet, but preparation for the next
commit.
Reduces VO access and makes the code more self-contained. (One day the
windowing backend code should not access the VO anymore. We're just not
quite there yet.)
Instead of displaying it only on playback start (or after switching
tracks), always display it even after a seek.
This helps with --lavfi-complex. You can now overlay e.g. audio
visualizations over cover art, and it won't break after a seek.
The downside is that this might make seeks with huge cover art slower.
There is also a glitch on seeking: since cover art pictures always
have timestamp 0, the playback time will be 0 for a moment after seek,
and then revert to audio PTS (as video is considered EOF). This is also
due to how lavfi's overlay filter behaves. (I'm not sure how to tell
lavfi that it's just a single frame.)
Like dxinterop, this uses StretchRect or RGB conversion. This is unavoidable as
long as we use the dxva2 API, as there is no way to access the raw hardware
decoded Direct3D9 surfaces.
This is simpler, because it doesn't have to wait from both threads for
synchronization.
Apart from being simpler/cleaner, this serves vague plans to stop/start
the demuxer thread itself automatically on demand (for the purpose of
reducing unneeded resource usage).
This pause stuff is bothersome and is needed only for a few corner-
cases. This commit removes it from the demuxer public API and replaces
it with a demux_run_on_thread() function and refactors the code which
needed demux_pause(). The next commit will change the implementation.
Changing the byte stream position without cooperation of the demuxer
seems a bit insane, and is certainly useless. A user should do factor
seeks instead. For formats like ts, this will actually translate to byte
seeks, while treating the rest of the playback chain a bit more
gracefully. With this argument, remove write access to this property.
If someone really complains, proper byte seeks could be added as seek
mode (although I'm going to need a convincing argument for this).
Read access changes too, but in a more subtle way.
No need to have them everywhere. The only exception/annoyance is
MAX_OSD_PARTS, which is now basically duplicated (and at runtime
initialization is checked with an assert()).
Until now, there was only 1 global ASS overlay that could be set by all
scripts. This was often perceived as bug when multiple scripts tried to
set their own ASS overlay.
This was kind of hard to solve because the script could set its own ASS
PlayResX/Y, which makes it impossible to share a single ASS_Renderer for
multiple scripts. The OSC unfortunately makes use of this feature (and
unfortunately can't be fixed because it's a POS), so we're stuck with
this complication.
Implement the worst-case solution and fix this by creating separate ASS
track and renderer objects for each script that wants to set an ASS
overlay.
The z-order is decided by the order the scripts set their text first.
This is essentially random, unless you do it at script init, and you
pass scripts in a specific order. Script initialization is currently
serialized (as a feature), so the first loaded script gets lowest
Z-order.
The Lua script API interestingly remains the same. (And also will remain
undocumented, unsupported, and potentially volatile.)
Instead of passing an explicit cache to the function, the res parameter
is used. Also, instead of replacing its contents, sub bitmaps are now
appended to it (all assuming the format doesn't actually change).
This is preparation for the following commits.
The default of 1.0 was basically making half the algorithm do nothing,
since it turned off all diagonal contributions. The upstream default is
0.6, and this produces a more reasonable image.
The values were changed to reflect an upstream change in the source for
the super-xBR implementation.
The anti-ringing code was basically not working at all, the new
algorithm _significantly_ improves the result (reduces ringing).
This is a fresh implementation from scratch that carries with it
significantly less baggage and verbosity from the previous (ported)
version.
The actual values for the masks and such were copied from the
current code. Behavior and performance should be unaffected.
An important difference between the old code and the new code is that
the new code always explicitly samples from the first component, rather
than being able to process multiple planes at once.
Since prescale-luma only affects luma, I deemed this unnecessary. May
change in the future, if prescale-chroma ever gets implemented. But
prescaling multiple planes would be slow to do this way. (Better would
be to generalize it to differently-sized vectors)
Do not scale OSD mouse input to the ASS OSD script resolution. The
original idea of this mechanism was that the user doesn't have to care
about the actual resolution of anything, and can just use the OSD
resolution consistently. But this made things worse.
Remove the implicit scaling, and always use the screen resolution.
(Except with --vo=xv, where additional scaling is forced upon
everything.)
Drop get_osd_resolution(). There is no replacement. Rename
get_screen_size() and get_screen_margins() to use "osd" instead of
"screen". For anything but --vo=xv these are equivalent, but with
--vo=xv the OSD resolution has additional implicit scaling.
Add code to osc.lua which emulates the old behavior.
Note that none of the changed functions were public API, so implicit
breakage of scripts which used it is just going to happen.
Deselecting cover art and then reselecting it did not work. The second
time the cover art picture is not displayed again. (This seems to break
every other month...)
The reason is commit 6640b22a. It mutates the input packet. And it is
correct that we don't own d_video->header->attached_picture at this
point. Fix it by creating a new packet reference.
Subtitles can be preloaded, which means they're fully read and copied
into ASS_Track. This in turn is mainly for the sake of being able to do
subtitle seeking (when it comes down to it, subtitle seeking is the
cause for most trouble here).
Commit a714f8e92 broke preloaded subtitles which have events with
unknown duration, such as some MicroDVD samples. The event list gets
cleared on every seek, so the property of being preloaded obviously gets
lost.
Fix this by moving most of the preloading logic to dec_sub.c. If the
subtitle list gets cleared, they are not considered preloaded anymore,
and the logic for demuxed subtitles is used.
As another minor thing, preloadeding subtitles did neither disable the
demux stream, nor did it discard packets. Thus you could get queue
overflows in theory (harmless, but annoying). Fix this by explicitly
discarding packets in preloaded mode.
In summary, now the only difference between preloaded and normal
demuxing are:
1. a seek is issued, and all packets are read on start
2. during playback, discard the packets instead of feeding them to the
subtitle decoder
This is still petty annoying. It would be nice if maintaining the
subtitle index (and maybe a subtitle packet cache for instant subtitle
presentation when seeking back) could be maintained in the demuxer
instead. Half of all file formats with interleaved subtitles have
this anyway (mp4, mkv muxed with newer mkvmerge).
Commit 503c6f7f essentially removed timestamps from "laces" (Block sub-
divisions), which means many audio packets will have no timestamp.
There's no reason why bitrate calculation can't just delayed to a point
when the next timestamp is known.
Fixes#2903 (no audio bitrate with mkv files).
Although there is logic to prune subtitles as soon as they get too old
in this mode, this is not done for the _currently_ shown subtitles. Thus
explicitly clearing subtitles on seek is required to avoid duplicate
subtitles in certain cases when seeking.
Instead of hard-coding the logic and planes to skip, factor this out
to a reusible function, and instead add the number of relevant
coordinates to the texture state.
Since prescale now literally only affects the luma plane (and the
filters are all designed for luma-only operation either way), the option
has been renamed and the documentation updated to clarify this.
This is a pretty major rewrite of the internal texture binding
mechanic, which makes it more flexible.
In general, the difference between the old and current approaches is
that now, all texture description is held in a struct img_tex and only
explicitly bound with pass_bind. (Once bound, a texture unit is assumed
to be set in stone and no longer tied to the img_tex)
This approach makes the code inside pass_read_video significantly more
flexible and cuts down on the number of weird special cases and
spaghetti logic.
It also has some improvements, e.g. cutting down greatly on the number
of unnecessary conversion passes inside pass_read_video (which was
previously mostly done to cope with the fact that the alternative would
have resulted in a combinatorial explosion of code complexity).
Some other notable changes (and potential improvements):
- texture expansion is now *always* handled in pass_read_video, and the
colormatrix never does this anymore. (Which means the code could
probably be removed from the colormatrix generation logic, modulo some
other VOs)
- struct fbo_tex now stores both its "physical" and "logical"
(configured) size, which cuts down on the amount of width/height
baggage on some function calls
- vo_opengl can now technically support textures with different bit
depths (e.g. 10 bit luma, 8 bit chroma) - but the APIs it queries
inside img_format.c doesn't export this (nor does ffmpeg support it,
really) so the status quo of using the same tex_mul for all planes is
kept.
- dumb_mode is now only needed because of the indirect_fbo being in the
main rendering pipeline. If we reintroduce p->use_indirect and thread
a transform through the entire program this could be skipped where
unnecessary, allowing for the removal of dumb_mode. But I'm not sure
how to do this in a clean way. (Which is part of why it got introduced
to begin with)
- It would be trivial to resurrect source-shader now (it would just be
one extra 'if' inside pass_read_video).
stream->info can be NULL if it's the cache wrapper. To be fair,
stream->info is considered private API anyway. So don't access it, but
check the URL instead.
Deals with broken mkv subtitle tracks generated by tvheadend. The subs
are srt, but without packet durations.
We need this logic for CCs anyway. CCs in particular will be unaffected
by this change because they are also marked with unknown duration. It
could be that there are actual demuxers outputting CCs - in this case,
we rely on the fact that they don't set a (meaningless) packet duration
(or we'd have to work that around).
Commit 8d4a179c made subtitle decoders pick up fonts strictly from the
same source file (i.e. the same demuxer).
It breaks some fucked up use-case, and 2 people on this earth complained
about the change because of this. Add it back.
This copies all attached fonts on each subtitle init. I considered
converting attachments to use refcounting, but it'd probably be much
more complex.
Since it's slightly harder to get a list of active demuxers with
duplicate removed, the prev_demuxer variable serves as a hack to achieve
almost the same thing, except in weird corner cases. (In which fonts
could be added twice.)
This reverts commit af66fa8fa5.
The reverted commit caused AVCodecContext.channel_layout to be set,
while requesting stereo downmix will make libavcodec output a stupid
message:
ac3: Channel layout '5.1' with 6 channels does not match specified number of channels 2: ignoring specified channel layout
The same happens with --demuxer=lavf (without this change too).
I'm not quite sure what acrobatics are required to shut up libavcodec,
but for now revert the commit. It was a rather minor, almost cosmetic
issue, which I consider less important than clean CLI terminal output.
Completely pointless abominations that FFmpeg refuses to remove. They
are ancient, long deprecated API which we can't use anymore. They
confused users as well.
Pretend that they don't exist. Due to the way --vd works, they can't
even be forced anymore. The older hack which explicitly rejects these
can be dropped as well.
The goal is reducing log messups (which happen surprisingly often) by
buffering partial lines in mp_log. This is still not 100% reliable, but
better.
The extrabuffers for MSGL_STATUS and MSGL_STATS are not needed anymore,
because a separate mp_log instance can be used if problems really occur.
Also, give up, and replace the snprintf acrobatics with bstr.
mp_log.partial has a quite subtle problem wrt. talloc: talloc parents
can not be used, because there's no lock around the internal talloc
structures associated with mp_log. Thus it has to be freed manually,
even if this happens through a talloc destructor.
We want to add a prefix to the ffmpeg log message, so we called mp_msg
multiple times until now. But logging such partial lines is a race
condition, because there's only one internal mp_msg buffer, and no
external mp_msg locks.
Avoid this by building the message on a stack buffer.
I might make a mp_log-local partial line buffer, but even then av_log()
can be called from multiple threads, while targetting the same mp_log.
(Really, ffmpeg's log API needs to be fixed.)
Until now, a rather large stack buffer was used for this, and also a
static buffer in mp_log_root. The latter was added to buffer partial
lines, and the stack buffer was used only for MSGL_STATUS and MSGL_STATS
(I guess because these are the most likely/severe to clash with partial
line buffering).
Make the buffer in mp_log_root dynamically sized, so we don't get cut
off log lines if the text is excessively large. (The OpenGL extension
list dumped by vo_opengl is such an example.)
Since we still have to support partial line buffering (FFmpeg's log
callbacks leave no other choice), keep the stack buffer. But make it
smaller; there's no way all ~6KB are going to be needed in any
situation.
Was only available via --vd=help and --ad=help (i.e. not at all via
client API). Not bothering with separating audio and video codecs, since
this list isn't all that useful anyway in general. If someone complains,
a type field could be added.
Export a number of container fields, which may or may not be useful in
some scenarios. They are explicitly marked as originating from the
demuxer, in order to make it explicit that they might be unreliable.
I'd actually like to remove all other cases where container information
is exported, but those numerous cases are going to be somewhat hard to
deprecate.
Also, not directly related, export the description of the currently
active decoder. (This has been requested before.)
Use the mp_lavc_set_extradata() function instead of setting up the
extradata manually. This takes care of the corner case when
extradata_len is 0.
This apparently fixes#2888.
Hr-seek was often off by one frame due to rounding issues, which have
been traditionally taken care off by adding a "tolerance". Essentially,
frames very close to the seek target PTS are not dropped, even if they
may strictly are before the seek target.
Commit 0af53353 accidentally removed this by always removing frames even
if they're within the "tolerance". Fix this by "unsharing" the logic and
making sure the segment code is inactive for normal seeks.
Ever since a change in mplayer2 or so, relative seeks were translated to
absolute seeks before sending them to the demuxer in most cases. The
only exception in current mpv is DVD seeking.
Remove the SEEK_ABSOLUTE flag; it's not the implied default. SEEK_FACTOR
is kept, because it's sometimes slightly useful for seeking in things
like transport streams. (And maybe mkv files without duration set?)
DVD seeking is terrible because DVD and libdvdnav are terrible, but
mostly because libdvdnav is terrible. libdvdnav does not expose seeking
with seek tables. (Although I know xbmc/kodi use an undocumented API
that is not declared in the headers by dladdr()ing it - I think the
function is dvdnav_jump_to_sector_by_time().) With the current mpv
policy if not giving a shit about DVD, just revert our half-working seek
hacks and always use dvdnav_time_search(). Relative seeking might get
stuck sometimes; in this case --hr-seek=always is recommended.
Adds always-on mode by internally utilizing hidetimeout as negative and
forbidding the user to set negative values.
This removes script-message to enable/disable the osc, and instead introduces a
combined 'visibility' control with the values never/auto/always.
It's available via script_opts and script_message as 'osc-visibility'.
As message, it also supports a 'cycle' value.
The del key is bound to cycling the visibility modes.
There were few issues:
- When it's disabled and then enabled, it was displaying the osc briefly and
then autohide right away. Don't do that.
- When it's enabled and then disabled, it was not removing the osc from screen
if called while the osc is visible (because tick() is responsible for the hide
but it doesn't render() the empty osc when the osc is disabled).
- Due to delayed/async unbinding of mouse events it was possible to show_osc()
after it got disabled e.g. from mouse_move. Prevent this.
This eliminates some intermittent pops heard in a HRT MicroStreamer DAC
uncorrelated with user interaction. As a bonus, this resolves#1773 which I can
o longer reproduce as of this commit. Leave the 50ms buffer for shared mode
since that seems to be working quite well.
This is also the way exclusive mode is done in the MSDN example code:
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370844%28v=vs.85%29.aspx
This was originally increased in c545c40 to mitigate glitches that subsequent
refactorings have eliminated.
A COM message loop is apparently totally inappropriate for a low latency
thread. It leads to audio glitches because the thread doesn't wake up fast
enough when it should. It also causes mysterious correlations between the vo
and ao thread (i.e., toggling fullscreen delays audio feed events). Instead use
an mp_dispatch_queue to set/get volume/mute/session display name from the audio
thread. This has the added benefit of obviating the need to marshal the
associated interfaces from the audio thread.
Don't wait for WASAPI to send another feed event if we detect an underfull
buffer. It seems that WASAPI doesn't always send extra feed events if
something causes rendering to fall behind. This causes every subsequent playback
buffer to under-run until playback is reset. The fix is simply to do a one-shot
double feed when this happens, which allows rendering to catch up with playback.
This was observed to happen when using MsgWaitForMultipleObjects to wait for the
feed event and toggling fullscreen with vo=opengl:backend=win. This commit
improves the behaviour in that specific case and more generally makes exclusive
mode significantly more robust.
This commit also moves the logic to avoid *over*filling the exclusive mode
buffer into thread_feed right next to the above described underfil logic.
OpenSL ES is used on Android. At the moment only stereo output is
supported. Two options are supported: 'frames-per-buffer' and
'sample-rate'. To get better latency the user of libmpv should pass
values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER)
and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).
_Of course_ the previous commit broke --force-window behavior (like it
does every single time I touch it).
vo_has_frame() gets cleared after a seek, so e.g. stopping playback of a
file and going to the next by keeping the seek key down will enter a
short moment without video at the end of the first file, which will set
the stalled_video variable to true. Prevent it by using the indication
whether the window was properly created (which is probably exactly what
we want here).
This function is also responsible for destroying the window when needed,
and obviously we should never do that while video is active. (This is
the actual bug, although the other change in this commit already hides
the common breakage it caused.)
If a stream is marked as EOF (due to no packets found in reach), then we
need to wakeup the decoder. This is important especially if no packets
are found at the start of the file, so the A/V sync logic actually
starts playback, instead of waiting for packets that will never come.
(It would randomly start playback when running the playback loop due to
arbitrary external events like user input.)
Some oddity that is not needed anymore. The only thing which still
referenced them was avoiding loading external files more than once,
which is now prevented by checking the list of tracks instead.
Why was this done so stupidly, with so many complicated special cases,
before? Declare it once so the shader bits don't have to figure out where
and when to do so themselves.
Commit 943f76e6, which already tried this, was very stupid: it didn't
actually override the samplerate for Opus, but overrode it for all
codecs other than Opus. And even then, it failed to use the overridden
samplerate. (Sigh...)
Fixes relative seeks. Without this, a seek back could skip so much data
that the seek would effectively jump forward. (Or insert silence for
files with video.)
There's the question whether the frontend should do this instead (by
using information from the decoders), but for now this seems more
proper.
demux_mkv.c does this already, sort of.
libavformat doesn't for seeks in .ogg (aka .opus), but might be doing it
for mkv. Seems to be a mess as well.
Fixes correctness_trimming_nobeeps.opus. One nasty thing is that this
mechanism interferes with the container-signalled mechanism with
AV_FRAME_DATA_SKIP_SAMPLES. So apply it only if that is apparently not
present. It's a mess, and it's still broken in FFmpeg CLI, so I'm sure
this will get fucked up later again.
I'm not quite sure what the FFmpeg AV_FRAME_DATA_SKIP_SAMPLES API
demands here. The code so far assumed that skipping can be more than a
frame, but not trimming. Extend it to trimming too.
This is actually already done by dec_audio.c. But if
AV_FRAME_DATA_SKIP_SAMPLES is applied, this happens too late here. The
problem is that this will slice off samples, and make it impossible for
later code to reconstruct the timestamp properly.
Missing timestamps can still happen with some demuxers, e.g. demux_mkv.c
with Opus tracks. (Although libavformat interpolates these itself.)
I think the conclusion is that AV_PKT_DATA_SKIP_SAMPLES is misdesigned
(at least for some formats), and an alternative mechanism using
durations would be better. (Combining it with a proper timebase would
keep sample-accuracy.)
This happens only if the new segment wasn't read yet.
This is not quite proper and a problem with dec_sub.c internals.
Ideally, it'd wait with rendering until a new enough segment has been
read. Normally, the new segment is available immediately, so the end
will be automatically clipped by switching to the right segment in the
exact moment it's supposed to become effective.
Usually shouldn't cause any problems, though.
Doing --hwdec=auto ends up picking dxva2, creating a decoder, and then
sending D3D frames down the video chain, which immediately fails and
falls back to software.
Consider dxva2 only if the VO provides a context. If this fails,
autoprobing will proceed to try dxva2-copy as usual.
Fixes#2844.
This is in preparation for a hypothetical API change in libavcodec,
which would allow the decoder to return multiple video frames before
accepting a new input packet.
In theory, the body of the if() added to vd_lavc.c could be replaced
with this code:
packet->buffer += ret;
packet->len -= ret;
but currently this is not needed, as libavformat already outputs one
frame per packet. Also, using libavcodec this way could lead to a
"deadlock" if the decoder refuses to consume e.g. garbage padding, so
enabling this now would introduce bugs.
(Adding this now for easier testing, and for symmetry with the audio
code.)
There is some strange code which sets the DTS of the packet to PTS (but
only if it's not AVI), which apparently helps with timestamp
determination with some broken files. This code is annoying because it
tries to avoid mutating the packet (which it logically doesn't own).
Move it to where it does and get rid of the packet_copy mess.
Needed for the following commit.
This tries to determine whether packet PTS values are accurate and can
be used for frame dropping during seeking. Move both checks (PTS is
missing; PTs is non-monotonic) to the earliest place where they can be
done.
The WGL_NV_DX_interop spec says that a shared IDirect3DSurface9 must not
be lockable, but off-screen plain surfaces are always lockable and using
them causes Nvidia drivers to crash. Use a rendertarget for the shared
surface instead.
This also changes the name of the DX_interop handle for the rendertarget
to match the name of the DirectX object (rather than the GL one) to
match the convention used in context_dxinterop.c.
This file was rewritten from scratch in 0cef033, so it should be okay.
As mentioned in #730, it's a complete rewrite referencing only MSDN and
POSIX, rather than the original code.
Apple crap (namely hardware decoding interop) forces us to use rectangle
textures for input. But after that we continue with normal textures.
This was not considered for debanding, and the sampler type used for it
can be different depending on the exact render chain. Simply use the
target type of the input texture.
* use mp_HRESULT_to_str/mp_LastError_to_str
* make some messages non-identical
* replace "GL" -> "OpenGL"
* change some MP_FATAL to MP_ERR that don't actually kill the vo
This is useful in particular for GetLastError, unfortunately, it's stil pretty
dumb with regards to WASAPI or D3D specific errors, so keep the
hresult_to_string switch.
Apparently, some drivers require you to allocate all of the decoder d3d surfaces
at once. This commit changes the strategy from allocating surfaces as needed via
mp_image_pool_set_allocator, to allocating all the surfaces in one call to
IDirectXVideoDecoderService_CreateSurface and adding them to the pool with
mp_image_pool_add.
fixes#2822
Provide a way for the user to add mp_images to the pool. This is required for
dxva2, for which using set_allocator is extremely awkward since all the d3d9
surfaces must be allocated in advance and all together.
I mistakenly copied the wrong license text into these files when
I created them. Since I'm the only one to have touched these files,
it should be OK to change them.
This is achieved indirectly by deslecting all streams for the non-
current segment (and if the segment doesn't share the demuxer with the
currently active one).
Restores functionality added with commit 46bcdb70.
This uses a different method to piece segments together. The old
approach basically changes to a new file (with a new start offset) any
time a segment ends. This meant waiting for audio/video end on segment
end, and then changing to the new segment all at once. It had a very
weird impact on the playback core, and some things (like truly gapless
segment transitions, or frame backstepping) just didn't work.
The new approach adds the demux_timeline pseudo-demuxer, which presents
an uniform packet stream from the many segments. This is pretty similar
to how ordered chapters are implemented everywhere else. It also reminds
of the FFmpeg concat pseudo-demuxer.
The "pure" version of this approach doesn't work though. Segments can
actually have different codec configurations (different extradata), and
subtitles are most likely broken too. (Subtitles have multiple corner
cases which break the pure stream-concatenation approach completely.)
To counter this, we do two things:
- Reinit the decoder with each segment. We go as far as allowing
concatenating files with completely different codecs for the sake
of EDL (which also uses the timeline infrastructure). A "lighter"
approach would try to make use of decoder mechanism to update e.g.
the extradata, but that seems fragile.
- Clip decoded data to segment boundaries. This is equivalent to
normal playback core mechanisms like hr-seek, but now the playback
core doesn't need to care about these things.
These two mechanisms are equivalent to what happened in the old
implementation, except they don't happen in the playback core anymore.
In other words, the playback core is completely relieved from timeline
implementation details. (Which honestly is exactly what I'm trying to
do here. I don't think ordered chapter behavior deserves improvement,
even if it's bad - but I want to get it out from the playback core.)
There is code duplication between audio and video decoder common code.
This is awful and could be shareable - but this will happen later.
Note that the audio path has some code to clip audio frames for the
purpose of codec preroll/gapless handling, but it's not shared as
sharing it would cause more pain than it would help.
When playback of a video ends, and the next file has no video at all (no
cover art or anything), then the window must be cleared.
This also resizes the window forcibly, which is by design.
Fixes#2825.
A client API user is allowed to call mpv_opengl_cb_uninit_gl() followed
by mpv_opengl_cb_init_gl(). This crashed; fix it by fixing the lifetime
of ctx->gl.
This is required so that the individual surfaces can pass beyond the dxva2
decoder and be passed to the vo.
This also adds additional data to mp_image->planes[0] for IMGFMT_DXVA2, which is
required for maintaining and releasing the surface even if the decoder code is
uninited.
The IDirectXVideoDecoder itself is encapsulated together with its surface pool
and configuration in a dxva2_decoder structure whose creation and destruction is
managed by talloc.
Especially useful to see what video formats are involved on the various
filter links.
I suspect this function is not available on Libav, so add necessary
ifdeffery preemptively.
Don't allow rounding to let it underflow to 0. 0 width or height is
simply not allowed and could cause problems otherwhere.
Indirectly fixes CID 1350057, which complains about not checking the
resulting output size values before using it in divisions.
It thinks that integer_conv_fbo[index] is implied to be accessed with up
to index=5. Although that is theoretical only, it has a point that this
makes no sense. Use the same constant for the array allocation, to make
it more uniform and robust.
Fixes CID 1350060.
The code is shared with the --vd-lavc-threads option, so using 0 for
auto-detection just works.
But no, this is not useful. Just change it for orthogonality.
Until now (and in mplayer traditionally), avi timestamps were handled
with a timestamp FIFO. AVI timestamps are essentially just strictly
increasing frame numbers and are not reordered like normal timestamps.
Limiting the FIFO is required because frames can be dropped. To make
it worse, frame dropping can't be distinguished from the decoder not
returning output due to increasing the buffering required for B-frames.
("Measuring" the buffering at playback start seems like an interesting
idea, but won't work as the buffering could be increased mid-playback.)
Another problem are skipped frames (packets with data, but which do
not contain a video frame).
Besides dropped and skipped frames, there is the problem that we can't
always know the delay. External decoders like MMAL are not going to
tell us. (And later perhaps others, like direct VideoToolbox usage.)
In general, this works not-well enough that I prefer the solution of
passing through AVI timestamps as DTS. This is slightly incorrect,
because most decoders treat DTS as mpeg-style timestamps, which
already include a b-frame delay, and thus will be shifted by a few
frames. This means there will be a problem with A/V sync in some
situations.
Note that the FFmpeg AVI demuxer shifts timestamps by an additional
amount (which increases after the first seek!?!?), which makes the
situation worse. It works well with VfW-muxed Matroska files, though.
On RPI, the first X timestamps are broken until the MMAL decoder "locks
on".
It would make somewhat sense for libcs which don't implement locales at
all, such as Bionic.
Beyond that, setlocale() is specified that it can return NULL, and we
shouldn't crash if that happens.
Unfortunately I see no better solution.
The refresh seek is skipped if the amount of buffered audio is not
overly huge.
Unfortunately softvol af_volume insertion still can cause this issue,
because it's outside of the normal dynamic filter chain changing code.
Move the video refresh call to reinit_video_filters() to make it more
uniform along with the audio code.
The ctx->redrawing field signals whether flip_page() should block. Do
not block if a black frame (i.e. nothing) is to be rendered.
Also, frame==NULL can never happen.
To be more specific, enable vo_opengl and vo_opengl_cb, if libmpv is
compiled, and the GL headers happen to be in the default search paths.
Although platform specific code can be useful for libmpv (for window
embedding, and even with vo_opengl_cb for certain forms of hardware
decoding), it's not a requirement to use the opengl_cb API.
Enabling vo_opengl is not useful here, but the rest of the build system
doesn't distinguish vo_opengl and vo_opengl_cb, and I see no reason to.
We either need to be on Windows, or have posix_spawn available.
If someone can come up with a system that is POSIX, but does not provide
posix_spawn, we could make it optional.
This was dumb. Could make it burn 100% CPU and not exit at the end.
(Because it would retry as instructed, instead of terminating playback.)
It also needs to consider EOF as waiting for input. lavfi_process() will
decide if it's really EOF, or if further input might come in the future.
Without this, it'd would think that it does not need to wait for input,
i.e. that new input will be available immediately.
(Not so fond of the duplication of subtle logic.)
fd339e3f53 introduced a regression that caused
segfault while uniniting dxva2 decoder (and possibly vdpau too). The problem was
that it freed the avctx earlier, before calling the backend-specific uninit
which referenced it.
Revert some of the changes of that commit, and avoid calling flush by
checking whether the codec is open instead.
(Based on a PR by Kevin Mitchell.)
Signed-off-by: wm4 <wm4@nowhere>
It doesn't provide this function. The code is not really designed to
work without it, so it will probably mess up big time, but at least
make it compile again.
See --lavfi-complex option.
This is still quite rough. There's no support for dynamic configuration
of any kind. There are probably corner cases where playback might freeze
or burn 100% CPU (due to dataflow problems when interaction with
libavfilter).
Future possible plans might include:
- freely switch tracks by providing some sort of default track graph
label
- automatically enabling audio visualization
- automatically mix audio or stack video when multiple tracks are
selected at once (similar to how multiple sub tracks can be selected)
The complex filter support that will be added makes much more complex
use of libavfilter, and I'm not going to bother with adding hacks to
keep libavfilter optional.
track can't be NLUL at this point, so the if is redundant. Remove it and
unindent the block. Also, make the function check whether the track is
selected at all, which makes it safer and idempotent.
This makes it possible to set video size and position using the
--geometry and/or --autofit options. It's also possible to switch
between fullscreen/non-fullscreen playback during runtime.
It can be "dangerous". In particular, the decoder might have failed to
initialize, and is now in a broken state. avcodec_flush_buffers() is not
expected to be called in this state, and could trigger undefined
behavior.
I can't explain this, but it seems to be a similar case to the ALSA HDMI
one. I find it hard to tell because of the slightly different names and
conventions in use in libavcodec, WAVEEXT channel masks, decoders, codec
specifications, HDMI, and platform audio APIs.
The fix is the same as the one for ao_alsa (see commit be49da72). This
should fix at least playing 7.1 sources on OSX with 7.1(rear) selected
in Audio MIDI Setup. The ao_alsa commit mentions XBMC, but I couldn't
find out where it does that or if it also does that for CoreAudio. It's
woth noting that PHT (essentially an old XBMC fork) also exhibited the
incorrect behavior (i.e. side and back speakers were swapped).
Remove flc-frc <-> sl<->sr. This was just plain wrong, and a mistaken
change to make 7.1 work properly on CoreAudio with 7.1(rear) layout.
Also see the following commit.
Add br-br <-> sl<->sr, because we decided that it makes sense.
Note that this "fudging" is applied only if the channel pairs are
replaced, i.e. they would get dropped and be replaced with silence. This
is done to compensate for libswresample's default rematrixing (which
takes care of some more common cases).
Avoids "problems". In particular, it makes MMAL output a NOPTS timestamp
if the input timestamp was NOPTS.
Don't do it for other decoders. Ideally, we will at some point in the
future switch to integer fractions for timestamps at least up until the
filter layer. But this would be a larger change, and for now I'd prefer
keeping the not-rounded demuxer timestamps (if we have them).
This is the "unicode" version of it. It appears Firefox uses it now?
I'm not sure if we still need to support the old variant, but hopefully
not.
Fixes#2782.
Will be helpful for the coming filter support. I planned on merging
audio/video decoding, but this will have to wait a bit longer, so only
remove the duplicate status codes.
Now I'm not sure if this is really necessary to explicitly mention. But
then there have been issues and pull requests without being properly
looked at for a week or longer.
Since there can be multiple backends for a single API (vaapi can use GLX
or EGL), not logging the exact backend name is annoying. So add it. At
the same time, there is no need to duplicate the name as used by the
--hwdec options, so replace it with using the numeric hwdec API ID.
Let's fix broken samples with questionable heuristic without real
reasoning. Until this gets fixed properly, this is a good compromise,
though. A proper fix would properly resync audio and video without
brutally resetting the decoders, but on the other hand not doing the
brutal reset would cause issues in other obscure corner cases such
resyncing might cause.
This code is tricky because it has to wakeup the mainloop to make
progressing during syncing audio, but also has to avoid waking it up
when it's not needed. Failure to do so either burns CPU by not ever
going to sleep, or causes apparent "freezes" by going to sleep (and it
will continue if the mainloop is woken up e.g. due to user input).
In this case, simply starting A/V playback with --start=5 and removing
an unrelated wakeup in osd.c can trigger such a "freeze". The unrelated
wakeup did hide this bug, nonetheless it's a bug.
(Can't wait to rewrite this shitty audio resync code. And it's all my
fault.)
Commit b53cb8de added a delay queue for decoded frames. This is supposed
to be used with copy-back decoders like dxva2-copy and vaapi-copy.
Surfaces returned by them can't be referenced after uninitializing the
decoders, so they have to be released before destroying the decoder.
Move the flush_all() call above decoder uninit accordingly. Also move
the destruction of the AVFrame used for decoding (just for being
defensive - normally it doesn't hold any reference).
We just need to provide an entrypoint for it, and move the main init
code to a separate function. This gets rid of the messy video chain full
reinit in command.c, which completely destroyed and recreated the video
state for the purpose of mid-stream hw/sw switching.
Don't give the "software_fallback_decoder" field special meaning. Alwass
set it, and rename it to "decoder". Whether hw decoding is used is
determined by the "hwdec" field already.
These changes don't make too much sense without context, but are
preparation for later. Then the audio_src/video_src fields will be
actually be NULL under circumstances.
Before this commit, reinit_audio_chain() did 2 things: create all the
management data structures and initialize the decoder, and handling lazy
filter/output init (as well as dealing with format changes). For the
second purpose, it could be called multiple times (even though it wasn't
really idempotent). This was pretty weird, so make them separate
functions. The new function is actually idempotent too.
It also turns out the reinit functions don't have to call themselves
recursively for the spdif PCM fallback.
Regression caused by commit 3b95dd47. Also see commit 4c25b000. We can
either use video_next_pts and add "delay", or we just use video_pts. Any
other combination breaks. The reason why the assumption that delay==0 at
this point was wrong exactly because after displaying the first video
frame (usually done before audio resync) a new frame might be "added"
immediately, resulting in a new video_next_pts and "delay", which will
still amount to video_pts.
Fixes#2770. (The reason why display-sync was blamed in this issue is
because enabling display-sync in the options forces a prefetch by 2
instead of 1 frames for seeks/playback restart, which triggers the
issue, even if display-sync is not actually enabled. In this case,
display-sync is never enabled because the frames have a unusually high
frame duration. This is also what exposed the initial desync issue.)
This codes tries to deal with broken PTS timestamps, but since commit
271cabe6 it didn't always overwrite the previous timestamp as it should
have. This mattered only if there were broken timestamps in the video
stream.
Also remove the pointless prev_codec_pts variables, since the decoder
doesn't overwrite these fields anymore.
This also makes it refcounted, i.e. the new AVFrame will reference the
mp_audio buffers, instead of potentially forcing the consumer of the
AVFrame to copy the data.
All the extra code is for handling the >8 channels case, which requires
very messy dealing with the extended_ fields (not our fault).
Correctly avoid a reload if the current device was specified by the user through
--audio-device. Previously, we only recognized if the user had specified
--ao=wasapi:device=.
FFmpeg can generate such files. It's unclear whether they're allowed by
Matroska. mkvinfo shows packet timestamps in both forms (one of them
must be a bug), and at last libavformat's demuxer treats timestamps
as signed.
GLES requires this. Some more common sampler types have default
precisions, but not usampler2D. Newer ANGLE builds verify this more
strictly than older builds, so this wasn't caught before.
Fixes#2761.
GLES does not support high bit depth fixed point textures for unknown
reasons, so direct 10 bit input is not possible. But we can still use
integer textures, which are supported by GLES 3.0. These store integer
data just like the standard fixed point textures, except they are not
normalized on sampling. They also don't support bilinear filtering, and
require a special sampler ("usampler2D").
While these texture formats enable us to shuffle the data to the GPU,
they're rather impractical with the requirements mentioned above and our
current architecture. One problem is that most code assumes it can
always use bilinear scaling (even if bilinear is never used when using
appropriate scale/cscale options). Another is that we don't have any
concept of running a function on a texture in an uniform way.
So for now, run a simple conversion step through a FBO. The FBO will use
the rgba16f format normally, which gives enough bits for 10 bit, and
will at least gracefully degrade with higher depth input.
This is bound to be much slower than a more "direct" method, but at
least it works and is simple to implement.
The odd change of function call order in init_video() is to properly
disable "dumb mode" (no FBO use) if these texture formats are in use.
This was never reset - absolutely can't be right. If the renderer
somehow switches back to another codepath, it certainly has to be reset.
Maybe this was hard to hit, as the normalization is going to be
idempotent in simpler cases (like rendering RGBA input).
Also get rid of the "merged" variable.
This seems generally easier when using libmpv (and was already requested
and implemented before: see commit 327a779a; it was reverted some time
later).
With the weird internal logic we have to deal with, in particular the
--softvol=no case (using system volume), and using the audio API's mixer
(--softvol=auto on some systems), we still can't avoid all glitches and
corner cases that complicate this issue so much. The API user is either
recommended to use --softvol=yes or auto, or to watch the new
mixer-active property, and assume the volume/mute properties have
significant values if the mixer is active.
Remaining glitches:
- changing the volume/mute properties has no effect if no internal mixer
is used (--softvol=no) and the mixer is not active; the actual mixer
controls do not change, only the property values
- --volume/--mute do not have an effect on the volume/mute properties
before mixer initialization (the options strictly are only applied
during mixer init)
- volume-max is 100 while the mixer is not active
Commit b53cb8de increased this by the number of additionally delayed
surfaces. But since this is only enabled in copy-back mode (which is
what process_image is about), the other additional surfaces accounted
for the direct rendering case can be ignored.
Often requested. The main argument, that prominent scalers like sharpen
change the image even if no scaling happens, disappeared anyway.
("sharpen", unsharp masking, is neither prominent nor a scaler anymore.
This is an artifact from MPlayer, which fuses unsharp masking with
bilinear scaling in order to make it single-pass, or so.)
Another fix for the crazy and insane nvidia preemption behavior.
This time, the situation is that we are using vo_opengl with vdpau
interop, and that vdpau got preempted in the background while mpv was
sitting idly. This can be e.g. reproduced by using:
--force-window=immediate --idle --hwdec=vdpau
and switching VTs. Then after switching back, load a video file.
This will not let mp_vdpau_handle_preemption() perform preemption
recovery, simply because it will do so only once vdp_decoder_create()
has been called. There are some other API calls which trigger
preemption, but many don't.
Due to the way the libavcodec API works, vdp_decoder_create() is way too
late. It does so when get_format returns. It notices creating the
decoder fails, and continues calling get_format without the vdpau
format. We could perhaps force it to reinit again (by adding a call to
vdpau.c, that checks for preemption, and sets hwdec_request_reinit), but
this seems too much of a mess.
Solve it by calling API in mp_vdpau_handle_preemption() that empirically
does trigger preemption: output_surface_put_bits_native(). This call is
useless, and in fact should be doing nothing (empty update VdpRect).
There's the slight chance that in theory it will slow down operation,
but in practice it's bound to be harmless. It's the likely cheapest and
simplest API call I've found that can trigger the fallback this way.
(The driver is closed source, so it was up to trial & error.)
Also, when initializing decoding, allow initial preemption recovery,
which is needed to pass the test mention above.
With the format left untouched, this would just try to reinit with a
spdif format again.
We're not clearing the format in reset_audio_state() so the audio chain
can be recreated any time without having to wait for a frame to be
decoded.
Some VOs had support for these - remove them.
Typically, these formats will have only some use in cases where using
RGB software conversion with libswscale is faster than letting the
VO/GPU do it (i.e. almost never). For the sake of testing this case,
keep IMGFMT_RGB565. This is the least messy format, because it has no
padding/alpha bits with unknown semantics.
Note that decoding to these formats still works. We'll let libswscale
repack the data to whatever the VO in use can take.
Notably, the address of the enumerator->count member is passed to
IMMDeviceCollection::GetCount(), which expects a UINT variable, not an int. How
did this ever work?
Previously, if the enumerator found no devices, attempting to get the default
device with IMMDeviceEnumerator::GetDefaultAudioEndpoint would result in the
cryptic (and undocumented) E_PROP_ID_UNSUPPORTED. This way, the user is given a
better indication of what exactly is wrong and isolates any other possible
triggers for this error.
At least DTV_ENUM_DELSYS is not available in older versions.
It's hard to tell when this identifier was introduced, but it appears it
was probably API version 5.5.
Even though the timing logic is correct, it tends to mess with looping
videos and such in unappreciated ways.
It also has to be admitted that most file formats seem not to properly
define the duration of the last video frame (or libavformat does not
export it in a useful way), so whether or not we should use the demuxer
reported framerate for the last frame is questionable. (Still, why would
you essentially just discard the last frame?)
The timing logic is kept, but disabled for video with "normal" FPS
values. In particular, we want to keep it for displaying images, which
implicitly set the frame duration to 1 second by reporting 1 FPS. It's
also good for slide shows with mf://.
Fixes#2745.
Most text subtitles are read completely on loading (libavformat works
this way, and there are good reasons to do it on the higher levels too).
This leads to some messy problems. For example, the subtitle path is the
only one which might read packets during decoder initialization. This is
not neccessary; get rid of it.
This fixes a potential problem of seeking to position 0 for image
subtitles on init, and if the start position is not at the beginning of
the timeline.
It doesn't need to be part of the big context, but is strictly part of
shuffling data from the audio filters to audio output, and thus belongs
into ao_chain.
It also turns out that clearing it in clear_audio_output_buffers() is
completely redundant.
(Of course ao_buffer is an abomination in the first place and shouldn't
exist at all.)
vo_chain_uninit() isn't supposed to care much about the decoder
(although decoders and outputs still go strictly together, so there is
not much of an actual difference now).
Also unset track.d_video correctly.
Remove a stale declaration from dec_video.h as well.
Similar to the video path. dec_audio.c now handles decoding only. It
also looks very similar to dec_video.c, and actually contains some of
the rewritten code from it. (A further goal might be unifying the
decoders, I guess.)
High potential for regressions.
Using the new API is a necessity for multiple-delivery-system
devices, since the old API does not offer a way to switch
the delivery system of the card.
This should in principle also be done for DVB-T / ATSC,
especially since most DVB-T devices also support DVB-C,
but I can not test such an implementation due to lack of hardware
(currently) so it seems better to leave the existing, tested code-path
in place for now.
No need use use all capital letters, and don't warn
if DVB-S2 is supported in addition since we handle that
in DVB-S case already.
Also, print the delivery system number for still unhandled
delivery systems to simplify debugging.
Saves one unnecessary additional ioctl per tuning
by just reusing existing information.
Should also fix the case of multiple supported delivery types
since we now rely on the initial query from the chosen
configuration after channel list parsing
instead of requerying the device.
Most common case would be DVB-C / DVB-T combination cards.
Cards with multiple delivery systems are only supported
starting from DVBv5 API (Kernel 2.6.38).
In this case, we loop over all delivery systems and
just treat them as different cards would be treated:
They all get their own TUNER-type, channel-list parsing etc.
Only request the current screen configuration instead of polling for new
screens, too. We're not interested in detecting any new screens as we're
merely enumerating what is currently connected and configured.
On some hardware (like mine) calling XRRGetScreenResources will stall
X11 for about 10 to 20 seconds. This has annoyed me for a few months
now and almost made me switch to VLC ;)
Signed-off-by: wm4 <wm4@nowhere>
This is probably the 3rd time the user-visible behavior changes. This
time, switch back because not normalizing seems to be the more expected
behavior from users.
Too many problems. Well, actually it's just Linux audio systems which
cause problems, and exclusive audio access on other platforms.
In any case, it seems you have to do some manual configuration if you
want multichannel audio output.
Seems useless.
This only helped in one case: one audio stream in the sample
av_find_best_stream_fails.ts had a AC3 packets which couldn't be
decoded, and for which avcodec_decode_audio4() returned 0 forever. In
this specific case, playback will now not start, and you have to
deselect audio manually.
(If someone complains, the old behavior might be restored, but
differently.)
Also remove the stale "bitrate" field.
While the situation is not really clear for the other rewritten
coreaudio code, it's very clear for the channel mapping code. It was all
written by us. (MPlayer doesn't even have any channel map handling.)
Do this to make the license situation less confusing.
This change should be of no consequence, since LGPL is compatible with
GPL anyway, and making it LGPL-only does not restrict the use with GPL
code.
Additionally, the wording implies that this is allowed, and that we can
just remove the GPL part.
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.
There are probably more files to which this applies, but I'm being
conservative here.
A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).
common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.
codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.
From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).
misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.
screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
Previously used opt_exclusive option to decide which volume control code to run.
The might not always reflect the actual state, for example if passthrough
is used. Admittedly, none of the volume controls will work anyway with
passthrough, but this is the right thing to do.
Slightly helps with timeline stuff, like EDL. There is no need to keep
network (or even just disk I/O) busy for all segments at the same time,
because 1. the data won't be needed any time soon, and 2. will probably
be discarded anyway if the stream is seeked when segment is resumed.
Partially fixes#2692.
Prevents channels from being dropped, e.g. when going 7.1 -> 7.1(wide)
and similar cases. The reasoning here is that channel layouts over HDMI
don't work anyway, and not dropping a channel and playing it on a
slightly "wrong" (but expected) speaker is preferable to playing silence
on these speakers.
Do this to remove issues with ao_coreaudio. Frankly I'm not sure whether
our mapping (between CA and mpv/FFmpeg speakers) is correct, but on the
other hand due to the reasons stated above it's not all that meaningful.
demux_lavf.c leaked the complete subtitle data if it was put through
iconv.
lavc_conv.c leaked AVCodecContext.subtitle_header (set by libavcodec),
which is fixed by using avcodec_free_context(). It also leaked the
subtitle that was decoded last.
Eventually we want the VO be driven by a A->V filter, so a decoder
doesn't even have to exist. Some features definitely require a decoder
though (like reporting the decoder in use, hardware decoding, etc.), so
for each thing which accessed d_video, it has to be redecided if and how
it can access decoder state.
At least the "framedrop" property slightly changes semantics: you can
now always set this property, even if no video is active.
Some untested changes in this commit, but our bio-based distributed
test suite has to take care of this.
This moves some code related to decoding from video.c to dec_video.c,
and also removes some accesses to dec_video.c from the filtering code.
dec_video.ch is starting to make sense, and simply returns video frames
from a demuxer stream. The API exposed is also somewhat intended to be
easily changeable to move decoding to a separate thread, if we ever want
this (due to libavcodec already being threaded, I don't see much of a
reason, but it might still be helpful).
Makes the next commit simpler. It's probably a bad idea to add more
fields to the global state, but on the other hand the client API state
is pretty much per-instance anyway. It also will help with things like
the proposed libmpv custom stream API.
The aspect ratio calculations are cached (mainly so that aspect ratio
related messages are not logged on every frame). The cache is not clared
anymore when video filters are reconfigured, but changing the
video-aspect-ratio property relied on it. Make it explicit.
Fixes#2714.
Since the streams are chosen from the full TS by the player frontend,
one should not expect that the program which is shown matches the chosen
channel which was used for tuning to the frequency.
Also, reformulate slightly to simplify reading.
This is labeled as "name of the service in broadcasting (channel name)"
and exported as a generic tag by avformat-demuxers, notably lavf
when handling mpegts-streams from DVB.
The binding is similar to the tv-binding, just with capital letters.
Switching the dvb-channel-name property compared to dvb-channel
means the channel-name is shown on-screen when switching instead of
"dvb-channel (error)" otherwise,
and switching anyways happens without changing the card.
Channel switching is treated inside the global DVB state
by now. Anyways the last switching direction is not really useful
and of no interest inside the player.
On read, it returns the name of the current DVB program,
on write, it triggers a channel-switch to the program
if it is found in the channel list of the currently active card.
Compared to the dvb-channel property which already exists
and is a pair of integers (card + channel number) this has the limitation
of not switching the card, but is probably of much more common use.
The mutex is used in dvbin_open and dvbin_close only since these are
the only entry / exit points to the stream.
When opening, it is first checked (mutexed) whether the state already exists
and is in use, then a STREAM_ERROR is returned,
since there may be only one stream_dvb active at a time.
State-creation itself is also protected by mutex.
In dvbin_close, the usage-bit is set to false (mutexed) in case
of channel switch.
In case of stream-teardown, the state is destructed
(also protected by mutex).
The state-structure is kept in a static pointer and reused on
recreation of the stream.
To not leak the state and the FDs within upon mpv shutdown,
the state-structure is still destructed gracefully in dvbin_close(),
unless a channel switch has been initiated directly before.
This fixes channel-switching for DVB which was broken since a609877.
The state-struct now contains everything which can be kept after initial initialization.
This includes the channel-lists, configuration, device-fds and also information like
current channel and current card.
The dvb_priv_t is kept containing the mp-options, a pointer to the state and to the logger.
After this restructuring, the state-struct contains all information which can be persisted
across channel switching.
Lots of noise to remove the vfilter/vo fields from dec_video.
From now on, video filtering and output will still be done together,
summarized under struct vo_chain.
There is the question where exactly the vf_chain should go in such a
decoupled architecture. The end goal is being able to place a "complex"
filter between video decoders and output (which will culminate in
natural integration of A->V filters for natural integration of
libavfilter audio visualizations). The vf_chain is still useful for
"final" processing, such as format conversions and deinterlacing. Also,
there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems
ideal, since otherwise there would be no natural way to handle all these
existing options and mechanisms.
There is still some work required to truly decouple decoding.
Instead of handling this on filter chain reinit, do it directly after
the decoder. This makes the code less entangled. In particular, this
gets rid of the really weird "override params" concept in the video
filter code.
The last_format/fixed_formats have some redundance with decoder_output,
but unfortunately the latter has a slightly different use.
UTF-16 subtitles are special in that they are usually read by
libavformat directly, even though they are not in UTF-8. This is
explicitly handled convert_charset() and skips conversion to UTF-8.
There was a bug due to not resetting the file position: if conversion
happens, the actual stream is replaced with a memory stream containing
the converted data, but if conversion is skipped, the original stream
with the wrong file position is kept.
Fix by always opening a memory stream. (We _could_ seek back, but there
is a slight possibility of additional failure due to unseekable
streams.)
Also, don't enter conversion if the subtitle is detected as UTF-8
either.
Fixes#2700.
Basically reimplement it. The old implementation was quite stupid, and
was probably done this way because video filtering and output used to be
way less decoupled. Now we can reimplement it in a very simple way: when
backstepping, seek to current time, but keep the last frame that was
supposed to be discarded when reaching the target time. When the seek
finishes, prepend the saved frame to the video frame queue.
A disadvantage is that the new implementation fails to skip over
timeline boundaries (ordered chapters etc.), but this never worked
properly anyway. It's possible that this will be fixed some time in the
future.
This is mainly a refactor. I'm hoping it will make some things easier
in the future due to cleanly separating codec metadata and stream
metadata.
Also, declare that the "codec" field can not be NULL anymore. demux.c
will set it to "" if it's NULL when added. This gets rid of a corner
case everything had to handle, but which rarely happened.
pyqtgraph's intColor() is less than ideal, especially on white
background. Can't see anything.
Unfortunately the rendering of the legend can't be fixed, because
pyqtgraph is terrible and hardcodes its rendering, including colors.
This slightly changes behavior when seeking with external audio/subtitle
tracks if transport streams and mpeg files are played, as well as
behavior when seeking with such external tracks.
get_main_demux_pts() is evil because it always blocks on the demuxer (if
there isn't already a packet queued). Thus it could lock up the player,
which is a shame because all other possible causes have been removed.
The reduced "precision" when seeking in the ts/mpeg cases (where
SEEK_FACTOR is used, resulting in byte seeks instead of timestamp seeks)
might lead to issues. We should probably drop this heuristic. (It was
introduced because there is no other way to seek in files with PTS
resets with libavformat, but its value is still questionable.)
Note that hresult_to_str() (coming from wasapi_explain_err()) is mostly
wasapi-specific, but since HRESULT error codes are unique, it can be
extended for any other use.
This is another attempt at making files with sparse video frames work
better.
The problem is that you generally can't know whether a jump in video
timestamps is just a (very) long video frame, or a timestamp reset. Due
to the existence of files with sparse video frames (new frame only every
few seconds or longer), every heuristic will be arbitrary (in general,
at least).
But we can use the fact that if video is continuous, audio should also
be continuous. Audio discontinuities can be easily detected, and if that
happens, reset some of the playback state.
The way the playback state is reset is rather radical (resets decoders
as well), but it's just better not to cause too much obscure stuff to
happen here. If the A/V sync code were to be rewritten, it should
probably strictly use PTS values (not this strange time_frame/delay
stuff), which would make it much easier to detect such situations and
to react to them.
mp_parse_escape() is used by the JSON parser in json.c, and JSON allows
escaping "/" (solidus).
Although it makes no sense, apparently Javascript traditionally allowed
that as escape sequence for working around issues with embedding
Javascript in HTML. (Or something like this must have been the history
of this issue.) Since it's valid in Javascript, it had to be valid in
JSON as well, and JSON explicitly specifies it as valid escape.
Fixes#2694.
AVComponentDescriptor.offset was introduced relatively recently. On
older releases, you have to use AVComponentDescriptor.offset_plus1,
which is now deprecated.
Instead of adding ifdeffery, assume AV_PIX_FMT_NV21 is the only format
for which this applies (and will remain the only case), which is
probably true enough.
Should take care of the planned FFmpeg AV_PIX_FMT_P010 addition. (This
will eventually be needed when doing HEVC Main 10 decoding with DXVA2
copyback.)
A format could declare that some or all LSBs in a component are padding
bits by setting a non-0 AVComponentDescriptor.shift value. This means we
would interpret it incorrectly, because until now we always assumed all
regular formats have the padding in the MSBs.
Not a single format that does this actually exists, though. But a NV12
variant will be added later in FFmpeg.
Windows definitely supports Unix-style fd inheritance. This mostly
worked when launched from mpv.exe, though mpv should change the file
mode to O_BINARY. When launched from mpv.com, the wrapper must pass the
list of handles (stored in the undocumented lpReserved2 and cbReserved2
fields) to the mpv process.
Commit 3909e4cd ended up losing the ability to tune the gaussian window,
which this commit trivially reintroduces.
The constant scaling factor (present in the code copied from glumpy)
also goes against filter_kernels.c conventions, which is that f(0.0) = 1
(and the invoking code takes care of normalization), and has been
removed.
The values of this new gaussian function corresponds to different
functions when compared against the old version. To translate the old
values p1 to the new values p2 requires solving 2^(e/p1) = e^(2/p2) or
p2 = p1 * 2/(e * ln(2)) ≈ p1 * 1.0615
In other words, to get the old default in the new function requires
setting scale-param1 to 1.0615. (The new function is *slightly* sharper
by default)
(Though most users should probably not notice the change)
To get a uniform license for this file, relicense the mpv parts to BSD
as well.
But leave the door open for a later change to LGPL. (All non-Glumpy code
was written within mpv, and all mpv authors have agreed to LGPL
relicensing.)
Closes#2688.
This file claims to be based on the "MPlayer VA-API patch", but this is
untrue. Only some glue code was copied from hwdec_vaglx.c, and this glue
code was never in MPlayer or the MPlayer VA-API patch in any form, and
instead part of the mpv-original way we do hardware decoding OpenGL
interop. The EGL interop method didn't exist at the time the MPlayer
VA-API patch was created either.
PT_RELOAD_FILE is a somewhat obscure case when using DVB or when
switching Matroska editions. Both cases were broken, because the
asynchronous playback abort mechanism was still triggered. This
mechanism is used to force the demuxer and stream layers to exit
immediately (instead of blocking on I/O possibly forever), and
is normally disabled on playback start. The reopen path is a bit
strange, and needs to reset it manually.
Pointed out in #2568.
If you do "mpv /bla/", and then branch out into sub-directories using
playlist navigation, and then used quit and watch later, then playing
the same directory did not resume from the previous point. This was
because resuming is based on the path hash, so a path prefix can't be
detected when resuming the parent directory.
Solve this by writing each path prefix when playing directories is
involved. (This includes all parent paths, so interestingly, "mpv /"
would also resume in the above example.)
Something like this was requested multiple times, and I want it too.
This commit replaces code based on AGG, taken from this source file:
http://vector-agg.cvs.sourceforge.net/viewvc/vector-agg/agg-2.5/include/agg_image_filters.h
The intention is that filter_kernels.c can be relicensed to LGPL or BSD.
Because the AGG author died, full replacement is the only way to achieve
it.
This affects only some filter functions. These are exclusively
mathematical functions for computing filter coefficients. (Other parts
in filter_kernel.c were originally written by me, with heavy additions
and refactoring done by other mpv contributors.) While the code is
mostly just well-known mathematical formulas written down in C form,
AGG copyright could perhaps be claimed anyway.
To remove the AGG code, I replaced it with the filter functions from:
https://github.com/glumpy/glumpy/blob/master/glumpy/library/build-spatial-filters.py
These functions conveniently compute exactly the same thing in mpv,
Glumpy, AGG (and about anything that will filter images using the same
mathematical principles).
First I ported the Python code in the file to C. Then I replaced all
functions in filter_kernels.c with this code that could be replaced.
Then I investigated whether the remaining functions were based on AGG
code and took appropriate action:
hanning(), hamming(), quadric(), bicubic(), kaiser(), blackman(),
spline16(), spline36(), gaussian(), sinc() were taken straight from
Glumpy.
For sinc(), re-add the "fabs(x) < 1e-8" check, which was added in commit
586dc557 for unknown reasons.
gaussian() loses its filter parameter for some reason. (Well, who cares,
not my problem.)
The really awkward thing is that the text for hanning() and hamming()
does not change. In theory these functions are now based on Glumpy code,
but it seems like this can be neither proven nor denied. (The same
happened in some other cases with at least a few lines of code.)
sphinx() was added in commit 586dc557, and looks suspiciously like
sinc() as well. Replace the first 3 lines of the body with the ported
function (of which 2 lines do not change; the first uses code only in
mpv, and the second is just "return 1.0;"). The 4th line is only similar
on an abstract level (and that because of the mathematical relation
between these functions). Although the original sinc() was probably used
as template for it, with the other lines replaced, I don't think you
could make the claim that it falls under AGG copyright.
jinc() was added in commit 26baf5b9, but the code for it might be based
on sinc(). Rewrite it based on the "new" sinc(). Some of the same
remarks as with sphinx() apply.
cubic_bc() was ported from Glumpy's Mitchell(). (As far as I'm aware,
with the default parameters it's called "the" Mitchell-Netravali filter,
but in mpv this function is used to generate a whole group of filters.)
spline64() was added in commit a8b67c66, and was probably derived from
spline36(). Re-derive it from the "new" spline36().
triangle() could be considered derived from the original bilinear().
This is this in the original commit:
static double bilinear(kernel *k, double x)
{
return 1.0 - x;
}
This _might_ be based on AGG's image_filter_bilinear:
struct image_filter_bilinear
{
static double radius() { return 1.0; }
static double calc_weight(double x)
{
return 1.0 - x;
}
};
Considering that the "framework" was written by me, and the only part
from AGG taken is "return 1.0 - x;", and this part is trivial and was
later thoroughly replaced, this is probably not under the AGG copyright.
I'm hoping this doesn't introduce regressions. But the main focus is not
being productive anyway, and I didn't rigorously check unintended
changes in functionality.
It existed for XP-compatibility only. There was also a time where
ao_wasapi caused issues, but we're relatively confident that ao_wasapi
works better or at least as good as ao_dsound on Windows Vista and
later.
Normally, PulseAudio accepts any combination of sample format, sample
rate, channel count/map. Sometimes it does not. For example, the channel
rate or channel count have fixed maximum values. We should not fail
fatally in such cases, but attempt to fall back to a working format.
We could just send pass an "unset" format to Pulse, but this is not too
attractive. Pulse could use a format which we do not support, and also
doing so much for an obscure corner case is not reasonable. So just pick
a format that is very likely supported.
This still could fail at runtime (the stream could fail instead of going
to the ready state), but this sounds also too complicated. In
particular, it doesn't look like pulse will tell us the cause of the
stream failure. (Or maybe it does - but I didn't find anything.)
Last but not least, our fallback could be less dumb, and e.g. try to fix
only one of samplerate or channel count first to reduce the loss, but
this is also not particularly worthy the effort.
Fixes#2654.
Apparently, the firmware will ignore pixel_x/pixel_y if the numeric
value of them gets too high (even if they indicate square pixel aspect
ratio). Even worse, the destination rectangle is ignored completely,
and the video frame is simply stretched to the screen. I suspect this
is an overflow or weird sanity check within the firmware.
Work it around by limiting the fields to 16000, which is an arbitrary
but apparently working limit.
There are a lot of incorrectly encoded subtitles with .ass extension
and non-ass subtitles (srt, ssa) with such extension, so we need to
try codepage detection even for .ass.
Signed-off-by: wm4 <wm4@nowhere>
Given 5.1(side), this lets it pick 5.1 from [5.1, 7.1]. Which was
probably the original intention of this replacement stuff. Until now,
the opposite was done in some cases.
Keep the old heuristic if the replacement is not perfect. This would
mean that a subset of the channel layout is an inexact equivalent, but
not all of it.
(My conclusion is that audio output APIs should be designed to simply
take any channel layout, like the PulseAudio API does.)
Unify and clean up listing and selection. Use common enumerator code for both
operations to avoid duplication or inconsistencies.
Maintain, but significatnly simplify manual device selection by id, name or
number. This actually fixes loading by name which didn't really work before
since the "name" displayed by --audio-device=help differed from that used to
match the selection, which used the device "description" instead.
Save the selected deviceID in the private structure for later loading. This will
permit moving the device selection into the main thread in a future commit.
GLSL in GLES 2.0 did not have line continuation in its preprocessor.
This broke shader compilation. It also broke subtitle rendering in
vo_rpi, which reuses some of the OpenGL code.
Line continuation was finally added in GLES 3.0, which is perhaps the
reason why ANGLE accepted it.
When using --start with timeline/ordered chapters, then the
timeline_switch_to_time() function will look at playback_initialized
whether to rselect the currently selected streams on the demuxer level.
So we need to set this field to true at an earlier stage during
initialization, and in particular before the code for --start is called.
Apparently it's only wine where the qpc_position returned by
IAudioClock_GetPosition can be overflowed. So actually do the rescaling
correctly, but throw away the result if it looks unreasonable.
this fixes a regression in 5afa68835a
In particular, the player inserts af_scaletempo by default now when
changing speed, so the suggestion to add this was missleading and
counter-productive.
Slightly change how it is decided when a new packet should be read.
Switch to demux_read_packet_async(), and let the player "wait properly"
until required subtitle packets arrive, instead of blocking everything.
Move distinguishing the cases of passive and active reading into the
demuxer, where it belongs.
Just so I can remove a few lines from dec_sub.c.
This is slightly inelegant, as the whole subtitle file has to be read
into memory, converted at once in memory, and then provided to
libavformat in an awkward way by creating a memory stream instead of
using demuxer->stream. It also won't be possible to force the charset on
subtitles in binary container formats - but this wasn't exposed before,
and we just hope this won't be ever needed. (One motivation was fixing
broken files with non-UTF8 muxed.) It also won't be possible to change
the charset on the fly, but this was not exposed either.
Just simplify by removing parts not needed anymore. This includes
merging dec_sub allocation and initialization (since things making
initialization complicated were removed), or format support queries (it
simply tries to create a decoder, and if that fails, tries the next
one).
So that the video FPs is not required at initialization, and can be set
later.
(As for whether this MicroDVD crap is worth the trouble to handle it
"correctly": MicroDVD files are unfortunately still around, and in at
least one case using the video FPS seemed to help indeed.)
Always preroll by default if the cue (index) information indicates
overlapping subtitles.
Increase the amount of maximum data it will skip to get such subtitles
to 10 seconds. Since the index information can reliably tell whether
reading earlier is needed, the maximum should be rarely actually used,
thus we can set it high. On the other hand, the "old" prerolling
mechanism always has to skip the maximum amount of data; thus the method
using the index gets its own option to control the maximum amount of
data to skip.
(As more and more files With newer mkvtoolnix versions are muxed, and
with this new and hopefully sane default established, these options can
probably be removed in the future.)
Untested, but should be fine. Broken by commit 0a0bb905.
Also fix the include statement in context_rpi.c, which caused another
compilation failure. Also untested. (Because I'm lazy.)
Fixes#2638.
Keeping ASS_Renderers around for a potentially large number of subtitle
tracks could lead to excessive memory usage, especially since the libass
cache is broken (caches even unneeded data), and might consume up to
~500MB of memory for no reason.
This includes the case of switching ordered chapter boundaries. It will
now be recreated on each timeline part switch. This shouldn't be much of
a problem with modern libass. (Older libass versions use fontconfig for
memory fonts, and will be very slow to reinitialize memory fonts.)
Since commit 6d9cb893, subtitle state doesn't survive timeline switches
(ordered chapters etc.). So there is no point in caching the state per
sh_stream anymore (which would be required to deal with multiple
segments). Move the cache to struct track.
(Whether it's worth caching the subtitle state just for the situation
when subtitle tracks get reselected is questionable. But for now, it's
nice to have the subtitles immediately show up when reselecting a
subtitle.)
For files with only 1 chapter, the "cycle" command was ignored. Reenable
it, but don't let it terminate playback of the file.
For the full story, see #2550.
Fixes#2550.
OK, this made the --sub-paths and --audio-file-paths synonyms, which is
not what we wanted. Actually restrict the type of file loaded as well.
Really fixes#2632.
Requested. It works like --sub-paths. This will also load audio files
from a "audio" sub directory in the config file (because the same code
as for subtitles is used, and it also had such a feature).
Fixes#2632.
When crossing timeline boundaries (such as switching to a new segment or
chapter with ordered chapters), clear the internal text subtitle list.
This breaks the sub-seek command, but is otherwise not too harmful.
Fixes Sub-OC-test-final7.mkv. (The internal text subtitle list is
basically a cache to make subtitles show up at the right time when
seeking back.)
I suspect this was caused by 76fcef61. The sample file times subtitles
slightly before the video frame when it should show up. This is to avoid
problems with subtitles showing up a frame later than intended. It also
means that a subtitle which is supposed to show up on the start of a
timeline part boundary actually might first be shown in a different
part. Since we now manipulate the packet timestamps, instead of
manipulating timestamps after the subtitle decoder, this means this
subtitle event would have 2 timestamps, which our code of course does
not handle.
If the two parts come one after another, this would actually work (since
the subtitle would have the same timestamps in the old and new part),
but it breaks if the new part (which follows the old part in the
physical file) is has a completely different start time in the timeline.
Essentially, the trick used to time subtitles correctly is incompatible
with the way we cache subtitles (to make them survive seeks).
The simple solution is just clearing the cached subtitles when crossing
chapter boundaries.
See #2609:
"When eof is reached it would be shown on the OSD and in the console.
Next try seeking to the middle. Seeking to the middle of the file will
only result in the OSD message being updated. Lua seems to fail to
observe the change in the property until the video is unpaused."
Merge blend_src8_alpha and blend_src16_alpha into blend_src_alpha, and
the same for blend_const_alpha.
One thing that changes is that the vertical loop is now shared for both
code paths.
I think this is slightly easier to read, and it's a bit shorter as well.
This removes the need to define IMGFMT_GBRAP, which fixes compilation
with the current Libav release.
This also makes it automatically pick up a GBRP format with the same bit
width. (Unfortunately, it seems libswscale does not support conversion
to AV_PIX_FMT_GBRAP16, so our code falls back to 8 bit, removing
precision for video covered by subtitles in cases this code is used.)
Also, when the source video is e.g. 10 bit YUV, upsample to 16 bit.
Whether this is good or bad, it fixes behavior with alpha. Although I'm
not sure if the alpha range is really correct ([0,2^16-1] vs.
[0,255*256]). Keep in mind that libswscale doesn't even agree with the
way we do it.
This actually treats destination alpha correctly, and gives much better
results than before. I don't know if this is perfectly correct yet,
though. Slight difference with vo_opengl behavior suggests it might not
be.
Note that this does not affect VOs with true alpha support. vo_opengl
does not use this code at all, and does the alpha calculations in OpenGL
instead.
Commit 1a6f3c56 added a fallback for the case when C11 TLS was not
available, but GCC TLS was. But it forget to enable VAAPI EGL interop in
the build system in this case.
Just remove the build system check. Should someone find a compiler that
works on Linux and does not support GCC extensions or C11, it will still
compile and just fail to init at runtime.
Actually fixes#2631 (hopefully).
The demuxer infrastructure was originally single-threaded. To make it
suitable for multithreading (specifically, demuxing and decoding on
separate threads), some sort of tripple-buffering was introduced. There
are separate "struct demuxer" allocations. The demuxer thread sets the
state on d_thread. If anything changes, the state is copied to d_buffer
(the copy is protected by a lock), and the decoder thread is notified.
Then the decoder thread copies the state from d_buffer to d_user (again
while holding a lock). This avoids the need for locking in the
demuxer/decoder code itself (only demux.c needs an internal, "invisible"
lock.)
Remove the streams/num_streams fields from this tripple-buffering
schema. Move them to the internal struct, and protect them with the
internal lock. Use accessors for read access outside of demux.c.
Other than replacing all field accesses with accessors, this separates
allocating and adding sh_streams. This is needed to avoid race
conditions. Before this change, this was awkwardly handled by first
initializing the sh_stream, and then sending a stream change event. Now
the stream is allocated, then initialized, and then declared as
immutable and added (at which point it becomes visible to the decoder
thread immediately).
This change is useful for PR #2626. And eventually, we should probably
get entirely of the tripple buffering, and this makes a nice first step.
The "script-binding" command is used by the Lua scripting wrapper to
register key bindings on the fly. It's also the only way to get fine-
grained information about key events (such as separate key up/down
events). This information is sent via a "key-binding" message when the
state of a key changes.
Extend it to send name of the mapped key itself. Previously, it was
assumed that the user just uses an unique identifier for the binding's
name, so it wasn't needed. With this change, a user can map exactly the
same command to multiple keys, which is useful especially with the next
commit.
Part of #2612.
gcc 4.8 does not support C11 thread local storage. This is a bit
annoying, so add a hack to use the gcc specific __thread extension if
C11 TLS is not available.
(This is used for the extremely silly mpv-internal way hwdec modules
access some platform specific handles. Disabling it simply made
hwdec_vaegl.c always fail initialization.)
Fixes#2631.
Add a "blend-tiles" choice to the "alpha" sub-option. This is pretty
simplistic and uses the GL raster position to derive the tiles. A weird
consequence is that using --vo=opengl and --vo=opengl-hq gives different
scaling behavior (screenspace pixel size vs. source video pixel size
16x16 tiles), but it seems we don't have easy access to the original
texture coordinates. Using the rasterpos is probably simpler.
Make this option the default.
Github will display a link to it when a user wants to create an issue or
pull request.
Also make some minor adjustments to DOCS/contribute.md, which is
developer oriented, and for which I see no reason to merge it with
the new file.
window-scale is now mapped to Alt+0 etc. by default (although these
bindings just use "set", not "cycle-values").
colormatrix can't be cycled anymore (would require using vf_format).
Make sure that subtraction of performance counters is done correctly.
Follow the *exact* instructions for converting performance counter to something
comparable to the QPCposition returned by IAudioClient::GetPosition
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370889%28v=vs.85%29.aspx
Also make sure that subtraction of unsigned integers is stored into a signed
integer to avoid nastiness. Also be more careful about overflow in the
conversion of the device position into number of samples.
Avoid casting mp_time_us() to a double, and use llrint to convert the
double precision delay_us back to integer for ao_read_data.
Finally, actually check the return value of ao_read_data and add a verbose
message if it is not the expected value. Unfortunately,
there is no way to tell WASAPI when this happens since the frame_count in
ReleaseBuffer must match GetBuffer.
This is for the sake of command.c and the "deinterlace" option/property.
Instead of forcing certain "better" defaults when inserting yadif,
change the actual "yadif" defaults.
I pondered not changing vf_yadif, and instead adding a trivial "yadif-
auto" wrapper filter, which would merely have different defaults. But
thinking about it, it doesn't make any sense for "deinterlace" to have
different defaults from vf_yadif, with vf_yadif having the "worse"
defaults. If someone wants the old behavior, the old behavior can be
forced in a backward and forward compatible way by setting the
suboptions.
Fixes#2539 (kind of).
When libuchardet returns an empty string, it can be either ASCII, UTF-8,
or an unknown encoding. Try to distinguish it from the unknown case by
checking for UTF-8. This avoids an annoying message, and avoids
unnecessary processing (we convert invalid UTF-8 sequences to latin1 to
workaround libavcodec's braindead UTF-8 check).
Commit 127da161 was not properly tested either - it did nothing, and
just made it use the video bitstream aspect ratio determined by
libavformat (which isn't always the correct one).
long is 64 bits on x86_64 on Linux, which means the check for the corner
case of computing the depth mask is wrong.
Also, X11 compositors seem to expect premultiplied alpha.
Do not try and set/get master volume in exclusive if there is no
hardware support. This would just uselessly change the master slider,
but have no effect on the actual volume.
Furthermore if getting hardware volume support information fails, then assume
it has none.
It was complicated and not even very intuitive to the user.
If you are controlling the master volume, you just have to be
prepared to deal with the consequences.
Both mpv and ffmpeg have their own internal pthreads wrappers. The mpv
one has been recently enabled by default as well. (It didn't work on XP,
but we dropped XP support.)
libwaio was added due to the complete inability to cancel synchronous
I/O cleanly using the public Windows API in Windows XP. Even calling
TerminateThread on the thread performing I/O was a bad solution, because
the TerminateThread function in XP would leak the thread's stack.
In Vista and up, however, this is no longer a problem. CancelIoEx can
cancel synchronous I/O running on other threads, allowing the thread to
exit cleanly, so replace libwaio usage with native Vista API functions.
It should be noted that this change also removes the hack added in
8a27025 for preventing a deadlock that only seemed to happen in Windows
XP. KB2009703 says that Vista and up are not affected by this, due to a
change in the implementation of GetFileType, so the hack should not be
needed anymore.
This sets the minimum supported Windows version to Windows Vista. The
subsystem version also affects some Windows API functions, including
GetSystemMetrics(SM_CXPADDEDBORDER).
CSIDLs have been deprecated in Windows Vista and are not recommended for
use in new code. They have been replaced with Known Folder IDs, which
are pretty much the same thing, except they use GUIDs.
Converted subtitles use a different method to avoid adding repeated
packets as duplicate subtitle events. The state for this mechanism must
be cleared as well if --sub-clear-on-seek is used.
MPlayer traditionally always used the display aspect ratio, e.g. 16:9,
while FFmpeg uses the sample (aka pixel) aspect ratio.
Both have a bunch of advantages and disadvantages. Actually, it seems
using sample aspect ratio is generally nicer. The main reason for the
change is making mpv closer to how FFmpeg works in order to make life
easier. It's also nice that everything uses integer fractions instead
of floats now (except --video-aspect option/property).
Note that there is at least 1 user-visible change: vf_dsize now does
not set the display size, only the display aspect ratio. This is
because the image_params d_w/d_h fields did not just set the display
aspect, but also the size (except in encoding mode).
Since alpha isn't pulled through the colormatrix (maybe it should?), we
reject alpha formats with odd sizes, such as yuva444p10.
But the awful tex_mul path in vo_opengl does this anyway (at some points
even explicitly), which means there will be a subtle difference in
handling of 16 bit yuv alpha formats. Make it consistent and always
apply the range adjustment to the alpha component. This also means odd
sizes like 10 bit are supported now.
This assumes alpha uses the same "shifted" range as the yuv color
channels for depths larger than 8 bit. I'm not sure whether this is
actually the case.
Now common.c only contains the code for the function loader, while
context.c contains the backend loader/dispatcher.
Not calling it "backend.c", because the central struct is called
MPGLContext.
This is used for dithering, although I'm not aware of anyone who got
higher than 8 bit depth support to work on Linux.
Also put this into egl_helpers.c. Since EGL is pseudo-portable at best I
have no hope that the EGL context creation code in all the backends can
be fully shared. But some self-contained functionality can definitely be
shared.
Store the determined framebuffer depth in struct GL instead of
MPGLContext. This means gl_video_set_output_depth() can be removed, and
also justifies adding new fields describing framebuffer/backend
properties to struct GL instead of having to add more functions just to
shovel the information around.
Keep in mind that mpgl_load_functions() will wipe struct GL, so the
new fields must be set before calling it.
Although the source file is named w32.c, the backend name was "win"
until recently. It was accidentally changed to "w32"; fix it.
Fixes#2608 (the manual is correct).
This actually alows to playback alternating videos with mpv.
Tested with actual file found in wild remuxed to mkv and changed props
with following command:
mkvpropedit /tmp/o.mkv --edit track:1 -s stereo-mode=13
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Apparently, this was replaced by the SD_CTRL_SET_VIDEO_PARAMS set
dimensions. But I can't find out when this happened - possibly, these
fields were never used by sd_lavc.c, and only by the (long removed)
MPlayer dvdsub decoder.
The previous commit turned sd_lavc_conv from a sd_driver to
free-standing functions. Do the rename to reflect this change
separately to avoid confusing git's content tracking. (Or did
git solve this, making separating renames and content changes
unnecessary?)
It was stupid. The only thing that still effectively used it was
sd_lavc_conv - all other "filters" were the subtitle decoder/renderers
for text (sd_ass) and bitmap (sd_lavc) subtitles.
While having a subtitle filter chain was interesting (and actually
worked in almost the same way as the audio/video ones), I didn't
manage to use it in a meaningful way, and I couldn't e.g. factor
secondary features like fixing subtitle timing into filters.
Refactor the shit and drop unneeded things as it goes.
This affects non-ASS text subtitles (those which go through libavcodec's
subtitle converter), which are muxed with video/audio. (Typically srt
subs in mkv.)
The problem is that seeking in the file can send a subtitle packet to
the decoder multiple times. These packets are interlaved with video,
and thus can't be all read when opening the file. Rather, subtitle
packets can essentially be randomly skipped or repeated (by seeking).
Until recently, this was solved by scanning the libass event list for
duplicates. Then our builtin srt-to-ass converter was removed, and
the problem was handled by fully clearing the subtitle list on each
seek.
This resulted in sub-seek not working properly for this type of file.
Since the subtitle list was cleared on seek, it was not possible to
do e.g. sub-seeks to subtitles before the current playback position.
Fix this by not clearing the list, and intead explicitly rejecting
duplicate packets. We use the packet file position was unique ID for
subtitles; this is confirmed working for most file formats (although
it is slightly risky - new demuxers may not necessarily set the file
position to something unique, or at all).
The list of seen packets is sorted, and the lookup uses binary search.
This is to avoid quadratic complexity when subtitles are added in
bulks, such as when opening a text subtitle file.
In some places, the code has to be adjusted to pass through the packet
file position correctly.
This can happen if the file references a track, but does not specify
an INDEX 01 for it. This would cause mpv to just segfault due to
dereferencing the null pointer as a string.
A file causing this was observed in the wild by
ExactAudioCopy v0.99pb4 for a disk that contained a data track at the
end.
With the FFmpeg subtitle decoder used for _all_ non-ASS text subtitle
format, this code is simply unused now.
Ironically, the FFmpeg subtitle decoder does not handle things correctly
in a bunch of cases. Should it turn out they actually matter, they will
have to hack back.
The extend_event one is a candidate, although even though there were
allegedly files which need it, I couldn't get samples from the user who
originally reported such files. As such, extend_event was only confirmed
to handle trailing events with no (endless) duration like with MicroDVD
and LRC, but FFmpeg "fudges" these anyway, so no special handling is
needed.
This code also had logic to handle seeking with muxed srt subtitles,
which made the sub-seek command work. But this has been broken before
this commit already. Currently, seeking with muxed srt subs will clear
all subtitles, as the broken FFmpeg ASS format output by the libavcodec
subtitle converters does not check for duplicates. Since the subtitles
are all cleared, ass_step_sub() can not work properly and sub-seek can
not seek to already seen subtitles.
Slightly simpler, and removes the need to pre-read all subtitle packets.
This still does the subtitle charset conversion on the packet level
(instead converting when parsing the file), so in theory this still
could provide a way to change the charset at runtime. But maybe even
this should be removed, as FFmpeg is somewhat likely to get its own
charset detection and conversion mechanism in the future. (Would have
to keep the subtitle file in memory to allow changing the charset on
the fly, I guess.)
The FFmpeg subtitle converter does the same. There used to be some
deficiencies in FFmpeg's code, but it seems at least some of them have
been fixed. There also used to be the timestamp issue (see previous
commit messages), but this doesn't matter anymore. So no reason to
keep this code - get rid of it.
This can be dropped for the same reasons as in the previous commits. It
removes MicroDVD conversion support on Libav, although MicroDVD files
couldn't be read in the first place ever since demux_subreader.c was
removed.
This restored timestamps when demuxing srt subtitles in Libav, which
was important for avoiding slightly overlapping subtitles. Since the
way this works was changed, there is no real reason to maintain proper
timestamps anymore on this level - this can be dropped without issues.
When a Direct3D 9Ex device fails to reset, it gets put into the lost
state, so set the lost_device flag and don't attempt to present until
the device moves out of that state. Failure to recreate the size-
dependent objects should set lost_device as well, since we shouldn't try
to present in that state.
Also, it looks like I was too eager to remove code that sets priv
members to NULL and I accidentally removed some that was needed.
Direct3D doesn't like 0-sized swapchain dimensions, even when those
dimensions are automatically set. Manually set them to a size that isn't
zero instead.
Why not.
Also, instead of disabling hue/saturation for RGB, just don't apply
them. (They don't make sense for conversion matrixes other than YUV, but
I can't be bothered to keep the fine-grained disabling of UI controls
either.)
A manually added af_volume could lead to muted audio when switching to a
new file. af_volume keeps the last volume set by AF_CONTROL_SET_VOLUME
to return it with AF_CONTROL_GET_VOLUME, but the initial value is 0. So
the mixer volume was forced to 0 when unintializing the filter chain and
reading back the previously set volume.
WGL_NV_DX_interop is widely supported by Nvidia and AMD drivers. It
allows a texture to be shared between Direct3D and WGL, so that
rendering can be done with WGL and presentation can be done with
Direct3D. This should allow us to work around some persistent WGL
issues, such as dropped frames with some driver/OS combos, drivers that
buffer frames to increase performance at the cost of latency, and the
inability to disable exclusive fullscreen mode when using WGL to render
to a fullscreen window.
The addition of a DX_interop backend might also enable some cool
Direct3D-specific enhancements in the future, such as using the
GetPresentStatistics API to get accurate frame presentation timestamps.
Note that due to a driver bug, this backend is currently broken on
Intel. It will appear to work as long as the window is not resized too
often, but after a few changes of size it will be unable to share the
newly created renderbuffer with GL. See:
https://software.intel.com/en-us/forums/graphics-driver-bug-reporting/topic/562051
All of these are supported by FFmpeg now. It was disabled by default
too (with FFmpeg).
If compiled against Libav, mpv will lose the ability to read some
subtitle formats (but the most important ones, srt and ass, still should
work).
This is only for specific Hauppage cards. According to the comments in
who is actively using this feature. Get it out of the way.
Anyone who still wants to use this should complain. Keeping this code
would not cause terribly much additional work, and it could be restored
again. (But not if the request comes months later.)
With default setting, the matrix for fruit dithering requires 12 bits
precision (values from 0/4096 to 4095/4096). But 16-bit float
provides only 10 bits. In addition, when `dither-size-fruit=8` is
set, 16 bits are required from the texture format.
Fix this by attempting to use 16 bit integer texture first. This is
still not precise, but should be better than using a half float.
This has no reason to be there. Put the functionality into another
function instead. While we're at it, also adjust for possible accuracy
issues with high bit depth YUV (matters for rendering subtitles into
screenshots only).
The recent LUT adjustment changes broke interpolation.
The concatenation of the shader stages is a bit messy, and it seems like
sampler_prelude is not a good place to add this macro. Always add the
macro to every shader instead. (While this doesn't seem too elegant,
this isn't too inelegant either, and goes these problems out of the
way.)
The computation of the tex_mul variable was broken in multiple ways.
This variable is used e.g. by debanding for moving expansion of 10 bit
fixed-point input to normalized range to another stage of processing.
One obvious bug was that the rgb555 pixel format was broken. This format
has component_bits=5, but obviously it's already sampled in normalized
range, and does not need expansion. The tex_mul-free code path avoids
this by not using the colormatrix. (The code was originally designed to
work around dealing with the generally complicated pixel formats by only
using the colormatrix in the YUV case.)
Another possible bug was with 10 bit input. It expanded the input by
bringing the [0,2^10) range to [0,1], and then treating the expanded
input as 16 bit input. I didn't bother to check what this actually
computed, but it's somewhat likely it was wrong anyway. Now it uses
mp_get_csp_mul(), and disables expansion when computing the YUV matrix.
It turns out that with accurate lookup we can decrease the
default size of texture now. Do it to compensate the performance
loss introduced by the LUT_POS macro.
Define a macro to correct the coordinate for lookup texture. Cache
the corrected coordinate for 1D filter and use mix() to minimize the
performance impact.
It always was a weird artifact - VOCTRLs are meant _not_ to require
special handling in the code that passes them through (like in vo.c).
Removing it is also interesting to further reduce the dependency of
backends on struct vo. Just get rid of it.
Removing it is somewhat inconvenient, because in many situations the UI
window is created after the first VOCTRL_UPDATE_WINDOW_TITLE. This means
these backends have to store it in a new field in their own context.
Fixes a reported sample, that has a sign interrupted by a few frames
(for which --sub-fix-timing would remove the wanted gap).
The list of tags in has_overrides() is taken from libass. It has a
similar function (which even checks whether the tag are within the { }
delimiters). Unfortunately, this function is not public, so we just have
a simpler one which does roughly the same. It doesn't matter that this
function sometimes returns false positives.
The awkward "preprocess" step of putting the subtitles through single
filters before doing something else was made unnecessary by the recent
changes.
(Fun fact: I originally planned to move these extra things, like fixing
subtitle gaps/overlaps, to filters - but this would suffer from various
complications, and moving them to the renderers seems much simpler.)
I feel like it's better there. Note that there is no reduced
functionality, as bitmaps subs (i.e. not handled by sd_ass.c) were never
fully read on init, and thus never went through sub_read_all_packets().
On the other hand, this might lead to confusion, as --sub-fps etc. will
now also affect muxed subtitles (which makes not much sense).
Instead of messing with the subtitle packet timestamps, do it on output.
We work on the libass event list. If there is an unwanted gap or
overlap, we render the timestamp at another position where there is no
gap or overlap.
This is somewhat more robust, and even works with demuxed subs (to some
degree - depends whether the subs are prefected soon enough).
It's active even for native ASS subs. I wonder if this is a problem with
extended type setting. If it is, the heuristic that tries to avoid
interrupting such cases has to be improved.
While it probably would be ideal to do this after the subtitle decoder,
certain aspects are at least currently handled better in this place.
Image subtitles often use a "signaling" packet to set the end time of
the previous subtitle. As far as the libavcodec API is concerned, such
packets decode to empty AVSubtitles. Discard these after the end time of
the previous subtitle has been set.
Keep track of the per-subtitle end time better. This is for the sake of
improving sub_step/sub_seek. Without this, it would seek to the sub
before the previous sub, if the current sub has ended displaying.
Works roughly the same as the one in sd_ass for text subtitles. While
sub_step is very uninteresting, it comes for free with the support for
sub_seek.
The implementation is taken from ass_step_sub() from libass, with some
modifications
Until now, feeding packets to the decoder in advance was done for text
subtitles only. This was possible because libass buffers all subtitle
data anyway (in ASS_Track). sd_lavc, responsible for bitmap subs, does
not do this. But it can buffer a small number of subtitle frames ahead.
Enable this.
Repurpose the sub_accept_packets_in_advance(). Instead of "can take all
packets" it means "can take 1 packet" now. (The old meaning is still
needed locally in dec_sub.c; keep it there.) It asks the decoder whether
there is place for at least 1 subtitle packet. sd_lavc implements it and
returns true if its internal fixed-size subtitle queue still has a free
slot. (The implementation of this in dec_sub.c isn't entirely clean.
For one, decode_chain() ignores this mechanism, so it's implied that
bitmap subtitles do not use the subtitle filter chain in any advanced
way.)
Also fix 2 bugs in the sd_lavc queue handling. Subtitles must be checked
in reverse, because the first entry will often have endpts==NOPTS, which
would always match. alloc_sub() must cycle the queue buffer, because it
reuses memory allocations (like sub.imgs) by design.
If reinit after a fallback from hardware fails, this field can be NULL.
The check in control() was broken due to a typo (found by Coverity), and
decode() lacked the check entirely.
If the sampling point is placed diagonally, the radius difference
could be as large as sqrt(2.0). And a loosened check with (radius - 1)
would potentially include pixels out of the range.
Fix the check to handle those corner case properly to avoid
unnecessary texture lookup and improve the performance a bit.
reopen_console_handle() was never properly tested because mpv overrides
printf in most source files. Turns out that when there's no console on
startup, the CRT sets the fds of stdout and stderr to -2, so the old
method of using dup2 to manipulate these fds didn't work. As far as I
can tell, the only way to give stdout and stderr valid fds is to use
freopen, so this uses freopen to set them both to the console output.
This also uses dup2 to change STDOUT_FILENO and STDERR_FILENO, so low-
level functions like isatty still work.
Note that this means fileno(stdout) != STDOUT_FILENO. I don't think this
will cause any problems.
This should fix MPV_LEAK_REPORT on the Windows console.
Helps with files that have occasional broken timestamps. For larger
discontinuities, e.g. caused by actual timestamp resets, we still want
to realign audio.
(I guess in general, this should be removed and replaced by a more
general resync-on-desync logic, but not now.)
Sometimes QOpenGLWidget may be redirecting it's output to a framebuffer
object rather than the frontbuffer, in which case the current thread's
context render fbo is different from the widget's. Use the widget's
desired fbo instead.
There are claims that nnedi3.c doesn't constitute its own new
implementation, but is derived from existing HLSL or OpenCL shaders
distributed under the LGPLv3 license.
Until these are resolved, do the "correct" thing and require
--enable-gpl3 to build nnedi.
This makes no sense, because the client is obligated to react to this
event.
This also happens to fix a deadlock with JSON IPC clients sending
"disable_event all", because MPV_EVENT_SHUTDOWN was used to stop the
thread driving the socket connection (fixes#2558).
Approximately reverts commit 3ccac74d. This failed with some avi files,
which do pseudo-VFR by sending packets with empty frames (or repeat
frames, depending on point of view). Specifically, these packets are not
0 bytes, so they don't get skipped by libavformat, as with the usual VFR
avi hack. Instead, the packet contains a VOP with vop_coded=0, so
libavcodec will just return no frame. We could probably distinguish such
skipped frames and delayed frames by explicitly measuring the codec
delay by counting how long it takes to get the very first frame (and
then treat skips as explicit drops), but we may as well simply reinstate
the old code.
To appease to at least one semi-broken case, do not enable this logic on
the RPI, as the FFmpeg MMAL wrapper has arbitrary buffering (and MMAL
itself is asynchronous).
I guess gl_video->global was originally meant to be optional, but now it
crashes in some newer code with vo_opengl_cb, which tries to init it
with this field set to NULL (because normally it's not needed).
Probably fixes#2542.
Requested. Don't overwrite permanent OSD text set with e.g. --osd-msg1.
Instead, append the OSD message to it (on the next line).
Note that with --osd-msg1, seeking will still overwrite the OSD with the
playback status for a while. If you do not want this, use --osd-msg3
--osd-level=3 instead.
Don't use the average FPS if there are likely skipped vsyncs. Note that
we don't use the normal skip detection, as it is unreliable if the real
and assumed display FPS differ too much. The normal skip detection is
still in place as it's more reliable in the case when vsync jitters
much, but the display FPS is relatively exact.
Further improvement over commit 41f2c653.
Commit 2b07d3eb merged progbar and OSD text renderer into one ASS_Track,
but it confused the styles. Specifically, if both progbar and OSD are
visible, the create_ass_track() call made by the progbar code will reset
the style adjusted by the OSD text code.
Change create_ass_track() not to add any styles. Instead let the caller
manage the styles. They are now referenced by name, and lazily added if
they don't exist yet. This is also much cleaner.
Reduces memory usage and startup times. The implementation is a bit
weird, because both OSD parts have conflicting requirements on the used
ASS styles.
At least I hope so.
Deriving the duration from the pts was not really correct. It doesn't
include speed adjustments, and becomes completely wrong of the user e.g.
changes the playback speed by a huge amount. Pass through the accurate
duration value by adding a new vo_frame field.
The value for vsync_offset was not correct either. We don't need the
error for the next frame, but the error for the current one. This wasn't
noticed because it makes no difference in symmetric cases, like 24 fps
on 60 Hz.
I'm still not entirely confident in the correctness of this, but it sure
is an improvement.
Also, remove the MP_STATS() calls - they're not really useful to debug
anything anymore.
This was just converting back and forth between int64_t/microseconds and
double/seconds. Remove this stupidity. The pts/duration fields are still
in microseconds, but they have no meaning in the display-sync case (also
drop printing the pts field from opengl/video.c - it's always 0).
If we switched away from the system FPS, we were remaining in this mode
ssentially forever. There's no reason to do so; switch back if the
estimated FPS gets worse again. Improvement over commit 41f2c653.
Commit 12eb8b2d accidentally disabled framedropping in the audio timing
case. It tried to replace the last_flip field with the prev_vsync one,
which didn't work because prev_sync is reset to 0 if the timing code is
used. Fix it by always setting it properly. This field must (or should)
be reinitialized to something sensible when switching to display sync
timing mode; since prev_vsync is not reset anymore, the check when to
reinitialize this field has to be adjusted as well.
It's a bit weird that update_vsync_timing_after_swap() now does some
minor work for timing mode too, but I guess it's ok, if only to avoid
additional fields and timer calls.
If there were many AO drivers without device selection, this added a
"Default" entry for each AO. These entries were not distinguishable, as
the device list feature is meant not to require to display the "raw"
device name in GUIs.
Disambiguate them by adding the driver name. If the AO is the first, the
name will remain just "Default". (The condition checks "num > 1",
because the very first entry is the dummy for AO autoselection.)
Instead of periodically trying to enable it again. There are two cases
that can happen:
1. A random discontinuity messed everything up,
2. Things are just broken and will desync all the time
Until now, it tried to deal with case 1 - but maybe this is really rare,
and we don't really need to care about it. On the other hand, case 2 is
kind of hard to diagnose if the user doesn't use the terminal.
Seeking will reenable display-sync, so you can fix playback if case 1
happens, but still get predictable behavior in case 2.
If the system-reported display FPS (returned by the VO backends, or
forced with --display-fps) is too imprecise (deviating frame duration by
more than 1%). This works if the display FPS is off by almost 1 (typical
for old/bad/broken OS APIs). Actually it even works if the FPs is
completely wrong.
Is it a good idea? Probably not. It might be better to only output a
warning message. But unless there are reports about it going terribly
wrong, I'll go with this for now.
Actually I'm not content with the detection added in commit 44376d2d. It
triggers too often if vsync is very jittery. It's easy to avoid this: we
add more samples to the detection by reusing the drift computation loop.
If there's a significant step in the drift, we consider it a drop.
This adds basic support for ICC profiles. Per-monitor profiles are
supported. WCS profiles are not supported, but there is an API for
converting WCS profiles to ICC, so they might be supported in future.
I'm just not sure if anyone actually uses them.
Reloading the ICC profile when it's changed in the control panel is also
not supported. This might be possible by using the WCS APIs and watching
the registry for changes, but there is no official API for it, and as
far as I can tell, no other Windows programs can do it.
Of course, only FFmpeg has av_clipd(), while Libav does not. (Nevermind
that it doesn't do much more than the mpv MPCLAMP() macro. Supposedly,
libavutil can provide optimized platform-specific versions for av_clip*,
but of course nothing actually does for av_clipf() or av_clipd().)
Return the estimated/ideal flip time to the timing logic (meaning
vo_get_delay() returns a smoothed out time). In addition to this add
some lame but working drift compensation. (Useful especially if the
display FPS is wrong by a factor such as 1.001.)
Also remove some older leftovers. The vsync_interval_approx and
last_flip fields are redundant or unneeded.
For the vo-delayed-frame-count property.
Slightly less dumb than the previous one (which was removed earlier),
but still pretty dumb. But this also seems to be relatively robust, even
with strong vsync jittering.
This logic was kind of questionable anyway, and --display-sync should
give much better results. (I would even go as far as saying that the
FPS-dependent framedrop code made things worse in some situations. Not
all, though.)
This is simply the average refresh rate. Including "bad" samples is
actually an advantage, because the property exists only for
informational purposes, and will reflect problems such as the driver
skipping a vsync.
Also export the standard deviation of the vsync frame duration
(normalized to the range 0-1) as vsync-jitter property.
libswresample doesn't do it - although it should, but the patch is stuck
in limbo.
Probably reduces problems with artifacts on downmixing in some cases.
The OSD takes up an entire fullscreen dispmanx layer. Although the GPU
should be able to handle it (possibly even without any disadvantages),
it'll still be useful for debugging performance issues.
This is a hack, but unfortunately the DwmGetCompositionTimingInfo
heuristic does not work in all cases (with multiple-monitors on Windows
8.1 and even with a single monitor in Windows 10.) See the comment in
mp_w32_is_in_exclusive_mode() for more details.
It should go without saying that if any better method of doing this
reveals itself, this hack should be dropped.
Remove known useless device entries from the --audio-device list (and
corresponding property). Do this because the list is supposed to be a
high level list of devices the user can select. ALSA does not provide
such a list (in an useable manner), and ao_alsa.c is still in the best
position to improve the situation somewhat.
The ALSA doxygen says:
IOID - input / output identification ("Input" or "Output"), NULL
means both
This bug was blatantly introduced with commit cf94fce4.
It doesn't have any real purpose anymore. Up until now, it was still
implemented by vo_wayland, but since we changed how the frame callbacks
work, even that appears to be pointless.
Originally, the plan was to somehow extend this mechanism to all
backends and to magically fix frame scheduling, but since we can't hope
for proper mechanisms even on wayland, this idea looks way less
interesting.
10 bit HEVC would require DXVA2_ModeHEVC_VLD_Main10, and most a
different surface type (judging by lavfsplitter source code, both
P010 and P016 would work). Since I'm unable to test this stuff,
exclude 10 bit for now.
See #2516.
If a second subtitle is shown, it should be forced to display on top
of the screen. This was working only if --no-ass was passed, because
otherwise the subtitle was rendered normally (i.e. usually on the
bottom).
This is actually all bullshit. There are many factors that can ruin the
video playback experience (and most outside of our or the user's
control).
One thing that makes sense is that this declares incompatibility with
Windows XP (fixes#2473).
The D3D9 backend does not support GLES 3, which makes it pretty useless.
But it still might be a legitimate replacement of vo_direct3d.c on
Windows 7 machines.
Note that we could just use:
eglGetDisplay(EGL_D3D11_ELSE_D3D9_DISPLAY_ANGLE)
But for now I'll leave the old code. Maybe this can exclude use of
software rendering backends (EGL_PLATFORM_ANGLE_DEVICE_TYPE_WARP_ANGLE).
Since I'm not sure, I won't touch it.
Running mpv with default config will now pick up ANGLE by default. Since
some think ANGLE is still not good enough for hq features, extend the
"es" option to reject GLES backends, and add to to the opengl-hq preset.
One consequence is that mpv will by default use libswscale to convert
10 bit video to 8 bit, before it reaches the VO.
I decided that I actually can't stand how vo_opengl unnecessarily puts
the video through 3 shader stages (instead of 1). Thus, what was meant
to be a fallback for weak OpenGL implementations, the dumb-mode, now
becomes default if the user settings allow it.
The code required to check for the settings isn't so wild, so I guess
it's manageable. I still hope that one day, our rendering logic can
generate ideal shader stages for this case too.
Note that in theory, dumb-mode could be reenabled at runtime due to a
color management 3D LUT being set, so a separate dumb_mode field is
required. The dumb-mode option can't just be overwritten.
Unfortunately, color management can still not work, because no GLES
version specified so far support fixed-point 16 bit textures. Maybe
we could use integer textures, but these don't support filtering.
Using float textures would be another possibility.
Polar scalers use 1D textures, because they're slightly faster on some
GPUs than 2D textures. But 2D textures work too, so add support for
them.
Allows using these scalers with ANGLE.
Just like commit f9a2fc59. There are probably some more such cases.
The vec2 constructor calls are probably fine, but don't bother with
confusing inconsistencies.
While desktop GL's glTexImage2D() essentially accepts anything, GLES is
much stricter. The combination of allowed formats/types/internal formats
is exactly specified. The GLES 3.0.4 specification lists them in
table 3.2. (The ANGLE API validation code references this table.)
The table could probably be extended into a general declarative table
about GL formats covering other uses, but this would be a big
non-trivial project, so don't bother and accept a minor degree
of duplication with other tables.
Note that the format and type do (or should) not matter here, because
no image data is transferred to the GPU.
We don't only need float textures for advanced scaling - we also need
them to be filterable with GL_LINEAR. On GLES, this is not supported
until GLES 3.1, but some implementation expose them with extensions.
This makes advanced scaling sort-of work for GLES 3.0 (on ANGLE). It's
still not very advisable, as 8 bits might not be enough to avoid
debanding. (Ironically, the debanding filter can be enabled, and does
not raise any GL errors - but probably doesn't do anything useful.)
Turns out glGetTexLevelParameter, which is missing in ANGLE, is a
GLES3.1 function. Removing it from the list of core GLES3 functions
makes ANGLE work in GLES3 mode.
Apparently, some audio drivers do not support the DTS subtype, but
passthrough works anyway if the AC3 subtype is set. Just retry with
AC3 if the proper format doesn't work. The audio device which
exposed this behavior reported itself as
"M601d-A3/A3R (Intel(R) Display Audio)".
xbmc/kodi even always passes DTS as AC3.
ANGLE is a GLES2 implementation for Windows that uses Direct3D 11 for
rendering, enabling vo_opengl to work on systems with poor OpenGL
drivers and bypassing some of the problems with native GL, such as VSync
in fullscreen mode.
Unfortunately, using GLES2 means that most of vo_opengl's advanced
features will not work, however ANGLE is under rapid development and
GLES3 support is supposed to be coming soon.
This is another regression of the recently added start time probing. If
a seek is executed after opening the file (but before reading any
packets), the first block is discarded instead of indexed. If there are
no other keyframes in the file, seeking will fail completely.
Fix it by seeking to the cluster start if there aren't any index entries
yet. This will read the skipped packet again.
Fixes#2498.
This was used with --no-sub-ass (aka --no-ass). This option (which is
not yet removed) strips all styling from the subtitles, and renders them
as plaintext only. For some reason, it originally seemed convenient to
reuse all the OSD text rendering code (osd_libass.c). While this was
indeed simple, it had a bad influence on the rest of the code. For
example, it had to decide whether to go through the OSD code path, or
the proper subtitle renderer in sd_ass.c.
Kill the OSD subtitle renderer. Reimplement --no-sub-ass and also
"secondary" subtitles in sd_ass.c. fill_plaintext() contains some rather
minor code duplication with osd_libass.c for setting up a dummy
ASS_Event and escaping the stripped text. Since sd_ass.c already has to
handle "normal" text subtitles, and has code for stripping ASS tags,
this remains all relatively simple.
Remove all the unnecessary crap from the rest of the code.
Use the demux_set_ts_offset() added in the previous commit to base each
timeline segment to use timestamps according to its relative position
within the overall timeline. As a consequence we don't need to care
about these timestamps anymore, and everything becomes simpler.
(Another minor but delicious nugget of sanity.)
Most of this is explained in the DOCS additions.
This gives us slightly more sanity, because there is less interaction
between the various parts. The goal is getting rid of the video_offset
entirely.
The simplification extends to the user API. In particular, we don't need
to fix missing parts in the API, such as the lack for a seek command
that seeks relatively to the start time. All these things are now
transparent.
(If someone really wants to know the real timestamps/start time, new
properties would have to be added.)
Something goes wrong somewhere. Don't bother, it's only needed for
compatibility with our absolute baseline (GL 2.1/GLES 2).
On the other hand, we can process nv12 formats just fine.
For the sake of vaapi interop, we want to use EGL, but on the other
hand, but because driver developers are full of shit, vdpau interop will
not work on EGL (even if the driver supports EGL). The latter happens
with both nvidia and AMD Mesa drivers.
Additionally, EGL vaapi interop support can apparently only detected at
runtime by actually using it. While hwdec_vaegl.c already does this, it
would require initializing libva on _every_ system, which will cause
libav to print an unpreventable bullshit message to the terminal.
Try to counter these huge loads of bullshit by adding more fucking
bullshit.
We want the following behavior:
- VO probed, backend probed: only accept non-sw, fail completely
otherwise
- VO forced, backend probed: use the first non-sw, or if none is found,
fall back to the first working sw backend
- VO probed, backend forced: (I don't care about this case)
- VO forced, backend forced: just use that backend
Also, on backend probe failure the vo->probed field was left in its old
state.
This adds support for the progress indicator taskbar extension
that was introduced with Windows 7 and Windows Server 2008 R2.
I don’t like this solution because it keeps its own state and
introduces another VOCTRL, but I couldn’t come up with anything
less messy.
closes#2399
In the display-sync, non-interpolation case, and if the display refresh
rate is higher than the video framerate, we duplicate display frames by
rendering exactly the same screen again. The redrawing is cached with a
FBO to speed up the repeat.
Use glBlitFramebuffer() instead of another shader pass. It should be
faster.
For some reason, post-process was run again on each display refresh.
Stop doing this, which should also be slightly faster. The only
disadvantage is that temporal dithering will be run only once per video
frame, but I can live with this.
One aspect is messy: clearing the background is done at the start on the
target framebuffer, so to avoid clearing twice and duplicating the code,
only copy the part of the framebuffer that contains the rendered video.
(Which also gets slightly messy - needs to compensate for coordinate
system flipping.)
Currently, vo.c will always continue to render the currently queued
frame, which sets last_flip, which in turn confuses vo_get_delay(),
which in turn will show a bogus A/V desync message on unpause. So just
reset it again on unpause.
I guess the removed code is an old leftover, and makes no sense anymore.
Should fix weird A/V diff dropouts when frames are being dropped with
display-sync.
If the player sends a frame with duration==0 to the VO, it can trivially
underrun. Don't panic, but keep the correct time.
Also, returning the absolute time from vo_get_next_frame_start_time()
just to turn it into a float with relative time was silly. Rename it and
make it return what the caller needs.
80ms allowable desync was a bit too much. It'd allow for a range of
160ms, which everyone can notice. It might also be a bother to apply
compensation resampling speed for that long.
We always let audio slowly desync until a threshold is reached, and then
pushed it back by applying a maximum compensation speed. Refine what
comes afterwards: instead of playing with the nominal video speed, use
the actual required audio speed for keeping sync as measured by the A/V
difference. (The "actual" speed is the ideal speed with A/V differences
added.)
Although this works in theory, it's somewhat questionable how much this
works in practice. The ideal time value is actually not exact, but is
the time at which the frame is scheduled (could be compensated by using
the time_left calculations in handle_display_sync_frame()). It doesn't
account for speed changes or catastrophic discontinuities. It uses only
10 past frames.
As long as it's within the desync tolerance, do not change the audio
speed at all for resampling. This reduces speed changes which might be
caused by jittering timestamps and similar cases.
(While in theory you could just not care and change speed every single
frame, I'm afraid that such changes could possibly cause audio
artifacts. So better just avoid it in the first place.)
This is very "illustrative", unlike the video-speed-correction
property, and thus useful. It can also be used to observe scheduling
errors, which are not detected by the core. (These happen due to
rounding errors; possibly not evne our fault, but coming from
files with rounded timestamps and so on.)
Instead of looking at the current frame duration for the intended
speedup, look at all past frames, and find a good average speed. This
ties in with not wanting to average _all_ frame durations, which
doesn't make sense in VFR situations.
This is currently done in the most naive way possible, but already sort
of works for VFR which switches between frame durations that are
integer multiples of a base rate. Certainly more improvements could
be made, such as trying to adjust directly on FPS changes, instead of
averaging everything, but for now this is not needed at all.
Helps somewhat with muxer-rounded timestamps.
There is some danger that this introduces a timestamp drift. But since
they are averaged values (unlike as when using an incorrect container
framerate hint), any potential drift shouldn't be too brutal, or
compensate itself soon. So I won't bother yet with comparing the results
with the real timestamp, unless we run into actual problems.
Of course we still prefer potentially real timestamps over the
approximated ones. But unless the timestamps match the container FPS,
we can't know whether they are (no, checking whether the they have
microsecond components would be cheating). Perhaps in future, we could
let the demuxer export the timebase - if the timebase is not 1000 (or
divisible by it), we know that millisecond-rounded timestamps won't
happen.
Get rid of get_past_frame_durations(), which was a bit too messy. Add
a past_frames array, which contains the same information in a more
reasonable way. This also means that we can get the exact current and
past frame durations without going through awful stuff. (The main
problem is that vo_pts_history contains future frames as well, which is
needed for frame backstepping etc., but gets in the way here.)
Also disable the automatic disabling of display-sync if the frame
duration changes, and extend the frame durations allowed for display
sync. To allow arbitrarily high durations, vo.c needs to be changed
to pause and potentially redraw OSD while showing a single frame, so
they're still limited.
In an attempt to deal with VFR, calculate the overall speed using the
average FPS. The frame scheduling itself does not use the average FPS,
but the duration of the current frame. This does not work too well,
but provides a good base for further improvements.
Where this commit actually helps a lot is dealing with rounded
timestamps, e.g. if the container framerate is wrong or unknown, or
if the muxer wrote incorrectly rounded timestamps. While the rounding
errors apparently can't be get rid of completely in the general case,
this is still much better than e.g. disabling display-sync completely
just because some frame durations go out of bounds.
We need a frame duration even on start, because the number of vsyncs
the frame is shown is predetermined. (vo_opengl actually makes use of
this property in certain cases.)
"Missed" implies the frame was dropped, but what really happens is that
the following frame will be shown later than intended (due to the
current frame skipping a vsync).
(As of this commit, this property is still inactive and always
returns 0. See git blame for details.)
Apparently Windows treats windows that use OpenGL, cover an entire
screen and have the WS_POPUP style set or are topmost windows as
exclusive fullscreen windows that bypass DWM and cannot be covered
by other windows.
This means we can’t use dwmflush in fullscreen mode, and it also
means that no other window can cover mpv, and it makes the screen
flicker when switching to fullscreen mode.
This can be avoided by not setting the WS_POPUP flag.
Users can still access the old behavior by enabling stay-on-top
(which IMO at least makes sense—now we just need to get dwmflush
autodetection right to avoid nasty surprises).
fixes#2177
This loaded external .ass files via libass. libavformat's .ass reader is
now good enough, so use that instead.
Apparently libavformat still doesn't support fonts embedded into text
.ass files, but support for this has been accidentally broken in mpv for
a while anyway. (And only 1 person complained.)
Just set the ratio directly by working around the intended semantics of
the API function. The silly rounding stuff we had isn't needed anymore
(and not entirely correct anyway).
Note that since the compensation is virtually active forever, we need to
reset if it's not needed. So always run this code to be sure to reset
it.
Also note that libswresample itself had a precision issue, until it
was fixed in FFmpeg commit 351e625d.
When the audio format is not known yet and the audio chain is still
initializing, filter reinit will fail. Normally, attempts to
reinitialize filters at this stage should be rare (e.g. user commands
editing the filter chain). But it sometimes happened with track
switching in combination with the video code calling
update_playback_speed() at arbitrary times.
Get rid of the message by not trying to change the filters for the sake
of playback speed update while decoding is still being initialized.
Until now, we've relied on the following things:
- you can send flush packets to the decoder even if it's fully flushed,
- you can send new packets to a flushed decoder,
- you can send new packers to a partially flushed decoder.
("flushing" refers to sending flush packets to the decoder until the
decoder does not return new pictures, not avcodec_flush_buffers().)
All of these are questionable. The libavcodec API probably doesn't
guarantee that these work well or at all, even though most decoders have
no issue with these. But especially with hardware decoding wrappers
(like MMAL), real problems can be expected. Isolate us from these corner
cases by handling them explicitly.
This applies to unexpected freezes or deadlocks, not e.g. normal
framedrops. The verbose messages also might remind an API user if the
API usage is incorrect, such as not calling mpv_opengl_cb_draw() when a
redraw request was issued.
This commit introduces logic to read other volumes from the same source
as the primary archive. Both .rar formats as well as 7z are supported for now.
It also changes the libarchive callback structure to be per-volume
consistent with the libarchive intenal client data array constructed
with archive_read_append_callback_data.
Added open, close and switch callbacks. Only the latter is strictly
required to make sure that the streams always start at position 0, but
leaving all volumes open can eat a lot of memory for archives with many
parts.
The nnedi3 prescaler requires a normalized range to work properly,
but the original implementation did the range normalization after
the first step of the first pass. This could lead to severe quality
degradation when debanding is not enabled for NNEDI3.
Fix this issue by passing `tex_mul` into the shader code.
Fixes#2464
vo_opengl_cb is a special case, because we somehow have to render video
asynchronously, all while "trusting" the API user to do it correctly.
This didn't quite work, and a while ago a compromise using a timeout to
prevent theoretically possible deadlocks was added.
Make it even more synchronous. Basically, go all the way, and
synchronize rendering between VO and user renderer thread to the
full extent possible.
This means the silly frame queue is dropped, and we event attempt to
synchronize the GL SwapBuffer call (via mpv_opengl_cb_report_flip()).
The changes introduced with commit dc33eb56 are effectively dropped. I
don't even remember if they mattered.
In the future, we might make all VOs fetch asynchronously from a frame
queue, which would mostly remove the differences between vo_opengl and
vo_opengl_cb, but this will take a while (if it will even be done).
Pick the correct GLSL version from the GL_SHADING_LANGUAGE_VERSION
string. Might be somewhat questionable, as we expect the minor version
number not to have leading 0s.
Should help with cases when the reported GLSL version is much higher
than the equivalent of the reported GL version. This problem was
observed in combination with GL_ARB_uniform_buffer_object, which
can't be used if the declared GLSL version is too low.
Has the same function as setting the option.
This commit changes the property in a bunch of other ways. For example
if the VO is not created, it will return the option value.
Simplifies some auto detection matters.
I _still_ don't want to remove the lazy loading mechanism, because it's
still slightly useful for filters using the hwdec APIs. My main
motivation for not always preloading them is actually that libva prints
random useless crap to the terminal with no way to prevent this.
The examples demonstrates use with optical media, which is far from
mpv's main purpose.
The authors section is a leftover from MPlayer times. There are enough
other places which reiterate how mpv is based on mplayer2/MPlayer,
copyright statements, and so on.
Hint that the linked section contains information for Windows. (Well,
that's a lie, but it has a link to the Windows section.)
Avoid implying that lines in the config file end with ';'. Also, the <>
are probably just confusing.
Deal with jittering Matroska crap timestamps. This reuses the mechanism
that is needed for frames without PTS, and adds a heuristic to it. If
the interpolated timestamp is less than 1ms away from the real one, it
might be due to Matroska timestamp rounding (or other file formats with
such rounding, or files remuxed from Matroska).
While there actually isn't much of a need to do this (audio PTS
jittering by such a low amount doesn't negatively influence much), it
helps with identifying jitter from other sources.
Instead of requiring the decoder to set the PTS directly on the
dec_audio context (including handling absence of PTS etc.), transfer the
packet PTS to the decoded audio frame. Marginally simpler, and gives
more control to the generic code.
Notes:
- Unfortunately the only way to talk to EGL from within DRM I could find
involves linking with GBM (generic buffer management for Mesa.)
Because of this, I'm pretty sure it won't work with proprietary NVidia
drivers, but then again, last time I checked NVidia didn't offer
proper screen resolution for VT.
- VT switching doesn't seem to work at all. It's worth mentioning that
using vo_drm before introduction of VT switcher had an anomaly where
user could switch to another VT and input text to it, while video
played on top of that VT. However, that isn't the case with drm_egl:
I can't switch to other VT during playback like this. This makes me
think that it's either a limitation coming from my firmware or from
EGL/KMS itself rather than a bug with my code. Nonetheless, I still
left (untestable) VT switching code in place, in case it's useful to
someone else.
- The mode_id, connector_id and device_path should be configurable for
power users and people who wish to watch videos on nonprimary screen.
Unfortunately I didn't see anything that would allow OpenGL backends
to register their own set of options. At the same time, adding them to
global namespace is pointless.
- A few dozens of lines could be shared with vo_drm (setting up VT
switching, most of code behind page flipping). I don't have any strong
opinion on this.
- Sometimes I get minor visual glitches. I'm not sure if there's a race
condition of some sort, unitialized variable (doubtful), or if it's
buggy driver. (I'm using integrated Intel HD Graphics 4400 with Mesa)
- .config and .control are very minimal.
Signed-off-by: wm4 <wm4@nowhere>
While it seemed like a pretty good idea at first, it's just a dead end
and works only in the simplest cases. While it may or may not help
slightly with audio sync mode, the display-sync mode already compensates
this in a better way. The main issue is that timestamps at this layer
are not in order, so it can look at single timestamps only.
Essentially we'd use something random, just because it's part of the srt
of traditionally used ALSA channel mappings. But each driver can do its
own things.
This doesn't let me sleep at night, so remove it.
This could accidentally change some spdif formats to AAC (because AAC is
the first on the list and will match first). spdif formats are
inherently uninterchangeable, so treat them as their own class of
formats (like int vs. float).
Might fix some issues with ao_wasapi.c.
glXCreateContextAttribsARB() by design can throw some X11 errors. We
ignore these, but we generally still print error messages to the
terminal. This was confusing/annoying users, so silence it. The stupid
part is that the Xlib error handler is global, so we have to be slightly
careful here.
They are evil and should be eradicated. Some of these were pretty dumb
anyway.
There are probably some more around in platform specific code or other
code not enabled by default on Linux.
This is based on an older patch by James Ross-Gowan. It was rebased and
cleaned up. Also, the DWM API usage present in the older patch was
removed, because DWM reports nonsense rates at least on Windows 8.1
(they are rounded to integers, just like with the old GDI API - except
the GDI API had a good excuse, as it could report only integers).
Signed-off-by: wm4 <wm4@nowhere>
This simplifies update_screen_rect a bit. Unless --fs-screen=all is
used, it will always get an HMONITOR and call GetMonitorInfo to
determine its dimensions. This will make it easier for the next few
commits to determine the colour profile and the refresh rate from the
HMONITOR.
There is a slight change in behaviour. When selecting a screen that is
out of range, such as --screen=9 on a machine with only two monitors,
the old code would silently select the last existing monitor. The new
code prints an error message and falls back to the default screen (same
as the Cocoa code.)
Signed-off-by: wm4 <wm4@nowhere>
The call to EnumDisplaySettings seems to be a relic from when MPlayer
ran on systems that didn't have GetMonitorInfo or SM_CX/CYVIRTUALSCREEN.
GetMonitorInfo was loaded dynamically, so it was possible for MPlayer to
run without it and use the values returned by EnumDisplaySettings.
These are always present in modern versions of Windows, so the values
returned from EnumDisplaySettings are always overwritten. Remove the
call to EnumDisplaySettings and assume SM_CX/CYVIRTUALSCREEN is always
present.
Signed-off-by: wm4 <wm4@nowhere>
Commit 27dc834f added it as such.
Also remove the check for glUniformBlockBinding() - it's part of an
extension, and the check glGetUniformBlockIndex() already checks whether
the extension is fully available.
Implement NNEDI3, a neural network based deinterlacer.
The shader is reimplemented in GLSL and supports both 8x4 and 8x6
sampling window now. This allows the shader to be licensed
under LGPL2.1 so that it can be used in mpv.
The current implementation supports uploading the NN weights (up to
51kb with placebo setting) in two different way, via uniform buffer
object or hard coding into shader source. UBO requires OpenGL 3.1,
which only guarantee 16kb per block. But I find that 64kb seems to be
a default setting for recent card/driver (which nnedi3 is targeting),
so I think we're fine here (with default nnedi3 setting the size of
weights is 9kb). Hard-coding into shader requires OpenGL 3.3, for the
"intBitsToFloat()" built-in function. This is necessary to precisely
represent these weights in GLSL. I tried several human readable
floating point number format (with really high precision as for
single precision float), but for some reason they are not working
nicely, bad pixels (with NaN value) could be produced with some
weights set.
We could also add support to upload these weights with texture, just
for compatibility reason (etc. upscaling a still image with a low end
graphics card). But as I tested, it's rather slow even with 1D
texture (we probably had to use 2D texture due to dimension size
limitation). Since there is always better choice to do NNEDI3
upscaling for still image (vapoursynth plugin), it's not implemented
in this commit. If this turns out to be a popular demand from the
user, it should be easy to add it later.
For those who wants to optimize the performance a bit further, the
bottleneck seems to be:
1. overhead to upload and access these weights, (in particular,
the shader code will be regenerated for each frame, it's on CPU
though).
2. "dot()" performance in the main loop.
3. "exp()" performance in the main loop, there are various fast
implementation with some bit tricks (probably with the help of the
intBitsToFloat function).
The code is tested with nvidia card and driver (355.11), on Linux.
Closes#2230
Add the Super-xBR filter for image doubling, and the prescaling framework
to support it.
The shader code was ported from MPDN extensions project, with
modification to process luma only.
This commit is largely inspired by code from #2266, with
`gl_transform_trans()` authored by @haasn taken directly.
This check disables the display-sync resample method. If the filters
convert PCM to AC3, we can still insert a filter to change speed. This
is because filters are inserted at the beginning of the filter chain.
Actually, it didn't really require that before (most work was avoided),
but some bits had to be run anyway. Separate the speed change into a
light-weight function, which merely updates already created filters, and
a heavy-weight one which messes with filter insertion.
This also happens to fix the case where the filters would "forget" the
current speed (force resampling, change speed, hit a volume control to
force af_volume insertion - it will reset speed and desync).
Since we now always run the light-weight function, remove the
af_scaletempo verbose message that is printed on speed setting. Other
than that, all setters are cheap.
Move it (in a cosmetic sense), and also move its invocation to below all
the video handling.
All other changes remain cosmetic, including moving the framedrop
calculation code, and getting rid of the video_speed_correction
variable.
For some reason, the encoder didn't like that the AVPacket already had
fields set. I'm not quite sure, but this might just be invalid API
usage. Do it as it's recommended.
We still have a sample-based buffer between filters and audio outputs.
In order to avoid cutting frames into half (which can upset receivers),
we strictly need to align the boundaries on which we cut the audio.
Update msg.c state immediately if a terminal or logging setting is set.
Until now, this was delayed until mp[v]_initialize() was called. When
using the client API, you could easily miss logged error messages, even
when logging was initialized early on by calling
mpv_request_log_messages().
(Properties can't be used for this either, because properties do not
work before mpv_initialize().)
The noframe event is logged whenever there is no new frame. This can
happen due to normal redraws, but also due to video frame queue
underflow.
The mpv_opengl_cb_report_flip() API function is currently pretty
useless, because blocking on the video frame queue is more reliable and
simpler. But at least we can log the actual vsync.
next_vsync/prev_vsync was only used to retrieve the vsync duration. We
can get this in a simpler way.
This also removes the vsync duration estimation from vo_opengl_cb.c,
which is probably worthless anyway. (And once interpolation is made
display-sync only, this won't matter at all.)
We need to effectively swap the last channel pair. See commit 4e358a96
and 5a18c5ea for details.
Doing this seems rather strange, as 7.1 just extends 5.1 with 2 new
speakers, and 5.1 doesn't need this change. Going by the HDMI standard
and the Intel HDA sources (cited in the referenced commits), it also
looks like 7.1 should simply append two channels to 5.1 as well. But
swapping them is apparently correct. This is also what XBMC does. (I
didn't find any other applications doing 7.1 PCM using the ALSA channel
map API. VLC seems to ignore the 7.1 case.) Testing reveals that at
least the end result is correct.
"Normal" ALSA 7.1 is unaffected by this, as it reports a different
(and saner) channel layout.
Instead of constructing an ALSA channel map from mpv ones from scratch,
try to find the original ALSA channel map again. Th result is that we
need to convert channel maps only in one direction. If we need to map
a mp_chmap to ALSA, we fetch the device's channel map list, convert
each entry to mp_chmap, and find the first one which fits.
This seems helpful for the following commit. For now, this only gets rid
of mapping back the trivial MONO mapping, which alone would still be
acceptable, but with other channel layout mogrifications it gets messy
fast. While we need to do something awkward to keep our channel map
reordering for VAR chmaps (which basically gives nicer output and
possibly slightly better performance), this is still the better
solution.
This reverts commit 4e358a9636.
Testing shows the channel pairs must indeed be swapped (details see
commit message of the reverted commit). Making the downmix code move
sl/sr to sdl/sdr is not an appropriate solution anymore, and it's
better to fix the unusual channel layout in ao_alsa.c directly.
(Not reverting the change in chmap.c; this is still correct.)
This affects only the display-sync code path, as for normal timing the
wakeup_pts stuff handles proper wakeup. It's probably mostly a
theoretical issue.
Discontinuities (like toggling fullscreen) can cause multiple frames to
be dropped in succession, which sounds very weird. It's better to drop
some video frames instead to compensate for larger desyncs.
We roughly base it on the maximum allowed speed changes (audio change is
"additional" to the video change to account for deviations when playing
at max. video speed change).
A hw decoder might fail to decode a frame for multiple reasons, and not
always just because decoding is impossible. We can't generally
distinguish these reasons well. Make it more tolerant by accepting
failures of 3 frames, but not more. The threshold can be adjusted by the
repurposed --vd-lavc-software-fallback option.
(This behavior was suggested much earlier in some PR, but at the time
the "proper" hwdec fallback was indistinguishable from decoding error.
With the current situation, "proper" fallback is still instantious.)
ao_alsa: attempt to fix 7.1 over HDMI
The last 2 channels of 7.1 (RLC/RRC in ALSA) were exported as sdl/sdr
instead of sl/sr (I don't even know why I chose sdl/sdr, but SL/SR
and RLC/RRC are different in the ALSA API). libsw/avresample do not
move the sl/sr channels to sdl/sdr when rematrixing, so silence was
sent for 2 channels. If my selection of sdl/sdr is essentially API
abuse, there's no reason why they should do this differently.
The mess here is really that ALSa doesn't map the HDMI layouts cleanly.
Most ALSA drivers export 7.1 in a way compatible to our expectations,
but Intel HDA/HDMI does not:
mpv/ffmpeg: fl-fr-fc-lfe-bl-br-sl-sr
ALSA/generic: FL FR FC LFE RL RR SL SR [1]
ALSA/HDMI: FL FR LFE FC RL RR RLC RRC [2]
The HDMI layout is layout 0x13 (going by CEA-861-B). The comment in
the kernel code has to be correct too. The early standard defines only
1 other layout, which replaces RLC/RRC with FRC/FLC - this probably
corresponds to what we call "7.1(wide)".
So it appears when ALSA requests RLC/RRC, we should feed it sl/sr.
To make it more complicated, Kodi/xbmc apparently also have to deal with
ALSA being special, but instead of sending sl/sr to RLC/RRC, they swap
the last two pairs of the layout, and send sl/sr to RL/RR and bl/br to
RLC/RRC. Or I might have misunderstood their code. I don't have a
7.1-capable A/V receiver, so I can't test this.
For now, go with the simpler solution, and wait until someone tests it.
If the speakers end up swapped, a completely different solution will be
needed.
[1] https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/core/pcm_lib.c?id=refs/tags/v4.3#n2434
[2] https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/pci/hda/patch_hdmi.c?id=refs/tags/v4.3#n307
These calls actually can leave the ALSA configuration space empty (how
very useful), which is why snd_pcm_hw_params() can fail. An earlier
change intended to make this non-fatal, but it didn't work for this
reason.
Backup the old parameters, so we can retry with the non-empty
configuration space. (It has to be non-empty, because the previous
setters didn't fail.)
Note that the buffer settings are not very important to us. They're
a leftover from MPlayer, which needed to write enough data to the
audio device to not underrun while decoding and displaying a video
frame. In mpv, most of these things happen asynchronously, _and_
there is a dedicated thread just for feeding the audio device, so
we should be pretty imune even against extreme buffer settings. But
I suppose it's still useful to prevent PulseAudio from making the
buffer too large, so still keep this code.
Again, this could have bad access, is unlikely, and has no bad
consequences. It's noteworthy that vlc and the ALSA PCM example both do
this first, even if they set the sample rate later.
I'm worried that not restricting the access type before restricting the
format will cause problems. While it's unlikely, it might prevent
failures in some corner cases. Also, since we by default always use
interleaved access (buggy ALSA plugins), this will have no effects at
all.
If the API doesn't list padded channel maps, but the final device
channel map is padded, and if unpadded output is not possible (unlike in
the somewhat similar dmix case), then we shouldn't apply the channel
count mismatch fallback in the beginning. Do it after channel map
negotiation instead.
Doesn't matter much; effectively this prevents just log spam in some
cases where the map is legitimately padded. Normally this is really
only needed for the dmix ALSA case. (See git blame for details.)
The uninit() function was called twice if the uninit() function failed
(once by init(), once by vd_lavc.c code), which caused crashes due to
double-free. (This failure is a corner case, and all other hwdec
backends appear to handle this case gracefully.)
I do not think this code should be able to deal with uninit() being
called other than once. Guarantee that it's called exactly once.
Quoting MSDN: "Notifies the Desktop Window Manager (DWM) to opt in to or
out of Multimedia Class Schedule Service (MMCSS) scheduling while the
calling process is alive.". Whatever this means. (An application can
change the scheduling priority of the window manager?)
Does this improve anything? I have no idea. Certainly this is a program
that does multimedia and graphics, so we seem to be a good match for
this.
Is it bad if we enable this even while playback is inactive or paused? I
have no idea either.
Is there a magic cargo cult function that will mark our renderer thread
as multimedia thing? I have no idea. (We use a function to enable MMCSS
for our audio thread in ao_wasapi.)
Enable it by default, but not unconditionally. Add an "auto" mode, which
disable DwmFlush if the compositor is (probably) inactive. Let's see how
this goes.
Since I accidentally enabled DwmFlush always by default (more or less)
in a previous commit touching this code, this is probably mostly just
cargo-culting, and it's uncertain whether it does anything.
Note that I still got bad vsync behavior when fullscreening mpv, and
making another window visible on the same screen. This happens even if
forcing DWM.
update_av_diff() works on the timestamps, while time_left is in real
time. When playing at not-1 speed, these are very different, and cause
the A/V difference to jitter. Fix this by scaling the expected A/V
desync to the correct range.
Commit acd5816a broke this. It was stopping playback occasionally.
Another case where the non-display-sync interpolation mode
(in->vsync_timed==true) is causing a lot of subtle issues and will be
removed soon.
PAL8 is the only format that is RGB, has only 1 component, is byte-
aligned. It was accidentally detected by the GBRP case as planar RGB.
(It would have been ok if it were gray; what ruins it is that it's
actually paletted, and the color values do not correspond to colors (but
palette entries).
Pseudo-pal formats are ok; in fact AV_PIX_FMT_GRAY is rightfully marked
as MP_IMGFLAG_YUV_P.
This didn't show up with cases where the frame pattern has a cycle of 1
or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show
up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default
settings.)
In this case, we must not add the timing error of the next frame to the
A/V difference estimation of the current frame. Use the previous timing
error instead.
This is another bug resulting from the confusion about whether we
calculate parameters for the currently playing frame, or the one we're
about to queue.
Regression since commit 93db4233. I think the bit that was forgotten
here was to remove the vo_w32_config() return value completely. The VO
failed to init because that function always returned 0. This commit
removes these bits and fixes the VO.
Fixes#2434.
Yet another relatively useless option that tries to make OpenGL's sync
behavior somewhat sane. The results are not too encouraging. With a
value of 1, vsync jitter is gone on nVidia, but there are frame drops
(less than with glfinish). With 2, I get the usual vsync jitter _and_
frame drops.
There's still some hope that it might prevent too deep queuing with some
GPUs, I guess.
The timeout for the wait call is 1 second. The value is pretty
arbitrary; it should just not be too high to freeze the process (if
the GPU is un-nice), and not too low to trigger the timeout in normal
cases, even if the GPU load is very high. So I guess 1 second is ok
as a timeout.
The idea to use fences this way to control the queue depth was stolen
from RetroArch:
https://github.com/libretro/RetroArch/blob/df01279cf318e7ec90ace039d60515296e3de908/gfx/drivers/gl.c#L1856
Commit a1315c76 broke this slightly. Frame drops got counted multiple
times, and also vo.c was actually trying to "render" the dropped frame
over and over again (normally not a problem, since frames are always
queued "tightly" in display-sync mode, but could have caused 100% CPU
usage in some rare corner cases).
Do not repeat already dropped frames, but still treat new frames with
num_vsyncs==0 as dropped frames. Also, strictly count dropped frames in
the VO. This means we don't count "soft" dropped frames anymore (frames
that are shown, but for fewer vsyncs than intended). This will be
adjusted in the next commit.
vo_frame.num_vsyncs can be != 1 in some cases in normal sync mode too.
This is not a very exact fix, but in exchange it's robust. (These
vo_frame flags are way too tricky in combination with redrawing and
such.)
There were occasional shader compilation and rendering failures if FBOs
were unavailable. This is caused by the FBO caching code getting active,
even though FBOs are unavailable (i.e. dumb-mode).
Boken by commit 97fc4f.
Fixes#2432.
Fixes linker failure. How did this ever work? Apparently it did most of
the time, but apparently we just got the first case where it didn't.
Fixes#2433.
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Martin Herkt