This doesn't take a ',' separated list. --script is just an alias for
--scripts--append. --scripts accepts a list, but uses the
mplayer-inherited platform-dependent path separator.
Fixes: #5996
JSON doesn't support these for some god-awful reason. (JSON would have
been so much better if it weren't based on JavaScript, the plague of
this world.)
We don't really care whether these specific values "round trip", so we
might as well write them in a standard-compliant way.
Untested. I was too lazy to even run this, but it probably works.
See #6691.
Internally, vo_gpu uses NaN for some options to indicate a default value
that is different depending on the context (e.g. different scalers).
There are 2 problems with this:
1. you couldn't reset the options to their defaults
2. NaN is a damn mess and shouldn't be part of the API
The option parser already rejected NaN explicitly, which is why 1.
didn't work. Regarding 2., JSON might be a good example, and actually
caused a bug report.
Fix this by mapping NaN to the special value "default". I think I'd
prefer other mechanisms (maybe just having every scaler expose separate
options?), but for now this will do. See you in a future commit, which
painfully deprecates this and replaces it with something else.
I refrained from using "no" (my favorite magic value for "unset" etc.)
because then I'd have e.g. make --no-scale-param1 work, which in
addition to a lot of effort looks dumb and nobody will use it.
Here's also an apology for the shitty added test script.
Fixes: #6691
Apparently this was only added and used for cache update stuff, which
was changed in commit 8dbc93a94c. Now it's unused, messy, ugly, and
is in the way, so get rid of it.
Before this commit, the status line used terminal control codes only if
stderr was a terminal. I'm not sure why this was done, and git blame
tracks it back to a huge commit by me, which changed all of the terminal
handling.
A user complained, so just stop treating this specially for no reason.
Fixes: #6617
The code is very basic:
- only handles gamepads, could be extended for generic joysticks in the
future.
- only has button mappings for controllers natively supported by SDL2.
I heard more can be added through env vars, there's also ways to load
mappings from text files, but I'd rather not go there yet. Common ones
like Dualshock are supported natively.
- analog buttons (TRIGGER and AXIS) are mapped to discrete buttons using an
activation threshold.
- only supports one gamepad at a time. the feature is intented to use
gamepads as evolved remote controls, not play multiplayer games in mpv :)
This can be used by distros to disable all known FFmpeg ABI violations.
Currently only 1 is known, in demux_lavf.c. In addition to if-defing out
the access to the private FFmpeg field, this disables the possibly
fragile nested open callbacks, which make sense only if the
aforementioned field can be accessed.
Form some reason (and because of my fault), vf_format converts image
formats, but nothing else. For example, setting the "colormatrix"
sub-parameter would not convert it to the new value, but instead
overwrite the metadata (basically "reinterpreting" the image data
without changing it).
Make the historical mistake worse, and go all the way and extend it such
that it can perform a conversion. For compatibility reasons, this needs
to be requested explicitly. (Maybe this would deserve a separate filter
to begin with, but things are messed up anyway. Feel free to suggest an
elegant and simple solution.)
This demonstrates how zimg can properly perform some conversions which
swscale cannot (see examples added to vf.rst).
Stupidly this requires 2 code paths, one for conversion, and one for
overriding the parameters.
Due to the filter bullshit (what was I thinking), this requires quite
some acrobatics that would not be necessary without these abstractions.
On the other hand, it'd definitely be more of a mess without it. Oh
whatever.
f_reset, which is called on seeks, was a good place for resetting the
warning flag (so the warning would be print again). Except some other
code abused f_reset when all metadata was read (in both cases you want
to clear the metadata). Instead of spending more time on getting this
flag reset correctly, just never reset it.
There's 2 stupid things here that need to be fixed. First of all,
vulkan wasn't actually using presentation time because somehow the
get_vsync function in context.c disappeared. Secondly, if the mpv window
was hidden it was updating the ust time based on the refresh_usec but
really it should simply just not feed any information to the vsync info
structure. So this adds some logic to assume whether or not a window is
hidden.
Normally, input and output are orthogonal. But zimg may gain image
formats not supported by FFmpeg, which means the conversion will only
work if zimg is used at all. This on the other hand, depends on whether
the other format is also supported by zimg. (For example, a later commit
adds RGB30 output to zimg. libswscale does not support this format. But
if you have P010 as input, which the zimg wrapper does not support at
all, the conversion won't work.)
This makes such a function needed; so add it.
The FFmpeg version was last bumped a long time ago, except in commit
1638fa7b46, where it was used for some obscure pixel
format.
This is pretty annoying, so make it optional.
The RGB pack/unpack code in theory supports packed, non-subsampled YUV,
although in practice FFmpeg defines no such formats. (Only one with
alpha, but all alpha input is rejected by the current code.)
This would in theory have failed, because we would have selected a GBRP
format (instead of YUV), which makes no sense and would either have been
rejected by zimg (inconsistent parameters), or lead to broken output
(wrong permutation of planes).
Select the correct format and don't permute the planes in the YUV case.
As suggested by the zimg author. This is mostly related to XYZ support.
It's unclear whether this works. Using the only XYZ test sample we know,
and the next commits to consume the pixfmt, it looks wrong.
According to the zimg author, YUV->GREY conversion does not even read
the chroma planes, as long as no matrix conversion is involved. Since we
try to avoid the latter anyway by forcing the source parameters on the
target image, passing only the Y plane will not help with anything.
An unscientific test seems to confirm this, so remove this.
This would probably help with libswscale (I didn't test this), but on
the other hand, libswscale will rarely be used in cases where we can
extract the Y plane. (Except nv12, which should probably be added to the
zimg wrapper's unpacking.)
And update the comment both explaining why this defaulting matters and
why we use BT.2020 instead.
tl;dr BT.709 clips even the one test file we *do* have, so if we don't
handle XYZ "natively" in vo_gpu we might as well at least handle it in a
way that runs less risk of clipping
This no longer hard-codes BT.709, it converts to whatever primaries are
tagged in the same metadata struct. The actual BT.709 defaulting comes
from `mp_image_params_guess_csp`.
This will perform conversion and scaling of video with zimg, if
--sws-allow-zimg is used.
The performance probably depends on how well the compiler optimizes the
RGB pack code in zimg.c, which is written in C.
Awful shit. I probably wouldn't accept this code from someone else, just
so you know.
The idea is that a sws_utils user can automatically use zimg without
large code changes. Basically, laziness. Since zimg support is still
very new, and I don't want that anything breaks just because zimg was
enabled at build time, an option needs to be set to enable it. (I have
especially especially obscure stuff in mind, which is all what
libswscale is used in mpv.)
This _still_ doesn't cause zimg to be used anywhere, because the
sws_utils user has to opt-in by setting allow_zimg. This is because some
users depend on certain libswscale features.
This provides a very similar API to sws_utils.h, which can be used to
convert and scale from one mp_image to another.
This commit adds only the code, but does not use it anywhere.
The code is quite preliminary and barely tested. It supports only a few
pixel formats, and will return failure for many others. (Unlike
libswscale, which tries to support anything that FFmpeg knows.)
zimg itself accepts only planar formats. Supporting other formats
requires manual packing/unpacking. (Compared to libswscale, the zimg API
is generally lower level, but allows for more flexibility.) Only BGR0
output was actually tested. It appears to work.
This used to be needed for the "GPU memcpy" (shitty Intel methods to
deal with certain uncached memory types). This is now done in FFmpeg,
and the code in mp_image.c was just unnecessarily convoluted.
The plane pointer checking assert() triggered at least on gray8, because
that has a "pseudo palettes" in ffmpeg, which mpv refuses to allocate.
Remove a strange duplicated printf().
Log the component type where available.
(Why is this even here, I hate it when there are commented test programs
in source files.)
Since a track may not be selected twice, it makes sense e.g. for
secondary subtitles to select the next best match and avoid the
duplicate selection.
This allows for example `--slang=en,ja --secondary-sid=auto` to select
'en' as primary and 'ja' as secondary without needing to know the actual
sid for 'ja'.
This was obviously missing from the recent commit, which probably broke
10 bit decoding. The original commit didn't test this for lack of
working hardware; this commit isn't tested either.
Fixes: a1c7d61393
This was speculatively added 2 years ago in preparation for something
that apparently never happened. The D3D code was added as an "example",
but this too was never used/finished.
No reason to keep this.
This syscall avoids the need to guess an unused filename in /dev/shm and
allows seals to be placed on it. We immediately return if no fd got
returned, as there isn’t anything we can do otherwise.
Seals especially allow the compositor to drop the SIGBUS protections,
since the kernel promises the fd won’t ever shrink.
This removes support for any platform but Linux from this vo.
On a audio/video desync by more than 0.5 seconds, display-sync mode was
disabled, and not enabled again (until playback restart, e.g. a seek).
The idea was that it this only happens when this playback mode is broken
and can't perform well anyway (A/V desync is a clear indication that
something is very wrong). Instead of behaving like a god damn POS, it
should revert to the more robust audio-sync mode.
Unfortunately, this could happen sporadically due to temporary system
performance problems, such as toggling fullscreen. Users didn't like
this, and asked for a function to disable it, or to recover in some
other way.
This mechanism is questionable anyway. If an ignorant user enables
display-sync, and encounters problems with it (without being able to
determine that display-sync is messing up), the player will still behave
like a POS on every playback, and even after every seek. It might
actually be helpful to fail more consistently. Also, I've found that
it's sill relatively reliable anyway even without this mechanism.
So just remove the fallback.
Fixes: #7048
vo_wayland was removed during the wayland rewrite done in 0.28. However,
it is still useful for systems that do not have OpenGL.
The new wayland_common code makes vo_wayland much simpler, and
eliminates many of the issues the previous vo_wayland had.
With the previous commit, this is dead code.
This also makes the f_autoconvert.c code for this dead code
(fortunately). Will probably remove this later.
2 years ago, ANGLE removed the old NV12-specific extension, and added
a new one that supports a number of formats, including P010. Actually
they just renamed it and removed their initial annoying and obvious
design error (bravo, Google).
Since it broke 2 years ago, nobody should give a shit about this code,
and it should just be removed. But for some reason I still dived the
shit-tank (Windows development).
I guess Intel code monkeys can't write drivers (or maybe the issue is
because we're doing zero-copy, which probably maybe is not actually
allowed by D3D11 due to array textures, see --d3d11va-zero-copy), so
the P010 path is completely untested. It doesn't work, I'll delete all
this ANGLE hwdec code.
Fixes: #7054
The --cache option does not take a number anymore. (Oh boy, this is
going to break a lot of user configs?)
The cache site is now configured with those obscure-sounding --demuxer
options.
--cache-secs is not useful anymore. The default is very high, so the
obscure-sounding --demuxer options determine how much is cached.
Advertise the --cache-on-disk option a bit. I found it useful once, and
it will trick users into wearing out their SSD for no gain, or so.
OK, so --cache-secs is useless, because the default is set to 10 hours.
And that part about the "maximum" was obviously a lie (I wonder if it
simply changed at some point).
Some failures by youtube-dl prompt the user to submit a bug report.
If such a failure occurs, we can compare youtube-dl's version to the
current calendar date to see how old it is. We don't make this check
on every youtube-dl failure, as failing to extract an URL is quite
common, and waiting for a second blocking python interpreter startup
for every such case would be a bit unpleasant.
Here the assumption is made that any youtube-dl version older than
3 months is probably severely out of date. Users will be warned about
this.
We also output the trimmed stderr of youtube-dl with msg.error, as this
appeared to have been the behaviour of utils.subprocess without stderr
capturing. Since this uses mp.command_native now, we'll have to do this
ourselves where appropriate.
Query information on the system output most linked to the swap chain,
and either utilize a user-configured format, or either 8bit
RGBA or 10bit RGB with 2bit alpha depending on the system output's
bit depth.
mpv warned if the FFmpeg runtime library version was not exactly the
same as the build version. This seemed to cause frequent conflicts. At
this point, most mpv code probably adheres to the FFmpeg ABI rules, and
FFmpeg stopped breaking ABI "accidentally". Another source of problems
were mixed FFmpeg/Libav installations, something which nobody does
anymore. It's not "our" job to check and enforce ABI compatibility
either. So I guess this behavior can be removed.
OK, still check for incompatible libraries (according to FFmpeg
versioning rules), i.e. different major versions, or if the build
version is newer than the runtime version. For now.
The comment about ABI problems is still true. In particular, the
bytes_read field mentioned in the removed comment is still accessed, and
is still an ABI violation. Have fun.
This was all dead code. Commit 995c47da9a (over 3 years ago) removed all
uses of the controls.
It would be nice if AOs could apply a linear gain volume, that only
affects the AO's audio stream for low-latency volume adjust and muting.
AOCONTROL_HAS_SOFT_VOLUME was supposed to signal this, but to use it,
we'd have to thoroughly check whether it really uses the expected
semantics, so there's really nothing useful left in this old code.
See previous commits. ao_sdl is worthless, but it might be a good test
for pull-based AOs.
This stops using the old underrun reporting if the new one is enabled.
Also, since the AO's behavior can in theory not be according to
expectations, this needs to be enabled for every single pull AO
separately.
For some reason, in certain cases I get multiple underrun warnings while
cache-pausing is active. It fills the cache, restarts the AO,
immediately underruns again, and then fills the cache again. I'm not
sure why this happens; maybe ao_sdl tries to catch up when it shouldn't.
Who knows.
I think this was _always_ wrong. Due to the line above the first changed
line, buffered_bytes==bytes always. I can only hope I broke this in a
less under-tested edit when I originally wrote this.
Fixes: c5a82f729b
The --cache-pause feature (enabled by default) will pause playback for a
while if network runs out of data. If this is not done, then playback
will go on frame-wise (as packets are slowly read from the network and
then instantly decoded and displayed). This feature is actually useless,
as you won't get nice playback no matter what if network is too slow,
but I guess I still prefer this behavior for some reason.
This commit changes this behavior from using the demuxer cache state
only, to trying to use underrun information from the AO/VO. This means
if you have a very large audio buffer, then cache-pausing will trigger
once that buffer is depleted, which will be some time _after_ the
demuxer cache has run out.
This requires explicit support from the AO. Otherwise, the behavior
should be mostly the same as before this commit.
This does not care about the AO buffer. In theory, the AO may underrun,
then the player will write some data to the AO buffer, then the AO will
recover and play this bit of data, then the player will probably trigger
the cache-pause behavior. The probability of this happening should be
pretty low, so I will hold off fixing this until the next refactor of
the AO chain (if ever).
The VO underflow detection was devised and tested in 5 minutes, and may
not be correct. At least I'm fairly sure that the combination of all the
factors should make incorrect behavior relatively unlikely, but problems
are possible.
Also, the demux_reader_state.underrun field may be inaccurate. It's only
the present state at the time demux_get_reader_state() was called, and
may exclude past underruns. In theory, this could cause "close" cases to
be missed. Then you might get an audio underrun without cache-pausing
acting on it. If the stars align, this could happen multiple times in
the row, effectively making this feature not work.
The most user-visible consequence of this change is that the user
will now see an AO underrun warning every time the cache runs out.
Maybe this cache-pause feature should just be removed...
AOs can now call ao_underrun_event() (in any context) if an underrun has
happened. It will print a message.
This will be used in the following commits. But for now, audio.c only
clears the underrun bit, so that subsequent underruns still print the
warning message.
Since the underrun flag will be used in fragile ways by the playback
state machine, there is the "reports_underruns" field that signals
strong support for underrun reporting. (Otherwise, underrun events will
not be used by it.)
This commit tries to prepare for better underrun reporting. The goal is
to report underruns relatively immediately. Until now, this happened
only when play() was called. Change this, and abuse that get_delay() is
called "relatively often" - this reports the underrun immediately in
practice.
Background:
In commit 81e51a15f7 (and also e38b0b245e), we were quite confused
about ALSA underrun handling. The commit message showed uncertainty how
case 3 happened, but it's blindingly obvious and simple.
Actually reading the code shows that ALSA does not have a concept of a
"final chunk" (or we don't use it). It's obvious we never pass the
AOPLAY_FINAL_CHUNK flag along to the ALSA API in any way. The only thing
we do is simply writing a partial fragment. Of course this will cause an
underrun. Doing a partial write saves us the trouble to pad the last
frame with silence, or so.
The main reason why the underrun message was avoided was that play() was
never called with a non-0 sample count again (except if reset() was
called before that). That was OK, at least the goal of avoiding the
unwanted message was reached. (And the original "bogus" message at end
of playback was perfectly correct, as far as ALSA goes.)
If network stalls, play() will called again only once new data is
available. Obviously, this could take a long time, thus it's too late.
The old way of using wayland in mpv relied on an external renderloop for
semi-accurate timings. This had multiple issues though. Display sync
would break whenever the window was hidden (since the frame callback
stopped being executed) which was really annoying. Also the entire
external renderloop logic was kind of fragile and didn't play well with
mpv's internal structure (i.e. using presentation time in that old
paradigm breaks stats.lua).
Basically the problem is that swap buffers blocks on wayland which is
crap whenever you hide the mpv window since it looks up the entire
player. So you have to make swap buffers not block, but this has a
different problem. Timings will be terrible if you use the unblocked
swap buffers call.
Based on some discussion in #wayland, the trick here is relatively
simple and works well enough for our purposes. Instead we basically
build a way to block with a timeout in the wayland buffer swap
functions.
A bool is set in the frame callback function that indicates whether or
not mpv is waiting for a frame to be displayed. In the actual buffer
swap function, we enter into a while loop waiting for this flag to be
set. At the same time, the wl_display is polled to block the thread and
wakeup if it receives any events from the compositor. This loop only
breaks if enough time has passed or if the frame callback bool is
received.
In the near future, it is better to set whether or not frame a frame has
been displayed in the presentation feedback. However as a first pass,
doing it in the frame callback is more than good enough.
The "downside" is that we render frames that aren't actually shown on
screen when the player is hidden (it seems like wayland people don't
like that). But who cares. Accurate timings are way more important. It's
probably not too hard to add that behavior back in the player though.
The externally driven renderloop was originally added for the wayland
context (to make display sync somewhat work), but it has a lot of issues
with mpv's internal structure. A different approach should be used.
This reverts commit a743fef837.
This affects hwdec_dxva2dxgi, which uses ra_d3d11_wrap_tex to wrap RGB
video frames that are shared with a D3D9 device. Without it, mpv uses
nearest instead of bilinear scaling with --scale=bilinear (the default)
and --hwdec=dxva2. It's kind of hard to believe this bug has gone
unnoticed for almost two years, but that seems to have been the case.
Fixes: #7042
In pseudo-DASH mode, we may have no real streams opened until the
demuxer layer is fully loaded and playback actually starts. The only
hint that the stream is from network is, at that point, the init
segment, which is only opened as stream, and then separately as demuxer
(which is dumb but happened to fit the internal architecture better).
So just propagate the flags from the init segment stream. Seems like an
annoyance, but doesn't hurt that much I guess. (Until someone gets the
idea to pass the init segment data inline or so, but nothing does that.)
The sample link in the linked issue will probably soon switch to another
format, because that service always does this after recent uploads or
so.
Fixes: #7038
mpv typically decodes and filters at least 2 frames before starting
playback. This happens during seeks, as well as when playback starts
from the beginning of the file.
skip-logo.lua receives notifications for all filtered frames, even
during seeking. It should interrupt during seeking, so as a crude
heuristic, it ignored all frames while the player was seeking. This does
not mean all these frames are skipped due to seeking (thus it's a "crude
hueristic"). In particular, it means that the first 2 frames of a video
cannot be skipped, since they're filtered within the playback restart
phase (equivalent to "seeking").
Fix this by making the heuristic slightly less crude. Since we observe
the property as "none", the property is not actually read until we do it
explicitly. By not reading it during seeking, we can let the frames
internally queue up (vf_fingerprint discards them in a ringbuffer-like
fashion if they're too many). Then, if seeking ends, we get the current
playback timestamp, and check queued up frames that are at or after that
timestamp. (In some ways, this duplicates what the player's seeking
logic does.)
A disadvantage is that this is racy. While playback-time is guaranteed
to be set when seeking changes from false to true, playback could
already have progressed to the next frame (or more) before the script
gets time to react. In theory, we could add a seek restart hook or so,
but I don't want to. A property that returns the last playback restart
time would also do it, but feels to special. Not an important problem
in practice anyway.
Someone crazy is trying to mix images with videos in EDL files. Putting
an image as first thing into the EDL disabled audio, because the first
EDL entry was used to define the layout.
Change this. Make it user-configurable, and use a "better" heuristic to
select the default otherwise.
In theory, EDL could be easily extended to specify track layout and
mapping of parts to virtual EDL tracks manually and in great detail. But
I don't think it's worth it - who would bother using it?
Fixes: #6764
Unless --video-latency-hacks, always decode 2 frames on playback
restart. This in turn will always compute the correct frame duration
(even for the first frame), which in turn happens to fix that playback
with an image at the beginning breaks display.
If a still image precedes video, and the size/format of the frame is
different from that of the video following it, the incorrect frame
duration caused vo_reconfig2() to be called early, causing the window to
resize, and the renderer to clear the image to black. Specifically, it
hit the default value of 1 second duration (for still images), so the
image was displayed for 1 second, and changed to black until the next
proper video frame was displayed.
Normally this does not happen. Even if a video file displays still
images, it normally repeats the still image at the video's FPS (which is
sane). But you can construct such files, or use EDL to construct
something similarly behaving.
This change may increase seek latency a bit in audio video-sync mode
(the default). It needs to wait until 2 frames are decoded, before it
bothers to display the first frame. This is done even when seeking. In
theory it might be good to introduce a "seek preview" mode, which shows
the target image without all the preparations needed for starting
playback. (For example, it could not decode audio.) But since I'm using
video-sync=display-resample, which already needed to always decode 2
frames, I don't think this is a terribly high priority, nor do I
consider the slightly slower seeking a regression.
Fixes: #6765
In this case, gapless will most likely not work. It will result in (very
slight) desync, or (more commonly with small buffer sizes), in an
underflow.
I think it would be legitimate to disable gapless at end of playback
completely if video is enabled at all. But this would need an exception
for cover art mode, so I guess the current solution is OK as well.
It turns out that case 2) mentioned in the previous commit happened
quite often when playback ended normally.
There is probably a legitimate underrun with normal buffer sizes (100
ms, 4 fragments, gapless audio in "weak" mode). This is a result of the
player waiting for video to end, and/or the time needed to kill the
video window. The former case means that it depends on your test case
whether it happens (a file where video ends slightly before audio is
less likely to trigger it).
This in turn is due to how gapless playback works. Achieving not having
a "gap" requires queuing the audio of the next file without playing a
partial chunk (as AOPLAY_FINAL_CHUNK would do). The partial chunk is
then played as part of the first chunk played from the next file. But if
it detects "later" that there is no next file, it still needs to get rid
of the last fragment with AOPLAY_FINAL_CHUNK. At this point it's too
late, and an underrun may have actually happened. The way the player
uninits and reinits the entire playback engine for the next file in a
"serial" manner means it cannot know in advance whether this works.
This is the reason why the idiot who added the underrun exception for
the last chunk in play() was wrong (I wrote that btw., before you accuse
me of being rude). Yes, it's a real underrun, and you could probably
hear it.
This XRUN (aka underrun) message was printed in the following
situations:
1) legitimate underrun during playback
2) legitimate underrun when playing final chunk
3) bogus underrun when playing final chunk
The old underrun case (in play()) happens in cases 1) and 2) as well,
but 3) did not happen. It appears 3) is indeed something that happens,
although it's not known for sure. It's still pretty annoying, so remove
the new XRUN message.
When testing, care should be taken to play with buffer sizes, video
versus no video, and gapless enabled/disabled. Also, suspending the
player with Ctrl+Z in the terminal (SIGSTOP) and then resuming is a good
way to trigger a "normal" underrun.
all the get_property_* usages were removed because in some circumstances
they can lead to deadlocks. they were replaced by accessing the vo and
mp_vo_opts structs directly, like on other vos.
additionally the mpv helper was split into a mpv and libmpv helper, to
differentiate between private and public APIs and for future changes
like a macOS vulkan context for vo=gpu.
Another thing nobody will read. I'm attempting to document the rules by
which incompatible changes can be made. These rules have always been
present in this project, but I don't think they were written down. Or
maybe they were, but I forgot where.
I think due to the time of the day it became increasingly incoherent
(not necessarily near the end of the text). Hopefully no logical or
freudian lapses in there.
Although it was not true at the time this was written, both the
"program" and "cache-size" are gone now.
Since the changelog is for the entire next release, it makes no sense to
mention these removed properties.
It also happens to make the description of this much simpler, because
it's a non-issue now. It's probably not even worth mentioning anymore.
The justification for this is the fact that the `video-aspect` property
doesn't work well with `cycle_values` commands that include the value
"-1".
The "video-aspect" property has effectively no change in behavior, but
we may want to make it read-only in the future. I think it's probably
fine to leave as-is, though.
Fixes#6068.
The description of the "playback_only" field in the "subprocess" command
says "you can't start it outside of playback". This did not work
correctly: if the player was started in idle mode in the first place,
the subprocess was allowed to run even with playback_only=yes.
This is a bug, and this change fixes it. Add a test for this to
command-test.lua.
For #7025.
There's potential confusion about how long a process started with the
"subprocess" command is allowed to live. Add some more explanations
regarding "subprocess" specifics (such as the playback_only field), and
things that apply to asynchronous commands in general.
Partially for #7025.
This is the proper fix to the memory leak @wm4 pointed out. It turns out
that when you autoprobe opengl and vo_wayland_init returns false,
vo_wayland_uninit is never actually executed. So you have a leftover
pointer. The vulkan context does this correctly which was why my old,
dumb "fix" broke it.
Dumb idea. The correct thing to do is to fix the preinit and context
creation so that the uninit is correctly executed when probing fails
(and then everything gets freed).
This reverts commit defc8f359c.
wm4 mentioned that the wayland autoprobe leaked. A simple oversight in
the wayland_common code forgot to free the vo_wayland_state if
vo_wayland_init returned false.
Leaks the entire zimg state on filter deinit. Not sure what I was
thinking; with some luck, I just didn't give a shit about this case, but
most likely I was thinking the same thing as always: nothing.
As pointed out by @olifre in #7016, this line of code was wrong. p->opts
at this point is a struct allocated and managed by m_config. opts->file
is a string, and m_config explicitly frees it on destruction. The line
of code in question replaced the opts->file value, and made both the old
and new value children of the talloc allocation, so they were _also_
freed on destruction.
This crashed due to a double-free. First, talloc auto-freeing freed the
string, then m_config explicitly called talloc_free() on the stale
pointer again.
As @v-fox pointed out, commit 36dd2348a1 seems to have triggered the
crash. I suspect this code merely worked out of coincidence, since it
allowed m_config to free the value first. This removed it from the
auto-free list, and thus did not result in a double-free. The change
in order of calling alloc destructors changed the order of these calls.
There is no strong reason why new behavior (as introduced by commit
36dd2348a1) would be wrong (it feels like cleaner behavior). On the
other hand, what the vf_vapoursynth code did is clearly unclean and
going by the m_config API, you're not at all supposed to do this.
m_config manages all memroy referenced by option structs, the end.
@olifre's suggested fix also would have been correct (not just hiding
the issue), I prefer my fix, as it doesn't mess with the option struct
in tricky ways.
This wouldn't have happened if mpv were written in Haskell.
I previously skipped creating the wl_output if the --fullscreen flag
with no --fsscreen_id was inputted, so the fullscreen video lands on the
correct output (where mpv was launched). This has breakage if someone
combines the --autofit flag (or other similar options with it). Instead,
just actually read xdg_shell spec and realize that you can pass NULL to
xdg_toplevel_set_fullscreen and let the compositor choose the output if
the user doesn't specify it. If this has issues, get a better
compositor.
This partially reverts commit a9d83eac40
("Remove optical disc fancification layers").
Mostly due to the timestamp crap, this was never really going to work.
The playback layer is sensitive to timestamps, and derives the playback
time directly from the low level packet timestamps. DVD/BD works
differently, and libdvdnav/libbluray do not make it easy at all to
compensate for this. Which is why it never worked well, but not doing it
at all is even more awful.
demux_disc.c tried this and rewrote packet timestamps from low level TS
to playback time. So restore demux_disc.c, which should bring behavior
back to the old often non-working but slightly better state.
I did not revert anything that affects components above the demuxer
layer. For example, the properties for switching DVD angles or listing
disc titles are still gone. (Disc titles could be reimplemented as
editions. But not by me.)
This commit modifies the reverted code a bit; this can't be avoided,
because the internal API changed quite a bit. The old seek resync in
demux_lavf.c (which was a hack) is replaced with a hack. SEEK_FORCE and
demux_params.external_stream are new additions.
Some of this could/should be further cleaned up. If you don't want
"proper" DVD/BD support to disappear, you should probably volunteer.
Now why am I wasting my time for this? Just because some idiot users are
too lazy to rip their ever-wearing out shitty physical discs? Then why
should I not be lazy and drop support completely? They won't even be
thankful for me maintaining this horrible garbage for no compensation.
Instead of using custom code.
Now if only f_lavfi knew what formats FFmpeg's vf_yadif accepts, this
could look much nicer, and wouldn't require the additional converter
filter setup.
This adds the function as seen in the f_autoconvert.h part of the patch.
It's pretty simple, but goes along with an intrusive code move. I guess
the resulting code is slightly nicer anyway.
Before this commit, enabling hardware deinterlacing via the
"deinterlace" option/property just failed if no hardware deinterlacing
was available. An error message was logged, and playback continued
without deinterlacing.
Change this, and try to copy the hardware surface to memory, and then
use yadif. This will have approximately the same effect as
--hwdec=auto-copy. Technically it's implemented differently, because
changing the hwdec mode is much more convoluted than just inserting a
filter for performing the "download". But the low level code for
actually performing the download is the same again.
Although performance won't be as good as with a hardware deinterlacer
(probably), it's more convenient than forcing the user to switch hwdec
modes separately. The "deinterlace" property is supposed to be a
convenience thing after all.
As far as the code architecture goes, it would make sense to auto-insert
such a download filter for all software-only filters that need it.
However, libavfilter does not tell us what formats a filter supports
(isn't that fucking crazy?), so all attempts to work towards this are
kind of hopeless. In this case, we merely have hardcoded knowledge that
vf_yadif definitely does not support hardware formats. But yes, this
sucks ass.
This just wraps the mp_image_hw_download() function as a filter and adds
some minor caching/error logging. (Shame that it needs to much
boilerplate, I guess.)
Will be used by the following commit. Wrapping it as filter seemed more
convenient than other choices.
Can be used with mp_chain_filters() to combine multiple filters into a
single one. This is a bit silly, but whatever. I'm making it an explicit
separate filter, because it lets the user access mp_filter.ppins against
all conventions.
Basically predicts what mp_image_hw_download() will do. It's pretty
simple if it gets the full mp_image. (Taking just a imgfmt would make
this pretty hard/impossible or inaccurate.)
Used in one of the following commits.
This reverts commit 6385a5fd1b, and in
addition removes the code that automatically inserts the vavpp filter.
The reason is the same as the commit that is being reverted: this
filter seems to trigger driver bugs. It can cause GPU freezes or
just doesn't work.
This variant of disabling the filter is better. There was no way to
add the "force" parameter to the automatically inserted filter, so
the old approach just made manual filter insertion (with the --vf
option or "vf" command) more cumbersome.
I don't think MPlayer/mplayer2 and Libav are well-known enough anymore
to warrant such a prominent place in the top-level README file of this
project. It's just useless noise to most users. So I've moved these
things to the FAQ.
Update some other minor things.
Autoload script now suppports loading of not only video, but also
image and audio files, in a manner, where one can configure which
of the groups (audio, videos, images) is currently enabled.
Use file script-opts/autoload.conf with key=value configuration keys
disabled,images,videos,audio to configure autoload script.
See documentation on top of the script
This is realized by dvbin-channel-switch-offset,
which is a numeric offset on the channel initially tuned to.
Since the channel list is kept in the stream alone
depending on detected hardware and chosen card,
and no available backchannel to the player, there's no direct
property which could be switched.
Using input.conf like:
H cycle dvbin-channel-switch-offset up
K cycle dvbin-channel-switch-offset down
Q set dvbin-prog "ZDF HD"
allow fast and reliable channel switching again.
Reinitializes the player as is needed when
tuning to a new DVB channel.
Currently triggered when dvbin-prog is written to,
i.e. when the user explicity switches to a channel by name.
If any parameters have been updated, reinitiate streaming_start.
Throttle checks since we poll from streaming_read.
This also requires the player to re-initialize, since all video
and audio streams and even the transport format may change.
This is added in the next commit.
This is now treated in dvb_parse_path consistently
instead of logic scattered over various functions.
This is a requirement to sensibly re-evaluate config
after options have been changed, since we have two ways
to configure the stream (decorated URI and config parameters).
This was added in 585f9ff42f by @bbarenblat (github handle). We
don't do this. This file alone probably has multiple dozen of authors (I
didn't count, but it has a history of 15 years). If everyone added their
names with each small change, this project would have giant lists of
contributing authors on every source file.
Neither copyright law nor any of the used licenses require listing
authors in the license header. Authorship is recorded in the git log.
So don't start with this, and remove this recent case to avoid setting a
precedent.
Some files still have an author in the header. These cases are
grandfathered, and usually are the actual authors of the original code.
Still trying to get people to read it. Even though I wanted to make it
less of a wall of text and more readable, it got bigger. Oops.
While I'm at it, violate my own rules and mix these mostly cosmetic
changes with some actual rule changes and clarifications.
How hard can it be? I know contribute.md is a shitty wall of text, but
that doesn't make it less important, and each violation will make it
take longer by adding another review round anyway.
And we really don't need lazy pull requests. If you can't be assed to
follow a few simple rules, your code is probably shit or you wanted to
be quick and lazy. Why should we accept it? We're the ones who have to
maintain it and fix bugs in it, and if the contributor is lazy, the
chance of you maintaining it is probably slim as well. On the other
hand, WE the maintainers are not obligated to anything.
Don't say that though, the first contact doesn't need to be negative. I
don't know if the "lazy pull requests" is still too strong, but I can't
tell.
demux_mkv may seek to the end of the file to read certain headers (which
should probably be called "footers", but in theory they are just headers
that have been placed at the end of the file unfortunately).
This commit changes behavior not to seek if the stream is not marked as
seekable. Before this, it only checked whether the stream size was
unknown (end negative). In practice it doesn't make much of a
difference, since seekable usually equals known stream size.
Also improve the wording, and distinguish between actual incomplete
files, and unseekable ones.
Normalize nullptr and an empty string both to nullptr to simplify
handling. API users cannot set a value back to nullptr, so both
an empty string as well as nullptr should behave the same.
Turns out clearing all frambuffers in reconfig isn't such a great idea
when you also end up here when setting pan/scan.
I guess this is just a leftover from a previous iteration of vo_drm
where doing this made sense.
This passed all streams to mp_recorder_create(), even disabled ones. The
disabled streams never get packets, so recorder.c eventually errors out
with unrelated-looking errors. The reason is that recorder.c waits for
packets to appear on other streams, which in turn is because libavformat
refuses to mux empty streams anyway.
recorder.c could call demux_stream_is_selected(), which would have made
the patch much smaller. But this feels like a bad idea, since recorder.c
should use sh_stream only for metadata (and not in an "active" way), nor
should it care what demux.c is currently doing with it. So make the API
user (demux.c) pass only the streams it really wants.
Fixes: #6999
Looks like this didn't actually work. Prefetching will do nothing if
there isn't a thread to "drive" it, and the demuxer thread needs to be
explicitly enabled. (I guess I did the worst possible job in verifying
whether this actually worked when I implemented it. On the other hand,
the user didn't confirm back whether it worked, so who cares.)
Like in the previous commit, bad factoring makes everything worse. It
duplicates logic and implementation of enable_demux_thread(), since the
opener thread cannot access the mpctx->opts field freely. But it's deep
night, so fuck it.
Fixes: c1f1a0845eFixes: #6753
demux_start_prefetch() was called unconditionally in two cases. This is
completely wrong. I'm not sure what part of my brain died off that
something this obviously wrong went in.
The prefetch case is a bit more complicated. It's a different thread, so
you can't access just access mpctx->opts there. So add an explicit field
for this, which is the simplest way to get this done. (Even if it's bad
factoring.)
Fixes: c1f1a0845e
Fixes: 556e204a11
Although this was sort of elegant, it just seems to complicate things
slightly. Originally, the API meant that you cache mp_recorder_sink
yourself (which would avoid the mess of passing an index around), but
that too seems slightly roundabout.
In a later change, I want to change the set of streams passed to
mp_recorder_create(), and then I'd have to keep track of the index for
each stream, which would suck. With this commit, I can just pass the
unambiguous sh_stream to it, and it will be guaranteed to match the
correct stream.
The disadvantages are barely worth discussing. It's a new linear search
per packet, but usually only 2 to 4 streams are active at a time. Also,
in theory a user could want to write 2 streams using the same sh_stream
(same metadata, just writing different packets or so), but in practice
this is never done.
Add yet another variant of the stream open function. This time, make it
so that it's possible to add new open parameters in an extendable way,
which should put an end to having to change this every other year.
Effectively get rid of the overly special stream_create_instance()
function and use the new one instead, which requires changes in
stream_concat.c and stream_memory.c. The function is still in private in
stream.c, but I preferred to make the mentioned users go through the new
function for orthogonality. The error handling (mostly logging) was
adjusted accordingly.
This should not have any functional changes. (To preempt any excuses, I
didn't actually test stream_concat and stream_memory.)
Previously, there appeared to be implicit synchronisation in the
GL interop path, and we never observed any visual glitches. However,
recently, I started seeing stuttering in the GL path and on closer
examination it looked like read-before-write behaviour where GL
would display an old frame again rather than the current one.
After verifying that disabling hwdec made the problem go away,
I tried adding a cuStreamSynchronize() after the memcpys and that
also resolved the problem, so it's clearly sync related.
cuStreamSynchronize() is a CPU sync and so more heavy-weight than
you want, but it's the only tool we have. There is no mechanism
defined for synchronising GL to CUDA (It looks like there is a way
to synchronise CUDA to EGL but it appears one way and so wouldn't
directly address this problem).
Anyway, empirically, the output now looks the same as with hwdec
off.
Swapchain depth currently hard-coded to 3 (4 buffers).
As we now avoid redrawing on repeat frames (we simply requeue the same fb
again), this should give a nice performance boost when playing videos with a
lower FPS than the display FPS in video-sync=display-resample mode.
Presentation feedback has also been implemented to help counter the
significant amounts of jitter we would otherwise be seeing.
This allows stream_cb backends to implement blocking
behavior inside read_fn, and still get notified when the user
wants to cancel and stop playback.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This wasn't used at all in my tests, because it simply passed the
frame directly to libswsresample. (And, by the way, will always do
that, because s64 is so obscure literally NOTHING uses it except
a sample specifically created to test this code. Screw FFmpeg.)
This can be due to unsupported sample formats (see previous commits),
minor allocation failures, and similar things. For identifying the exact
cause it's buried too deep in abstractions. But most time it doesn't
happen anyway, since it's extremely rare that new audio formats are
added.
What an idiotic format. It makes no sense, and should have been
converted to S32 in the demuxer, rather than plague everyone with
another extremely obscure nonsense format. Why doesn't ffmpeg add S24
instead? That's an actually useful format.
May cause compilation failure with old FFmpeg or Libav libs, but I don't
care.
TARGET_OS_TV seems to always be defined, but set according to the
build configuration. This fixes all Apple configurations being
mis-identified as tvOS.
The completion function itself now parses --list-options on the first
tab press and caches the results. This does mean a slight delay on that
first tab press, but it will only do this if the argument being
completed looks like an option (i.e. starts with "-"), so there is never
a delay when just completing a file name. I've also put some effort into
making it reasonably fast; on my machine it's consistently under 100 ms,
more than half of which is mpv itself.
Installation of zsh completion is now done unconditionally because it's
nothing more than copying a file. If you really don't want it installed,
set zshdir to empty: `./waf configure --zshdir= ...`
Improvements in functionality compared to the old script:
* Produces the right results for mpv binaries other than the one it was
installed with (like a dev build for testing changes).
* Does not require running mpv at build time, so it won't cause
problems with cross compilation.
* Handles aliases.
* Slightly nicer handling of options that take comma-separated values
and/or sub-options: A space is now inserted at the end instead of a
comma, allowing you to immediately start typing the next argument,
but typing a comma will still remove the automatically added space,
and = and : will now do that too, so you can immediately add a
sub-option.
* More general/flexible handling of values for options that print their
possible values with --option=help. The code as is could handle quite
a few more options (*scale, demuxers, decoders, ...), but nobody
wants to maintain that list here so we'll just stick with what the
old completion script already did.
This allows to use drm hwaccels that require a hwdevice.
Tested with v4l2request hwaccel and cedrus driver on an allwinner device
running mpv with --vo=gpu --gpu-context=drm --hwdec=drm.
At first, this code used only 1 plane, so the compatibility stuff was
sufficient. But then use of planes 1 and 2 was added, without extending
the compatibility stuff.
I think I've seen a case recently where this broke the build and caused
users to apply invalid fixes, but I don't remember where.
It's possible that I didn't get all defines that are needed.
I think I was wrong about having to reset the stats when mpv stops
producing frames, eg. when it's paused. As long as the swapchain doesn't
underflow, last_queue_display_time will still be accurate, because the
next frame produced should still be presented one vsync after the
last one in the swapchain.
If the swapchain underflows (which is the common case for when mpv is
paused for more than 150ms,) the next predicted frame time should be in
the past. It should be fine to leave last_queue_display_time unset in
this case, since vo.c will use the current time instead, which is a
decent guess (though it doesn't take vsync phase into account.)
last_sync_refresh_count and last_sync_qpc_time should be kept on
swapchain underflow as well. Assuming the display refresh rate doesn't
change while mpv is paused, they'll only provide a more accurate guess
of the vsync duration when mpv starts playing again. Also,
vsync_duration_qpc never needs to get reset. It will get overwritten
immediately in most cases, and when it doesn't, it's still a better
guess of the vsync duration than nothing.
That's just a single one. It used to be more, when FFmpeg still required
using pointless accessors for tons of fields, which historically broke
compatibility with Libav. (I think I wrote the patch to deprecate that
crap and to allow direct access myself.)
There may be more exceptions, but I forgot about them. Another point is
that we don't really trust FFmpeg ABI stability, though.
In spdif mode, there are hacks that try to cut audio on frame boundaries
(blame spdif, which is a hack in itself). The "alignment" is used in a
bunch of places, but --end does not respect it. This leads to some audio
that can't be pushed because the alignment is off (I don't know why, not
do I care), which puts audio into an underrun state forever.
Fix this by discarding unusable extra samples if no new data can be
expected.
Fixes: #6935
The C11 situation is complicated. For example, MinGW doesn't seem to
have a full C11 implementation, but we pretty much rely on C11 atomics.
Regarding "#pragma once": they say it's not standard because of unsolved
(admittedly valid) issues. Btu still, fuck writing include guards, I
just can't be bothered with this crap.
(Does anyone even read this document?)
I think a popular libmpv application did exactly this: enabling advanced
control, and then receiving deadlocks. I didn't confirm it, though. In
any case, the API docs should avoid tricking users into making this easy
mistake.
The render API (vo_libmpv) had potential deadlock problems with
MPV_RENDER_PARAM_ADVANCED_CONTROL. This required vd-lavc-dr to be
enabled (the default). I never observed these deadlocks in the wild
(doesn't mean they didn't happen), although I could specifically provoke
them with some code changes.
The problem was mostly about DR (direct rendering, letting the video
decoder write to OpenGL buffer memory). Allocating/freeing a DR image
needs to be done on the OpenGL thread, even though _lots_ of threads are
involved with handling images. Freeing a DR image is a special case that
can happen any time. dr_helper.c does most of the evil magic of
achieving this. Unfortunately, there was a (sort of) circular lock
dependency: freeing an image while certain internal locks are held would
trigger the user's context update callback, which in turn would call
mpv_render_context_update(), which processed all pending free requests,
and then acquire an internal lock - which the caller might not release
until a further DR image could be freed.
"Solve" this by making freeing DR images asynchronous. This is slightly
risky, but actually not much. The DR images will be free'd eventually.
The biggest disadvantage is probably that debugging might get trickier.
Any solution to this problem will probably add images to free to some
sort of queue, and then process it later. I considered making this more
explicit (so there'd be a point where the caller forcibly waits for all
queued items to be free'd), but discarded these ideas as this probably
would only increase complexity.
Another consequence is that freeing DR images on the GL thread is not
synchronous anymore. Instead, it mpv_render_context_update() will do it
with a delay. This seems roundabout, but doesn't actually change
anything, and avoids additional code.
This also fixes that the render API required the render API user to
remain on the same thread, even though this wasn't documented. As such,
it was a bug. OpenGL essentially forces you to do all GL usage on a
single thread, but in theory the API user could for example move the GL
context to another thread.
The API bump is because I think you can't make enough noise about this.
Since we don't backport fixes to old versions, I'm specifically stating
that old versions are broken, and I'm supplying workarounds.
Internally, dr_helper_create() does not use pthread_self() anymore, thus
the vo.c change. I think it's better to make binding to the current
thread as explicit as possible.
Of course it's not sure that this fixes all deadlocks (probably not).
When the (float) bitrate is returned, it is implicitely converted to an
int64 value, merely discarding the fractional part.
However the bitrate of a CBR track can vary a bit due to timestamp
precision loss after clock conversion (this can affect MPEG-TS audio
tracks). So a bitrate like 191999.999... results in 191999 when
being returned - instead of 192000.
To fix this, apply proper rounding, as already done for the "old" case.
Hereby refactoring the "old" case to also use `llrint`.
libass had an API to configure this since 2013. mpv always used
ASS_FONTPROVIDER_AUTODETECT, because usually there's little reason to
use anything else. The intention of the now added option is to allow
users to disable use of system fonts.
I didn't consider it worth the trouble to add the coretext and
directwrite enum items from ASS_DefaultFontProvider. The "auto" choice
will have the same effect if they're available. Also, the part of the
code which defines the option does not necessarily have libass available
(it's still optional!), so defining all enum items as choices is icky. I
still added fontconfig, since that may be nice to emulate a nostalgic
2010 feeling of mpv freezing on fontconfig.
The option for OSD is even less useful. (But you get it for free, and
why pass up a chance to add yet another useless option?)
This is not quite what was requested in #6947, but as close as it gets.
in->eof is used as an indicator whether reading packets still makes
sense. (Without this, the prefetcher would obviously burn CPU by
retrying reading even though everything has been read.)
This was not reset properly after seeks were performed. It led to
getting stuck in at least one corner case: when enabling a track, the
demuxer would seek backwards to get new packets from the current
playback position ("refresh seeks"). But if playback was paused, and EOF
was previously reached, it would not try to read packers again due to
in->eof being false. There was not anything else that would make it
retry reading, so it was stuck in a weird underrun/buffering state.
Fixes: #6986
This was a leftover from commit b2752321 which fixed#6522 but after
the recent demux refactoring this fix is superseded by commit 0f6cda4ab.
Remove the redundant update call.
This detected the first packet demuxed after a seek as timestamp
discontinuity. Obviously this is non-sense. Since the OGG radio streams
for which this feature was introduced are normally unseekable, it's
simple to fix this: simply disable it (if in auto mode, the default) as
soon as a seek is performed. This code is never called if the stream is
considered unseekable, unless the user forced it.
There's still a chance this linearization is performed before a seek
happens. This will be a bit awkward, but no worse than without this
feature, since seeking with timestamp resets is inherently broken in
both mpv and libavformat.
Fixes: #6974
Fixes: 27fcd4d
since the loading order of rpaths is system wide lib path, dev tool path
and then bundle lib path it's possible for the xcode swift libs to be
incompatible with the libs the bundle was build with. this leads to
possible segfaults. if we distribute the bundle we don't want to load
the libs from the dev tools anyway.
in xcode 11 the dynamic swift libraries were moved to a separated
versioned swift folder, which can't be used for linking and only for
distribution. additional to the std dynamic swift lib folder the system
wide folder is needed for linking too.
First we shift the values up to the actual amount of bits in draw_ass,
so that they will be drawn correctly when using formats with more than
8 bpc. (draw_rgba is already correct w.r.t. RGB formats with 9 or more
bpc)
Then, in scale_sb_rgba, by setting the amount of bits per channel used
for planar RGB formats (formats are always planar at this point in
draw_bmp) to be the same as the source from 9 to 16 bpc (in effect all
the various GBRP formats) we manage to fit the special case that does
not require any conversion in chroma_up and chroma_down when handling
these formats (as long as the source itself is a planar format),
instead writing directly to the combined dst/src buffer. This in turn
works around a bug (incorrect colors) in libswscale when scaling
between GBRP formats with 9 or more bpc. Additionally this should be
more efficient, since we skip up- and down-conversion and temporary
buffers.
We default to EGL instead of GLX now, which means vdpau only works
if we explicitly specify that we want a GLX context, as vdpau lacks
interop for EGL.
Update the hwdec documentation to reflect this.
Concerns #6980.
This adds vsync reporting to the D3D11 backend using the presentation
feedback provided by DXGI, which is pretty similar to what's provided by
GLX_OML_sync_control in the GLX backend. In DirectX, PresentCount is the
SBC, PresentRefreshCount and SyncRefreshCount are kind of like the MSC
and SyncQPCTime is the UST.
Unlike GLX, the DXGI API makes it possible for PresentCount and
SyncQPCTime to refer to different physical vsyncs, in which case
PresentRefreshCount and SyncRefreshCount will be different. The code
supports this possibility, even though it's not clear whether it can
happen when using flip-model presentation. The docs say for flip-model
apps, PresentRefreshCount is equal to SyncRefreshCount "when the app
presents on every vsync," but on my hardware, they're always equal, even
when mpv misses a vsync. They can definitely be different in exclusive
fullscreen bitblt mode, though, which mpv doesn't support now, but might
support in future.
Another difference to GLX is that, at least on my hardware,
PresentRefreshCount and SyncRefreshCount always refer to the last
physical vsync on which mpv presented a frame, but glxGetSyncValues can
apparently return a MSC and UST from the most recent physical vsync,
even if mpv didn't present a new frame on it. This might result in
different behaviour between the two backends after dropped frames or
brief pauses.
Also note, the docs for the DXGI presentation feedback APIs are pretty
awful, even by Microsoft standards. In particular the docs for
DXGI_FRAME_STATISTICS are misleading (PresentCount really is the number
of times Present() has been called for that frame, not "the running
total count of times that an image was presented to the monitor since
the computer booted.")
For good documentation, try these:
https://docs.microsoft.com/en-us/windows/win32/direct3ddxgi/dxgi-flip-modelhttps://docs.microsoft.com/en-us/windows/win32/direct3d9/d3dpresentstatshttps://docs.microsoft.com/en-us/windows-hardware/drivers/ddi/content/d3dkmthk/ns-d3dkmthk-_d3dkmt_present_stats
(Yeah, the docs for the D3D9Ex and even the kernel-mode version of this
structure are better than the DXGI ones. It seems possible that they're
all rewordings of the same internal Microsoft docs, but whoever wrote
the DXGI one didn't really understand it.)
The question came up on how a client would figure out where
screenshot-directory saved its screenshots if it contained
mpv-specific expansions. This command should remedy the situation
by providing a way for the client to ask mpv to do an expansion.
Previously the options were tested by compiling a test program with the
option, and support was "detected" if the compilation didn't fail.
However, both gcc and clang only issue a warning on unknown warning option,
hence it never failed and all the warn options were detected as supported.
Now the tested option is used together with -Werror, which makes it fail
if it warns that the warning option is unknown.
Fixes clang unknown option warnings -Wno-format-truncation since 4a6b56f .
Certain mpv config options require wl->current_output to be created
before the video can actually start rendering. Just always create it
here if the current_output doesn't exist (the one exception being the
--fs option with no --fs-screen flag). Incidentally, this also fixes
--fs-screen not working on wayland.
This test requires libavutil headers but there is no avutil
dependency for it to use. So let's add one, and also reorder the
ffmpeg tests ahead of the video tests so that the avutil dependency
can be used.
../misc/linked_list.h:71:34: warning: the address of ‘e6’ will always
evaluate as ‘true’ [-Waddress]
No shit, e6 is on the stack. But the macro argument is also allowed to
be NULL. Add some dumb nonsense to shut up the useless warning. (It's
probably useful in other contexts though, so don't disable it
completely.)
Just use cmocka's function. It takes an epsilon argument, which we now
provide directly.
There's no assert_double_equal() in cmocka (and the float variant
actually forces a conversion to the float type), but fortunately we
didn't use it.
(At first I left this intentionally, because the temporarily disabled
stream ctrl code used it, but there's actually no reason to annoy
everyone with the warning.)
Use `private-code` instead of `code` here. Also this bumps up the
required wayland-client/wayland-cursor version to 1.15 which is still
pretty old anyway.
reinit_audio_filters_and_output() can fully shutdown the audio chain on
failure. Specifically, it will deallocate mpctx->ao_chain. The value of
that field was cached in ao_c. The code after the call did not account
that the audio chain can be shutdown, and used the stale ao_c value.
Fixes: #6808
During initial testing with US closed captions, ARIB captions,
timed text in MP4 or the specific external SRT files I tested with
there were no hints that this flag would be needed for seeking to
work.
Unfortunately, that result seems to have been incorrect.
Fixes#6970
If a file format supports PTS resets, get_current_pos_ratio calculates
the ratio using the stored filepos in the demuxer struct instead of a
standard query of the current time in the stream and its total length.
This seems like a reasonable way to avoid weird PTS values, but in
reality this just causes problems and results in inaccurate ratio
values that can affect other parts of the player (most notably the osc
seekbar).
For libavformat formats, demux->filepos is obtained from the
demux_lavf_fill_buffer function which is called on the next packet. The
problem is that there is a slight delay between packets and in some
cases, this delay can be relatively large. That means the obtained
demuxer->filepos value will be very inaccurate since it obtains the pos
from the end of the upcoming packet and not its actual current position.
This is especially noticeable at the very beginning of playback where
get_current_pos_ratio sometimes returns a value of well over 2% despite
less than a second passing in the stream. Another telltale sign is to
simply watch the osc seekbar as a stream progresses and observe how it
loads in staggered steps as every packet is decoded. In contrast, the
seekbar progresses smoothly on the playback of a format that does not
support PTS resets. The simple solution is to instead use the query of
the current time and length of a stream and calculate the ratio that
way.
get_current_pos_ratio will still fallback on using the byte stream
position if the previous queries fail. However, get_current_time will
be more accurate in the vast majority of cases and should be the
preferred method of calculating the position ratio.
ioctl(..., SNDCTL_DSP_CHANNELS, &nchannels) for not supported
nchannels does not return an error and instead set nchannels to
the default value.
Instead of failing with no audio, fallback to stereo.
To find the correct ICC profile X atom, the screen number was calculated
directly from the xrandr order of the screens.
But if a primary screen is set, it should be the first Xinerama screen,
even if it is not the first xrandr screen.
Calculate the the proper atom id for each screen.
On wayland the cursor has to be configured each time the pointer enters.
Currently if the window (re)gains the focus, the pointer is not hidden,
even when configured. After the mouse has been moved the pointer hides
correctly.
https://wayland.freedesktop.org/docs/html/apa.html#protocol-spec-wl_pointer:
wl_pointer::enter - enter event
...
When a seat's focus enters a surface, the pointer image is undefined
and a client should respond to this event by setting an appropriate
pointer image with the set_cursor request.
Fixes#6185.
Signed-off-by: Thomas Weißschuh <thomas@t-8ch.de>
Previously, the only mouse buttons supported in wayland were left,
right, and middle click. This adds the thumb back/forward buttons as
valid bindings. Also it removes the old, default behavior of always
sending a right click if an unrecognized mouse button is clicked.
In a related but different fix, the magnitude of an axis event in
wayland is not important to mpv since it internally handles all scaling.
The only thing we care about is getting the sign when the event occurs.
This change adds a version of `mpv_command` that also returns a result.
The main rationale behind this is `mpv_command_node` requires defining
multiple structs before you can even use it, which results in a pretty
painful to use interface just to get the result from a command.
There isn't really a good name for this function, so I'm open to
suggestions on a better name for it.
Basically, both the license of the file and the preferred license of the
project (LGPLv2.1+) counts. I'm doing that so that files with more
liberal licenses don't get infected by LGPL, but allow copy & pasting to
LGPL source files without jumping through lawyer bullshit hoops.
Mention this in Copyright too.
Most files are LGPL anyway by now. contribute.md still requires
contributors to provide changes under LGPL, which should cover changes
to files that are LGPL, as well as GPL files.
This flag makes mpv continue using the PulseAudio driver even if the
sink is suspended.
This can be useful if JACK is running with PulseAudio in bridge mode and
the sink-input assigned to mpv is the one JACK controls, thus being
suspended.
By forcing mpv to still use PulseAudio in this case, the user can now
adjust the sink to an unsuspended one.
Replace the "+" with "/". The "+" was supposed to imply that the cache
is the sum of the time (demuxer cache) and the size in bytes (stream
cache). We could not provide something nicer, because we had no idea how
many seconds of media was buffered in the stream cache.
Now the stream cache is done, and both the duration and byte size show
the amount buffered in the demuxer cache. Hopefully "/" is better to
imply this properly. Update the manpage explanations too.
Check if eglGetPlatformDisplayEXT is available and try to
use it to obtain the display connection. Fall back to eglGetDisplay
if eglGetPlatformDisplayEXT is not available or failing.
From PR #5992
When update_prop() successfully fetched a changed property value, it
set prop->changed to true to indicate the success.
mark_property_changed() also always set
prop->changed to true and additionally prop->need_new_value to true
This is the case since 6ac0ef78
If the observed property changes every frame and then due to timing
the next mark_property_changed() is called before
gen_property_change_event() and therefore directly after update_prop(),
prop->need_new_value was again true and indicated that a new value
has to be retrieved with update_prop(). As a result the event for the
last successful update_prop() was never triggered. This meant that
a property change event were never generated for frame-based properties
only for properties that were observed with MPV_FORMAT_NONE or when the
timing was different and gen_property_change_event() was called after
update_prop().
To fix this, mark_property_change() and update_prop() should not use the
same flag to indicate different things and therefore a new flag for
successfully update a property is introduced. But with the now decoupled property
changed and updated the need_new_value flag is redundant and removed completely.
Fixes#4195
With the stream cache gone, this function had almost no use anymore
(other than opening the stream). Improve this by triggering demuxer
cache readahead.
This enables all streams. At this point we can't know yet what streams
the user's options would select (at least not without great additional
effort). Generally this is what you want, and the stream cache would
have read the same amount of data.
In addition, this will work much better for files that e.g. need to seek
to the end when opening (typically mp4, and mkv files produced by newer
mkvmerge versions).
Remove the deselection call from add_stream_track(). This should be
fine, as streams normally start out as deselected anyway. In the
prefetch case, some code in play_current_file() actually deselects it.
Streams that appear during demuxing are disabled by default, so this
doesn't break this logic either.
Fixes: #6753
Before this commit, EDL or CUE files did not properly enable the cache
if they were on "slow" media (stream->streaming==true). This happened
because the stream is unset for demux_timeline, so the streaming flag
could not be queried anymore.
Fix this by adding this flag to struct demuxer, and propagate it exactly
like the is_network flag. is_network is not used for checking the cache
options anymore, and its main function seems to be something else.
Normal http streams set the streaming flag already.
This should fix#6958.
This is mostly for the case when mpv_render_context_free() is called
while video is going on. This is supposed to gracefully stop video and
deinitialize everything properly. (I feel like it would put too much on
the API user to require that video is stopped before calling this
function. Whether video is running or not is a fairly highlevel thing,
and the API user could not do it in a race-free way.)
One problem was that unit() accessed ctx after ctx->in_use was set to
false. The update(ctx) call was basically a racy use-after-free. It
needed that call to wake up the mpv_render_context_free() loop that
waited for VO uninit. Fix this by triggering the wakeup inside the lock,
and then doing "barrier" locking in mpv_render_context_free().
Another problem was that the wait loop didn't really wait properly. IT
seems the had_kill_update field was a botched attempt to do that. It's
indeed quite hairy to do that with update(). Instead make use of the
dispatch queue (infinite timeout, using mp_dispatch_interrupt()), which
handles the problem of having to wait both for dispatch queue updates
and VO uninit at the same time.
This is because dr_helper.c calls pthread_self(). It's used to avoid
deadlocks. This was not a problem internal to mpv (dr_helper.c was first
created for vo_gpu.c), but it accidentally leaked as an unintended
consequence of an implementation detail to libmpv.
This problem existed for MPV_RENDER_PARAM_ADVANCED_CONTROL users, ever
since it was introduced.
Maybe this could be done differently, but it's certainly very tricky.
the pthread_t is used to avoid deadlocks when the caller is on the same
thread as the one which needs to handle the calls.
The critical code is in free_dr_buffer_on_dr_thread(). It's called when
a DR image is free'd. Freeing a DR image requires access to the GL
context, i.e. it just has to run on the "GL" thread. In libmpv's case,
this is done by calling the API user's wakeup call, and the user will
eventually call mpv_render_context_update() on the "GL" thread, which in
turn calls mp_dispatch_queue_process() on the dispatch queue that was
passed to dr_helper_create(), which then calls av_buffer_unref(), which
calls gl_video_dr_free_buffer(). (God who came up with this rube
goldberg shit.)
The above case will typically happen when e.g. vd_lavc.c (internal mpv
thread) frees images. Allocation works similarly; deallocation is just
trickier because calls to free images are everywhere.
Now consider if mpv_render_context_render() releases a DR image. This
function is always called on the "GL" thread. Going through the dispatch
queue would obviously deadlock, because according to the render API
rules, the user's wakeup callback must not mpv_render_context_update(),
instead it has to signal the "GL" thread, which must wait until
mpv_render_context_render() returns. To avoid this deadlock, dr_helper.c
checks whether the calling thread is the "GL" thread, and if so, allows
a reentrant call (basically gl_video_dr_free_buffer() is called
directly).
While with GL, you usually will stay on the same thread, API users were
in theory allowed to e.g. move the GL context to a different thread. But
this dr_helper issue means this is not possible. Moreover, other APIs
will not have the same thread-locality problems as GL.
FUCK THIS SHIT
In theory, libmpv could provide an API to "move" the context to a
separate thread, but let's not start with even more broken crap like
this. I'm not sure yet whether there is an easy solution. Maybe all
mpv_render() function could be guarded by entry/exit functions, which
set/unset a flag that dr_helper.c should use reentrant calls.
libmpv users which do not set MPV_RENDER_PARAM_ADVANCED_CONTROL were
never affected.
Preparation for the next commit. Until now, it was only needed if DR was
involved. One reason for not always creating it was that you normally
must not use it if advanced_control is not enabled. This is why e.g.
VOCTRL_SCREENSHOT now checks for that variable; it still can't use
ctx->dispatch if the render API user did not enable it.
render api needs to wait for vo to be destroyed before frees the context.
The purpose of kill_cb is to wake up render api after vo is destroyed,
but uninit did that before kill_cb, so kill_cb tries using the freed
memory. Remove kill_cb to fix the issue as uninit is able to do the
work.
Equalizer control was redone in 03cf150ff3 (over 2 years
ago). Ever since, the equalizer control structs and the GET voctrl have
been unused. Only the SET voctrl is still used as notification mechanism
(actually a bad hack to avoid some further option change handling
complexity).
Remove the unused parts.
I was assuming posix_memalign was the most portable function to use, but
MinGW does not provide it for some reason. Switch to C11 aligned_alloc()
which someone suggested was provided by MinGW (but actually isn't,
someone probably confused it with the incompatible _aligned_malloc),
and add a configure check.
Even though it turned out that MinGW doesn't provide it, the function
is slightly more elegant than posix_memalign(), so stay with it.
This field is documented as internal, so an API user should not
access it. However, this is the only way to get some read statistics
without replacing FFmpeg's entire HLS demuxer. (Using custom I/O as
workaround doesn't work: the HLS code uses some weird internal APIs
that cannot be provided by FFmpeg API users; I even made the author
of the relevant patch to provide a public API, but which was shot
down by another FFmpeg developer. So I take this as my right to
access this field.)
Mention this explicitly, as it affects ABI and API compatibility, and
I don't want that anyone claims this was a "mistake". Add some
explanations.
In theory, a 0 size allocation could have made it memset() on a NULL
pointer (with a non-0 size, which makes it crash in addition to
theoretical UB).
This should never happen, since even packets with size 0 should have an
associated allocation, as FFmpeg currently does. But avoiding this makes
the API slightly more orthogonal and less tricky, I guess.
This reverts commit 95636c65e7.
This change shouldn't be needed, and in fact it's wrong. The FFmpeg API
function could do anything it wants with the packet, including changing
the packet data pointer. Likewise, it's not guaranteed that the
referenced packet's fields mirror the current state of the mpv packet
struct (the AVPacket is only kept for the AVBuffer and the side data
stuff).
skip-logo.lua is just what I wanted to have. Explanations are on the top
of that file. As usual, all documentation threatens to remove this stuff
all the time, since this stuff is just for me, and unlike a normal user
I can afford the luxuary of hacking the shit directly into the player.
vf_fingerprint is needed to support this script. It needs to scale down
video frames as part of its operation. For that, it uses zimg. zimg is
much faster than libswscale and generates more correct output. (The
filter includes a runtime fallback, but it doesn't even work because
libswscale fucks up and can't do YUV->Gray with range adjustment.)
Note on the algorithm: seems almost too simple, but was suggested to me.
It seems to be pretty effective, although long time experience with
false positives is missing. At first I wanted to use dHash [1][2], which
is also pretty simple and effective, but might actually be worse than
the implemented mechanism. dHash has the advantage that the fingerprint
is smaller. But exact matching is too unreliable, and you'd still need
to determine the number of different bits for fuzzier comparison. So
there wasn't really a reason to use it.
[1] https://pypi.org/project/dhash/
[2] http://www.hackerfactor.com/blog/index.php?/archives/529-Kind-of-Like-That.html
Until now they weren't observable and never reported any updates. Apply
a shitty hack to make them mostly-observable. It relies on the "idle"
event, which is basically triggered on every frame displayed, or
similar. This can lead to property change notifications not being sent
quickly enough.
The cleaner solution would be adding a notification mechanisms from
filters, but I'm too lazy for that.
For simplicity, these properties usually query the metadata from the
filter twice, even if it's not technically needed at all. The reason for
this is mostly the horrible (and legacy) sub-path access (which is why
tag_property() is so complex).
But for simple cases, we can easily avoid double querying, so do that.
The benefit is performance (well, won't matter), and supporting filters
that reset information on query (for later).
Generally, using x86 SIMD efficiently (or crash-free) requires aligning
all data on boundaries of 16, 32, or 64 (depending on instruction set
used). 64 bytes is needed or AVX-512, 32 for old AVX, 16 for SSE. Both
FFmpeg and zimg usually require aligned data for this reason.
FFmpeg is very unclear about alignment. Yes, it requires you to align
data pointers and strides. No, it doesn't tell you how much, except
sometimes (libavcodec has a legacy-looking avcodec_align_dimensions2()
API function, that requires a heavy-weight AVCodecContext as argument).
Sometimes, FFmpeg will take a shit on YOUR and ITS OWN alignment. For
example, vf_crop will randomly reduce alignment of data pointers,
depending on the crop parameters. On the other hand, some libavfilter
filters or libavcodec encoders may randomly crash if they get the wrong
alignment. I have no idea how this thing works at all.
FFmpeg usually doesn't seem to signal alignment internal anywhere, and
usually leaves it to av_malloc() etc. to allocate with proper alignment.
libavutil/mem.c currently has a ALIGN define, which is set to 64 if
FFmpeg is built with AVX-512 support, or as low as 16 if built without
any AVX support. The really funny thing is that a normal FFmpeg build
will e.g. align tiny string allocations to 64 bytes, even if the machine
does not support AVX at all.
For zimg use (in a later commit), we also want guaranteed alignment.
Modern x86 should actually not be much slower at unaligned accesses, but
that doesn't help. zimg's dumb intrinsic code apparently randomly
chooses between aligned or unaligned accesses (depending on compiler, I
guess), and on some CPUs these can even cause crashes. So just treat the
requirement to align as a fact of life.
All this means that we should probably make sure our own allocations are
64 bit aligned. This still doesn't guarantee alignment in all cases, but
it's slightly better than before.
This also makes me wonder whether we should always override libavcodec's
buffer pool, just so we have a guaranteed alignment. Currently, we only
do that if --vd-lavc-dr is used (and if that actually works). On the
other hand, it always uses DR on my machine, so who cares.
This matters when talloc allocations set destructors. Before this
commit, destructors were called in the same order as they were added to
the parent allocations. Now it happens in reverse order.
I think this makes more sense. It's reasonable to assume that an
allocation that was added later may depend on any of the previous
allocations, so later additions should be destroyed first. (Of course
other orders are entirely possible too.)
Hopefully this doesn't fix or break anything, but I can't be sure (about
either of those). It's risky. (Then why do it?)
The destructor of a parent allocation is called before its children. It
makes sense and must stay this way, because in most cases, the
destructor wants to access the children.
This is a reason why I don't really like talloc (it wasn't my idea to
use talloc, is my excuse). Quite possible that destructors should be
removed from talloc entirely. Actually, this project should probably be
rewritten in Rust (or a better language), but that would be even more of
a pain; also, I think this is just the right level of suffering and
punishment.
It seems using multiple prefixes for an option isn't supported out of
laziness (and shouldn't, because what the fuck). So assert() on this.
(Unfortunately this prefix nonsense is still needed. Especially AO and
VO options use this through the options_prefix field.)
A dumb thing that the cursed property-option bridge accidentally did.
Normal deprecated options on the other hand are fine in the property
list, because they're wanted for compatibility.
A previous commit changed m_config so that it always creates the shadow
thing, and the function's only remaining purpose was to initialize
mpv_global. It makes much more sense to do that at the caller, and it's
only 1 line of code too.
m_config has a m_config_option array, that is used for all option
access. The code maintaining shadow copies also tried to make use of it,
and did so by "cleverly" assigning each m_sub_options run a slice of
that array. But actually it's much simpler to, you know, directly access
the damn options.
This helps separation m_config and the general option code slightly.
Still seems to work after a superficial test, good enough.
This is good because a private thing is not so public anymore, and it's
also preparation for further changes.
Some tricky memory management issues: m_config_data (i.e. config->data)
now depends on m_config_shadow, instead of m_config. In particular,
free_option_data() accesses the m_config_shadow.groups array. Obviously
it must be freed before m_config_shadow.
Unused now. The old stream cache used it, but it was removed.
On a side note, the demuxer cache uses mp_mkostemps(). It looks like our
Windows open() emulation handles this correctly by using CREATE_NEW, so
no functionality gets lost by the "new" approach. On the other hand, the
demuxer cache does not set FILE_FLAG_DELETE_ON_CLOSE, but instead tries
to delete the file after opening (POSIX style), which probably won't
work on Windows. But I'm not sure how to make it use the DELETE_ON_CLOSE
flag, so whatever.
Move the comments documenting exported functions to the header. It looks
like the header is the preferred place for that (although I don't really
appreciate headers where you lose the overview because of all the
documentation comments). Add comments to some undocumented prototypes.
This was one of those "shouldn't exist" type of functions that could
access internals that were supposed to be isolated away, but some code
needed to access it anyway.
It looks like the last use of it went away in 2016, shortly after it was
introduced.
Dear diary,
today I fixed a shitty bug that was all my fault because I made a
horrible mess. (Except it was a horrible mess before I even touched
this shit, but let's not blame others.)
Sometimes, updates to VO option that control video sizing (like panscan)
didn't update the screen correctly. They were delayed until the next
option change or so.
It turns out that if the option update happens at the "same" time as a
VOCTRL, update_opts() doesn't actually notify the vo_driver of the
change. This in turn happened because run_control() called
m_config_cache_update(). The latter function returns true if the options
changed since the last call, and update_opts() also calls it (on the
same config cache) for the same purpose. The update_opts() call, which
is triggered by a third mechanism, comes later, but the cache update
call will return false (as it should). Basically, given the config API,
you can't act differently on multiple update calls and expect it to
work. The skipped handling in update_opts() meant that the notification
required to apply the changed option wasn't run.
Fix this by simply calling update_opts() directly instead. Now there's
only 1 m_config_cache_update() call on this specific instance. Fix the
call in run_reconfig() too, so the previous sentence isn't a lie (but it
probably doesn't make a difference in practice due to certain details).
I'm not sure how I even ran into this sort-of race condition. The VOCTRL
that messed up the option update was VOCTRL_UPDATE_PLAYBACK_STATE, which
happens semi-regularly.
Why this config cache shit and all the other shit? Rediscovering this
crap wasn't pleasant. It's a bunch of hacks that became necessary when
the ancient MPlayer architecture made it hard to move the VO to a
separate thread.
All the VO code typically accesses vo->opts (whose fields all used to be
global variables in MPlayer). The frontend changes these on user input.
Putting locking around all the options would be a nightmare, and keeping
a copy of the options in the thread was much simpler. You need a way to
propagate option changes, notify the thread, and update the local copy
too. And the result of these thoughts was the config cache mechanism.
In this specific case, the relevant cache update call in update_opts()
triggers a VOCTRL_SET_PANSCAN to the VO driver, which isn't related to
its former function anymore. Instead, it causes the VO driver to update
the video sizing/placing options, which the generic VO code can't do.
(Mostly because the VO driver includes the windowing stuff and is
responsible for resizing etc. itself.)
VOCTRLs sent by the frontend are even worse. MPlayer had no real runtime
option change mechanism. Some options were vaguely duplicated by
properties, so you could effectively change those options at runtime.
Each of these options had its own VOCTRL, which still exist today, e.g.
VOCTRL_FULLSCREEN, or VOCTRL_ONTOP. I tried to make all options runtime
changeable, and to unify properties with options. But I couldn't be
bothered with updating all VO drivers to listen to option changes
directly, because that would be pretty tedious. So the property code is
still all there and sends the old VOCTRLs. But of course you need to
sync up the options, which is why the run_control() code did that.
(Unrelated: VO_EVENT_FULLSCREEN_STATE is the worst shithack of them all.
Currently, only the frontend can actually write to options (for awful
reasons), so if the fullscreen state changes due to outside interaction,
the VO driver can't update the corresponding option fields. So the VO
notifies the frontend with said VO_EVENT_, and the frontend then sends
VOCTRL_GET_FULLSCREEN, and updates the global copy of the option with
the value returned by that. I still like to think the situation is not
that bad considering the monstrous effort of converting single-threaded
code that had hundreds of options in global variables to multi-threaded
code with no global variables at all.)
Helper for the ab-loop-dump-cache command, see manpage additions.
This is kind of shit. Not only is this a very "special" feature, but it
also vomits more messy code into the big and already bloated demux.c,
and the implementation is sort of duplicated with the dump-cache code.
(Except it's different.) In addition, the results sort of depend what a
video player would do with the dump-cache output, or what the user wants
(for example, a user might be more interested in the range of output
audio, instead of the video).
But hey, I don't actually need to justify it. I'm only justifying it for
fun.
This is the muxer used by all 3 stream recording features (why are there
so many?). It tried hard to avoid writing broken files. In particular,
it buffered packets until it new there was a keyframe packet (which, in
mpv's/FFmpeg's definition, mean seek points from which decoding can
resume), or final EOF. The danger that was probably considered here was
that due to video frame reordering, not muxing some trailing, missing
packets of a keyframe range could lead to broken decoding or skipped
frames, so better discard packets belonging to an incomplete range.
Sounds like a good idea so far.
Unfortunately, this will drop an entire keyframe range even if the
current packet run is complete and mp_recorder_mark_discontinuity() is
called, simply because recorder.c can not know that the next packet
would have been a keyframe.
It seems better to mux all packets to avoid losing valid data, even if
it means that sometimes packets/frames will be missing from the file. It
benefits especially the dump-cache command, which will call the function
to signal a discontinuity after every range. Before this commit, it
discarded the last packets, even if they were perfectly fine.
(An alternative solution for dump-cache would have been a second
discontinuity marker function, that communicates that the current packet
range is complete. But this commit's solution is simpler and overall
more robust, at the danger of producing more semi-broken files.)
This may make some of the complex buffering/waiting logic in recorder.c
pointless.
Untested (in this final form).
But don't tell the reader which those APIs are. Hope the user will just
search for "async" in the Lua section (lua.rst). But of course, nobody
will ever care about anything related to this.
That's right, and it's probably not the end of it. I'll just claim that
I have no idea how to create a proper user interface for this, so I'm
creating multiple partially-orthogonal, of which some may work better in
each of its special use cases.
Until now, there was --record-file. You get relatively good control
about what is muxed, and it can use the cache. But it sucks that it's
bound to playback. If you pause while it's set, muxing stops. If you
seek while it's set, the output will be sort-of trashed, and that's by
design.
Then --stream-record was added. This is a bit better (especially for
live streams), but you can't really control well when muxing stops or
ends. In particular, it can't use the cache (it just dumps whatever the
underlying demuxer returns).
Today, the idea is that the user should just be able to select a time
range to dump to a file, and it should not affected by the user seeking
around in the cache. In addition, the stream may still be running, so
there's some need to continue dumping, even if it's redundant to
--stream-record.
One notable thing is that it uses the async command shit. Not sure
whether this is a good idea. Maybe not, but whatever. Also, a user can
always use the "async" prefix to pretend it doesn't.
Much of this was barely tested (especially the reinterleaving crap),
let's just hope it mostly works. I'm sure you can tolerate the one or
other crash?
The screenshot command has this weird behavior that it shows messages
both on terminal and OSD by default, but that a command prefix can be
used to disable the OSD message.
Move this mechanism to common code, and make this available to other
commands too (although as of this commit only the screenshot commands
use it).
This gets rid of the weird screenshot_ctx.osd field too, which was sort
of set on a command, and sometimes inconsistently restored after the
command.
It makes some slight sense and helps with one of the following commits.
Also rename that other function to make it sound less similar to
find_seek_target().
Always set max_bytes_bw to 0 if seekable cache is disabled, instead at
the place of its use. This is the only use of it, so the commit should
not change any behavior.
(Alternatively, this could drop the max_bytes_bw variable, use the
option directly, and keep the old code that resets it on use of the
cache is disabled.)
Until now, the following could happen: if you set a 1GB forward cache,
and a 1GB backward cache, and you opened a 2GB file, it would prune away
the data cached at the start as playback progressed past the 50% mark.
With this commit, nothing gets pruned, because the total memory usage
will still be 2GB, which equals the total allowed memory usage of 1GB +
1GB.
There are no explicit buffers (every packet is malloc'ed and put into a
linked list), so it all comes down to buffer size computations. Both
reader and prune code use these sizes to decide whether a new packet
should be read / an old packet discarded. So just add the remaining free
"space" from the forward buffer to the available backward buffer. Still
respect if the back buffer is set to 0 (e.g. unseekable cache where it
doesn't make sense to keep old packets).
We need to make sure that the forward buffer can always append, as long
as the forward buffer doesn't exceed the set size, even if the back
buffer "borrows" free space from it. For this reason, always keep 1 byte
free, which is enough to allow it to read a new packet. Also, it's now
necessary to call pruning when adding a packet, to get back "borrowed"
space that may need to be free'd up after a packet has been added.
I refrained from doing the same for forward caching (making forward
cache use unused backward cache). This would work, but has a
disadvantage. Assume playback starts paused. Demuxing will stop once the
total allowed low total cache size is reached. When unpausing, the
forward buffer will slowly move to the back buffer. That alone will not
change the total buffer size, so demuxing remains stopped. Playback
would need to pass over data of the size of the back buffer until
demuxing resume; consider this unacceptable. Live playback would break
(or rather, would not resume in unintuitive ways), even normal streaming
may break if the server invalidates the URL due to inactivity. As an
alternative implementation, you could prune the back buffer immediately,
so the forward buffer can grow, but then the back buffer would never
grow. Also makes no sense.
As far as the user interface is concerned, the idea is that the limits
on their own aren't really meaningful, the purpose is merely to vaguely
restrict the cache memory usage. There could be just a single option to
set the total allowed memory usage, but the separate backward cache
controls the default ratio of backward/forward cache sizes. From that
perspective, it doesn't matter if the backward cache uses more of the
total buffer than assigned, if the forward buffer is complete.
The last_eof field is the last known EOF state from the underlying
demuxer. Normally, seeks reset it, because obviously if seek back into
the middle of the file, you don't want last_eof to have a "wrong" value
for a short time window (until a packet is read, which would reset the
field to its correct value).
This shouldn't happen during cache seeks, because you don't touch the
underlying demuxer state.
At first, I made this change because some other work in progress
required it. It turned out that it was unnecessary, but keep the change
anyway, since it's still correct and makes the logic cleaner.
m_geometry_apply() will read and modify the dummy variable. It's not
actually used for anything, but valgrind will still warn against
uninitialized data. I'm not sure whether this was UB, but in any case
it's annoying when running valgrind.
Determining how much memory something uses is very hard, especially in
high level code (yes we call code using malloc high level). There's no
way to get an exact amount, especially since the malloc arena is shared
with the entire process anyway. So the demuxer packet cache tries to get
by with an estimate using a number of rough guesses.
It seems this wasn't quite good. In some ways, it was too optimistic, in
others it seemed to account for too much data. Try to get it closer to
what malloc and ta probably do. In particular, talloc adds some
singificant overhead (using talloc for mass-data was a mistake, and it's
even my fault). The result appears to match better with measured memory
usage. This is still extremely dependent on malloc implementation and so
on.
The effect is that you may need to adjust the demuxer cache limits to
cache as much data as it did before this commit. In any case, seems to
be better for me.
If the disk cache is used, the AVPacket is not used anymore and is
completely deallocated when the packet is written to disk. As a minor
bug, the AVPacket allocation itself was not freed (although it wasn't a
memory leak, since talloc still automatically freed it when the entire
demux_packet was freed). For very large caches, this could easily add up
to over hundred MB, so actually free the unneeded allocation.
--hwdec=auto-copy was preferring vdpau over vaapi. In the HEVC 10 bit
case, this also led to hardware decoding not being enabled. (Probably
because the probing can't start over after enabling hw decoding fails at
runtime, or something like that.)
Possible that this subtly breaks on some setups. You can't always win.
During probing on a system with AMD GPU, mpv used to output the
following messages if hardware decoding was enabled:
[ffmpeg] AVHWFramesContext: Failed to create surface: 2 (resource
allocation failed).
[ffmpeg] AVHWFramesContext: Unable to allocate a surface from internal
buffer pool.
This commit removed the message, with hopefully no other side effects.
Long explanations follow, better don't read them, it's just tedious
drivel about the details. People should learn to write concise commit
messages, not drone on and on endlessly all while they have no fucking
point.
The code probes supported hardware pixel format, and checks whether they
can be mapped as textures. av_hwdevice_get_hwframe_constraints() returns
a list of hardware pixel formats in the valid_sw_formats field (the "sw"
means software, but they're still hardware pixel formats, makes sense).
This contained the format yuv420p, even though this is not a valid
hardware format. Trying to create a surface of this type results in VA
surface creation failure, upon which FFmpeg prints the error messages
above. We'd be fine with this, except FFmpeg has a global log callback,
and there's no way to suppress these messages without creating other
issues.
It turns out that FFmpeg's vaapi implementation returns all formats from
vaQueryImageFormats() if no "hwconfig" is provided. This list includes
yuv420p, which is probably supported for surface upload/download, but
not as native format. Following FFmpeg's logic, it should not appear in
the valid_sw_formats list, because formats for transfers are returned by
another roundabout API.
Idiotically, there doesn't seem to be any vaapi call that determines
whether a format is a valid surface format. All mechanisms to do this
are bound to a VAConfigID (= video codec or video processor), all while
the actual surface creation API strangely does not take a VAConfigID (a
big WTF).
Also, calling the vaCreateSurfaces() API ourselves for probing is out of
the question, because that functions is utterly and idiotically complex.
Look at the FFmpeg code and how much effort it requires to setup a
complete set of attributes - we can't duplicate this.
So the only way left to do this is the most idiotic and tedious way:
enumerating all VAProfile (and VAEntrypoints) to create all possible
VAConfigIDs. Each of the VAConfigIDs is associated with a list of
formats, which FFmpeg can return (by passing the ID along with the
"hwconfig"), and which is probed separately.
Note that VAConfigID actually refers to a dynamic instance of something,
and creating a VAConfigID takes not only the VAProfile and the
VAEntrypoint, but also an arbitrary attribute array. In theory, this
means our attempt to get to know all possible configurations cannot
work, but in practice this attribute array seems to be pointless for
decoding and video processing, and FFmpeg doesn't use it (though the
encoding path does use it). This probably just makes it _barely_ OK to
do it this way.
Could we discard all this probing shit, and somehow do it another way?
Probably not. The EGL API for mapping surfaces doesn't even seem to
provide a way to enumerate supported formats, we may not even know
whether DRM/dmabuf interop is actually supported (AFAIR the EGL
extensions are present even if they don't work), nor do we know whether
the VAAPI driver supports this interop (not sure). So actually trying is
the only way.
Further, mpv initializes the decoder on a another thread, where you
can't just access OpenGL state. This suckage is mostly to be blamed on
OpenGL itself and its crazy thread boundedness. In theory, this could be
done anyway (see how software decoding "direct rendering" tries to get
around this). But to make it worse, the decoder never cares about the
list of supported formats determined by this code; instead,
f_autoconvert.c tries to deal with it and insert a video processor
(well, good luck with this crap, I bet it doesn't even work). So this
whole endeavor might be pointless, other than the fact that failed
probing can disable use of vaapi (which is correct and necessary). But
if you have a shovel, you don't use it to smash the flat end on the heap
of shit that's piled up before you, or do you?
While this method probably works, it's still orgasmically tedious. It
was tedious before: we had to create a real surface, create a GL
texture, map the surface with it, then destroy everything again. But the
added code is tedious on its own. Highlights include the need to malloc
a FFmpeg struct just to pass a single damn integer, the need to
enumerate "entrypoints" for each VA profile, even though all profiles
have exactly 1 entrypoint, and the kind of obnoxious way how vaapi
requires you to preallocate arrays for returned things, even they could
for example reasonably be returned as immutable arrays or have some
other simpler API.
The main grand fuckup is of course that vaapi requires a VAConfigID to
query surface properties, but not for creating surfaces. This
awkwardness even affected the FFmpeg API design, which has a "hwconfig"
concept that is only used by vaapi (vaapi is only 1 out of 10 hardware
decoding APIs supported by the FFmpeg hwcontext stuff). Maybe I'm just
missing something. It's as if vaapi required setting radioactive shit on
fire. Look how clean the native D3D11 code is instead. (Even the ANGLE
code manages to avoid being this fucked up. Or the VDPAU code, despite
supporting multiple mapping methods.)
Another only barely related change is that the valid_sw_formats field
can be NULL, and the API explicitly documents this. Technically, the mpv
code was buggy for not checking this, although until now the FFmpeg
implementation so far could not return it when we still passed NULL for
the hwconfig parameter.
No functional changes, just preparation for the next commit. Split the
probing into multiple functions. Prepare for the yet unused possibility
to pass AVVAAPIHWConfig to probing. try_format_pixfmt() now assumes it
can be called multiple times with the same format, so it filters the
format.
The format probing is now something like O(n^2) for n formats, but n
will most likely remain something under 50 or so.
These macros explicitly honor MP_NOPTS_VALUE, instead of implicitly
relying on the fact that this value is the lowest allowed value.
In addition, this changes one case to use MP_NOPTS_VALUE instead of
INFINITY, also a cosmetic change.
Make most of the demuxer options runtime-changeable. This includes the
cache options and stream recording. The manpage documents some of the
possibly weird issues related to this.
In particular, the disk cache isn't shuffled around if the setting
changes at runtime.
Or at least I hope it's theoretical. This function is supposed to unset
any old listeners for the given cache, and the code works only if
there's at most 1. Add a defense break to avoid UB if there's more than
one, and add an assert() to check the assumption that there's at most
one.
The added comment is unrelated.
Although this is not useful in general, it makes --stream-record work
with a certain video streaming service by a large dystopian company.
In the general case, this fails because normal muxing can, quite
obviously, not handle the segmented metadata in the packets. (There
isn't even a file format which could handle these, except possibly mp4.)
On the other hand, ytdl merely uses timeline/EDL to emulate DASH
streaming (unfortunately), which does not use the segmented stuff, and
stream recording will actually work.
If this is used, the runtime expects that wmain() instead of main() is
defined. This caused me severe problems in a certain now irrelevant
case. I think it's a good idea to avoid this special case.
We can just use main() and call GetCommandLineW() instead. This function
returns a single string, so use CommandLineToArgvW() to split it, and
hope it has the same semantics. Should this ever return NULL, hope that
it leaves argc at 0.
Untested, I think.
_LARGEFILE_SOURCE and _LARGEFILE64_SOURCE are not required for 64 bit
off_t, only _FILE_OFFSET_BITS.
See somewhere on: https://wiki.musl-libc.org/faq.html
I didn't test this anywhere except 64 bit Linux. It's probably a good
idea to test on Windows and all Android versions.
I once created this because someone wanted to use vapoursynth without
the Python dependency. No idea if anyone ever actually used it. It's
sort of icky (it calls itself "lazy" to preempt complaints about how
much it sucks), and complicates the build process. Kill it.
It seems much more promising to have something like this:
https://github.com/vapoursynth/vapoursynth/issues/386
This would either solve the build distribution problem by relaxing the
Python dependency, and/or allow a Lua backend to be included without
pain.
This filter wasn't referenced anywhere and thus was dead code. It should
have been in the audio filter list in user_filters.c. This was intended
as compatibility wrapper (to avoid breaking old command lines and config
files), and has no real use. Apparently I forgot to add it to the filter
list (did I even test this shit?), and so it was rotting around for 1.5
years doing nothing (just like myself).
Note that users can just use the libavfilter provided filter to force
resampling, just that it has a different name and different options.
There's also af_format to force inserting auto conversion through the
internal f_swsresample filter.
Useless garbage.
This was once added to test whether vdpau presentation feedback could be
used. Results were always unsatisfactory, and now vdpau is dead.
The readahead time should be interesting for latency vs. underruns
(which idiot protocols like HLS suffer from).
The total byte usage is less interesting than I hoped; maybe the
frequency at which it samples should be reduced. (Kind of dumb - you
want high frequency for the readahead field, but much lower for byte
usage.)
Of course, the code was copy&pasted from the DS ratio/jitter stuff. Some
of the choices may not make any sense for the new code.
It prints "Unexpected end of file (no clusters found)" when opening a
webm init fragment. The warning is correct, but unwanted in this case.
Add tons of kludges to avoid it.
(Actually it prints that twice, for audio and video each.)
Also, suppress another warning about a seek head entry that points
exactly to the end of the file. This is a MATROSKA_ID_CUES, which is
harmless, and, very strangely, doesn't point at any cues when you
concatenate the init fragment with a media fragment. No idea what that
crap is supposed to be.
Retarded webshit streaming protocols (well, DASH) chop a stream into
small fragments, and move unchanging header parts to an "init" fragment
to save some bytes (in the case at hand about 300 bytes for each
fragment that is 100KB-200KB, sure was worth it, fucking idiots).
Since mpv uses an even more retarded hack to inefficiently emulate DASH
through EDL, it opens a new demuxer for every fragment. Thus the
fragment needs to be virtually concatenated with the init fragment. (To
be fair, I'm not sure whether the alternative, reusing the demuxer and
letting it see a stream of byte-wise concatenated fragmenmts, would
actually be saner.)
demux_lavc.c contained a hack for this. Unfortunately, a certain shitty
streaming site by an evil company, that will bestow dytopia upon us soon
enough, sometimes serves webm based DASH instead of the expected mp4
DASH. And for some reason, libavformat's mkv demuxer can't handle the
init fragment or rejects it for some reason. Since I'd rather eat
mushrooms grown in Chernobyl than debugging, hacking, or (god no)
contributing to FFmpeg, and since Chernobyl is so far away, make it work
with our builtin mkv demuxer instead.
This is not hard. We just need to copy the hack in demux_lavf.c to
demux_mkv.c. Since I'm not _that_ much of a dumbfuck to actually do
this, remove the shitty gross demux_lavf.c hack, and replace it by a
slightly less bad generic implementation (stream_concat.c from the
previous commit), and use it on all demuxers. Although this requires
much more code, this frees demux_lavf.c from a hack, and doesn't require
adding a duplicated one to demux_mkv.c, so to the naive eye this seems
to be a much better outcome.
Regarding the code, for some reason stream_memory_open() is never meant
to fail, while stream_concat_open() can in extremely obscure situations,
and (currently) not in this case, but we handle failure of it anyway.
Yep.
Timeline (demux_timeline, for EDL and mkv ordered chapters) are a mess,
because it's the only nested demuxer case. Part of the mess comes from
shared struct stream pointers. This makes no sense, because the wrapper
(demux_timeline) doesn't have any business setting it.
Try to lessen it by not passing down streams. Instead, pass down NULL.
This prevents unintended interference, and tightens the ownership rules.
Now a demuxer always owns its stream.
On the other hand, demuxer->stream can now be NULL. This was never the
case before, and consequently there will be new bugs. At least they will
be spotted, because they've been bugs before.
struct stream is also used to access stream properties (such as whether
something is considered a network stream). Most of these have been
mirrored in struct demuxer (because the frontend has been forbidden to
access struct stream because of threading). But during initialization
was still used, so introduce an awkward struct parent_stream_info, which
unifies these.
Commit e0419fb181 changed demux_is_network_cached() to use
demuxer->stream->streaming instead of demuxer->is_network. To enable
timeline stuff to use the cache anyway, change it so that both flags can
contribute to it. The stream NULL-check is obviously due to changes in
this commit.
Instead of having to rely on the protocol matching, make a function that
creates a stream from a stream_info_t directly. Instead of going through
a weird indirection with STREAM_CTRL, add a direct argument for non-text
arguments to the open callback. Instead of creating a weird dummy
mpv_global, just pass an existing one from all callers. (The latter one
is just an artifact from the past, where mpv_global wasn't available
everywhere.)
Actually I just wanted a function that creates a stream without any of
that bullshit. This goal was slightly missed, since you still need this
heavy "constructor" just to setup a shitty struct with some shitty
callbacks.
Raise it from 8KB to 512KB.
Do this because ytdl_hook.lua generated a 40KB EDL file (from 80KB
youtube-dl JSON output), and putting it into a .m3u file for easier
debugging failed due to the size limit.
I think I repeated this inline in some places (or maybe not), and some
experimental but discarded code used it. Add it anyway, maybe it'll be
useful. Or it'll give someone a chance to get a contribution into mpv by
removing an unused macro.
The previous commit broke audio playback (maybe this is what 4. was
about?). But it wasn't the fault of the commit; it just exposed
pre-existing issues.
If the packet queue search can't get all packets, it checked
queue->is_bof to see whether there could be previous packets. But
obviously, is_bof can be set, even if the search start packet wasn't the
first one.
This was especially observable with audio, because audio by default
needs preroll packets, and plays artifacts if they're missing.
Fix by using the BOF playback condition for this purpose too.
In theory, backward demuxing does way too much work by doing a full
cache seek every time you need to step back through a packet queue. In
theory, it would be exceedingly more efficient to just iterate backwards
through the queue, but this is not possible because I'm too stingy to
add 8 bytes per packet for backlinks. (In theory, you could probably
come up with some sort of deque, that'd allow efficient iteration into
any direction, but other requirements make this tricky, and I'm
currently too dumb/lazy to do this. For example, the queue can grow to
millions of elements, all while packet pointers need to stay valid.)
Another possibility is to "locally" seek the queue. This still has less
overhead than a full seek.
Also, it just so happens that, as a side effect, this avoids performing
range merging, which commit f4b0e7e942 "broke". That wasn't a bug at
all, but since range joining is relatively slow, avoiding it is good.
This is really just a coincidental side effect, I'm not even quite sure
why it happens this way.
There are 4 ugly things about this change:
1. To get a keyframe "before" a certain one, we recompute the target
PTS, and then subtract 0.001 as arbitrary number to "fudge" it. This
isn't the first place where this is done, and hey, it wasn't my damn
idea that MPlayer should use floats for timestamps. (At first, it even
used 32 bit timestamps.)
2. This is the first time reader_head is reset to an earlier position
outside of the seek code. This might cause conceptual problems since
this code is now "duplicated".
3. In theory, find_backward_restart_pos() needs to be run again after
the backstep. Normally, the seek code calls it explicitly. We could call
it right in the new code, but then the damn function would be recursive.
We could shuffle the code around to make it a loop, but even then
there'd be an offchance into running into an unexpected corner case (aka
subtle bug), where it would loop forever. To avoid refactoring the code
and having to think too hard about it, make it deferred - add some new
state and the check_backward_seek() function for this. Great, even more
subtle mutable state for this backwards shit.
4. I forgot this one, but I can assure you, it's bad.
Without doubt someone will have to clean up this slightly in the future
(or rip it out), but it won't be me.
This code tries to determine the "current" position, which is used as
base for the seek target when it needs to seek back more (the point is
to prevent seeking back too far). But compute_keyframe_times() almost
computes the same thing, so use that. Unfortunately needs a forward
declaration.
("Almost", because it differs in some details that should not really
matter.)
Subtitle packets with a timestamp before the seek target may overlap
with the seek target anyway. This is why this subtitle preroll crap
exists: it needs to return packets before the seek target to ensure that
the subtitle is displayed at the seek target.
This didn't always work. Maybe it's a regression, but it must have been
an old one. The breakage is triggered by heuristic that is to prevent
excessive queuing of packets in garbage files (this heuristic apparently
became immediately necessary when this preroll mechanism was
implemented).
If a video keyframe packet was found, but no audio packet yet, then
subtitle_preroll was set to 0, and since a_skip_to_keyframe was still 0,
the subtitle packet was discarded. The dumb thing is that subtitle and
video seeking is finished at this point, so the preroll crap should not
be applied at all.
Fix this by moving the preoll overflow code into the block that handles
preroll.
Normally I use the OSC like this: not at all, but have a key binding
that does "cycle osc" to show it. And in that case, I don't really want
it to overlap the damn video.
I could use the zoom/pan options to move the video out of the way, but
this is also sort of annoying. Likewise, you could write a script or so
which does this automatically if the OSC appears, but that's still
annoying, and computing values for these options such that the video is
moved correctly is tricky.
So I added a bunch of options that set explicit video borders (previous
commit), and a option for the OSC to use them (this commit).
Disabled by default, since I'm afraid this is too awkward and
unpolished, especially with OSC default settings.
I'm also using "osc-visibility=always". Effectively, making the OSC
appear will box the video, and making it disappear (by unloading
osc.lua) will restore the video back to normal.
Semantics a bit questionable. This is done for the OSC (next commit),
and a comment added the manpage explicitly states this. Meaning this is
probably garbage and needs to revisit when the OSC changes and/or
someone wants to use this margin feature for something else.
Not sure about the subtitle thing. It's imaginable that someone uses
these options to create empty borders for subtitles on the bottom, so
subtitles should be located there. On the other hand, this gives a
rather unpolished user experience when using the (later added) OSC
feature to not overlap with the video. There's not much of a point if
the OSC still overlaps the video. However, I'm too lazy to think about
this, so it stays like it is.
--video-margin-ratio-left=0.2 --video-margin-ratio-right=0.9 (added in
the the next commit) will set f_w to inf, resulting in some garbage
being propagated. Later, the OSD margins are computed from values before
various sanity clamping is applied, which makes libass suffer from
bullshit values.
I'm very sure it's OK and more correct to compute the OSD margins using
the later values, but I'm not sure about that.
demux_packet.next should not be used outside of demux.c, and in this
case it's a packet that was just passed to demux.c from the outside.
demux_packet.stream is already set by the demuxer, and this is assured
by the add_packet_locked() caller.
We don't care much about this case, because backward playback can fail
terribly without a good way to detect it, so this was fine.
However, this froze in certain situations. Reading from a subtitle file
for which backward demuxing failed could make it get stuck in
demux_read_packet_async() in unthreaded mode. (That we don't support
backwards subtitle decoding anyway doesn't matter for this.)
So aggressively disable backward demuxing to prevent worse in these
situations. The behavior will still be awful, because the frontend is
still in backwards playback mode, but at least it won't freeze.
Somewhat similar to the old --cache-file, except for the demuxer cache.
Instead of keeping packet data in memory, it's written to disk and read
back when needed.
The idea is to reduce main memory usage, while allowing fast seeking in
large cached network streams (especially live streams). Keeping the
packet metadata on disk would be rather hard (would use mmap or so, or
rewrite the entire demux.c packet queue handling), and since it's
relatively small, just keep it in memory.
Also for simplicity, the disk cache is append-only. If you're watching
really long livestreams, and need pruning, you're probably out of luck.
This still could be improved by trying to free unused blocks with
fallocate(), but since we're writing multiple streams in an interleaved
manner, this is slightly hard.
Some rather gross ugliness in packet.h: we want to store the file
position of the cached data somewhere, but on 32 bit architectures, we
don't have any usable 64 bit members for this, just the buf/len fields,
which add up to 64 bit - so the shitty union aliases this memory.
Error paths untested. Side data (the complicated part of trying to
serialize ffmpeg packets) untested.
Stream recording had to be adjusted. Some minor details change due to
this, but probably nothing important.
The change in attempt_range_joining() is because packets in cache
have no valid len field. It was a useful check (heuristically
finding broken cases), but not a necessary one.
Various other approaches were tried. It would be interesting to list
them and to mention the pros and cons, but I don't feel like it.
Supposed to follow the standard function.
The standard function is not standard, but a GNU extension. Adding some
ifdef mess is pointless too - it has no advantages other than having a
mess, and not spotting implementation bugs in the emulation due to
running it only on "obscure" platforms (like Windows, so most computers
actually, except the developer's platform).
There is mkstemp(), which at least is in POSIX 2008. But it's 100%
useless, except in some obscure cases: it doesn't set O_CLOEXEC, nor can
you pass it to it. Without O_CLOEXEC, we'd leak the temporary file to
all child processes. (The fact that the file, which is expected to reach
double or tripple digit GB sizes, will be deleted only once all
processes unreference the FD, makes this sort of a big deal. You could
ftruncate() it, but that doesn't fix all the other problems.)
Why did POSIX standardize mkstemp() and O_CLOEXEC apparently at the same
time, but provided no way to pass O_CLOEXEC to mkstemp()? With the
introduction of O_CLOEXEC, they acknowledged that there's a need to
atomically set the FD_CLOEXEC flag when creating file descriptors.
(FD_CLOEXEC was standard before that, but setting it with fcntl() is
racy.) You're much more likely to need a temp file that is CLOEXEC
rather than the opposite, and even if they were somehow opposed to
CLOEXEC by default (such as for compat. reasons), surely POSIX could
have standardized mkostemp() too or instead.
And then there's the fact that this whole O_CLOEXEC mess is stupid.
Surely there would have been a better way to handle this, instead of
requiring adding O_CLOEXEC to almost ALL instances of open() in all code
that has been written ever. The justification for this is that the
historic default was wrong, and you can't change it (e.g. this won't
work: changing the behavior of exec() and not inherit the FD to the
child process, unless a hypothetical O_KEEP_EXEC flag is set).
But on the other hand, surely you could have introduced an exec()
variant which does close all FDs, except a whitelist of FDs passed to
it. Let's call it execve2(). In fact, I'm going to argue that exec()
call sites are the most aware of whether (and which) FDs to inherit.
Some programs even tried to explicitly iterate over all opened FDs and
explicitly close "unwanted" FDs (which of course was problematic for
other reasons), and such an execve2() call would have been the ideal
solution.
Maybe this proposed solution would have had problems too. But surely
revisiting and reviewing every exec*() call would have been simpler than
reviewing every open() call. And more importantly, having to extend
every damn library function that either calls open() or creates FDs in
some other way, like mkstemp().
What argument are there going to be against this? That there will be
library code that can't keep working correctly with processes that use
the "old" exec? Well, what about all my legacy library code that uses
open() incorrectly, and that will break no matter what?
Well, I'm not going to claim that I can come up with better solutions
than POSIX (generally or in this case), but this situation is ABSOLUTELY
ATROCIOUS. It makes win32 programming look attractive compared to POSIX,
that standard pandering to dead people from the past. (Note: not trying
to insult dead people.)
I'm not sure what POSIX is even doing. Anything useful? Doesn't look
like it to me. Are they paid? Why? They didn't even fix the locale mess,
nor do they intend to. I bet they're proud of discussing compatibility
to 70ies code day in and day out iwtohut ever producing anything useful.
What a load of crap. They seriously got to do better than this.
Oh, and my wrapper is probably buggy. Fortunately that doesn't matter.
Also I'm dumping this into io.h. Originally, io.h was just supposed to
replace broken implementation of standard functions by MinGW (and then
by Android), but whatever, just give a dumping ground for shit code.
Backwards demuxing usually seeks back back by a "random" amount (set by
a user option) when it needs new preceding packets. It turns out a past
change made these backwards seek amounts add up when it didn't need to
(i.e. subtracting the amount from the seek pos without properly
resetting it), which could possibly slow down playback as it went on.
The reason for this was that back_seek_pos was set for every stream on
every seek. This made the reset not affect other streams (in particular
streams which weren't used and never were reset, or which didn't reset
that often). But as the commit adding it showed, this is needed only to
set the initial position. So do that.
Fixes: "demux: fix initial backward demuxing state in some cases"
When packet appending sets the start of the range, it adjusts the range
by seek_preroll. Do this when packets are pruned from the start of the
range too.
(Yeah, seek_preroll handling is probably broken in some other cases. It
was halfhearted to begin with.)
The main thing this commit does is removing demux_packet.kf_seek_pts. It
gets rid of 8 bytes per packet. Which doesn't matter, but whatever.
This field was involved with much of seek range updating and pruning,
because it tracked the canonical seek PTS (i.e. start PTS) of a packet
range. We have to deal with timestamp reordering, and assume the start
PTS is the lowest PTS across all packets (not necessarily just the first
packet). So knowing this PTS requires looping over all packets of a
range (no, the demuxer isn't going to tell us, that would be too sane).
Having this as packet field was perfectly fine. I'm just removing it
because I started hating extra packet fields recently.
Before this commit, this value was cached in the kf_seek_pts field (and
computed "incrementally" when adding packets). This commit computes the
value on demand (compute_keyframe_times()) by iterating over the placket
list. There is some similarity with the state before 10d0963d85,
where I introduced the kf_seek_pts field - maybe I'm just moving in
circles. The commit message claims something about quadratic complexity,
but if the code before that had this problem, this new commit doesn't
reintroduce it, at least. (See below.)
The pruning logic is simplified (I think?) - there is no "incremental"
cached pruning decision anymore (next_prune_target is removed), and
instead it simply prunes until the next keyframe like it's supposed to.
I think this incremental stuff was only there because of very old code
that got refactored away before. I don't even know what I was thinking
there, it just seems complex. Now the seek range is updated when a
keyframe packet is removed.
Instead of using the kf_seek_pts field, queue->seek_start is used to
determine the stream with the lowest timestamp, which should be pruned
first. This is different, but should work well. Doing the same as the
previous code would require compute_keyframe_times(), which would
introduce quadratic complexity.
On the other hand, it's fine to call compute_keyframe_times() when the
seek range is recomputed on pruning, because this is called only once
per removed keyframe packet. Effectively, this will iterate over the
packet list twice instead of once, and with some locality. The same
happens when packets are appended - it loops over the recently added
packets once again. (And not more often, which would go above linear
complexity.)
This introduces some "cleverness" with avoiding calling
update_seek_ranges() even when keyframe packets added/removed, which is
not really tightly coupled to the new code, and could have been in a
separate commit.
Removing next_prune_target achieves the same as commit b275232141,
which is hereby reverted (stale is_bof flags prevent seeking before the
current range, even if the beginning of the file was pruned). The seek
range is now strictly computed after at least one packet was removed,
and stale state should not be possible anymore.
Range joining may over-allocate the index a little. It tried hard to
avoid this before by explicitly freeing the old index before creating a
new one. Now it iterates over the old index while adding the entries to
the new one, which is simpler, but may allocate twice the memory in the
worst case. It's not going to matter for anything, though.
Seeking will be slightly slower. It needs to compute the seek PTS values
across all packets in the vicinity of the seek target. The previous code
also iterated over these packets, but now it iterates one packet range
more.
Another minor detail is that the special seeking code for SEEK_FORWARD
goes away. The seeking code will now iterate over the very last packet
range too, even if it's incomplete (i.e. packets are still being
appended to it). It's fine that it touches the incomplete range, because
the seek_end fields prevent that anything particularly incorrect can
happen. On the other hand, SEEK_FORWARD can now consider this as seek
target, which the deleted code had to do explicitly, as kf_seek_pts was
unset for incomplete packet ranges.
Obviously doesn't sense with this order. The git history shows that this
comment was touched multiple times, without ever fixing it. It was
originally added in 2016, where the "for" was missing. Later, the "for"
was added, but to the wrong position.
What the fuck?
The old implementation didn't work for the OGG case. Discard the old
shit code (instead of fixing it), and write new shit code. The old code
was already over a year old, so it's about time to rewrite it for no
reason anyway.
While it's true that the old code appears to be broken, the main reason
to rewrite this is to make it simpler. While the amount of code seems to
be about the same, both the concept and the actual tag handling are
simpler. The result is probably a bit more correct.
The packet struct shrinks by 8 byte. That fact that it wasted 8 bytes
per packet for a rather obscure use case was the reason I started this
at all (and when I found that OGG updates didn't work). While these 8
bytes aren't going to hurt, the packet struct was getting too bloated.
If you buffer a lot of data, these extra fields will add up. Still quite
some effort for 8 bytes. Fortunately, it's not like there are any
managers that need to be convinced whether it's worth doing. The freedom
to waste time on dumb shit.
The old implementation attached the current metadata to each packet.
When the decoder read the packet, the packet's metadata was made
current. The new implementation stores metadata as separate list, and
requires that the player frontend tells it the current playback time,
which will be used to find the currently valid metadata. In both cases,
the objective was to correctly update metadata even if a lot of data is
buffered ahead (and to update them correctly when seeking within the
demuxer cache).
The new implementation is actually slightly more correct, because it
uses the playback time for the metadata lookup. Consider if you have an
audio filter which buffers 15 seconds (unfortunately such a filter
exists), then the old code would update the current title 15 seconds too
early, while the new one does it correctly.
The new code also simplifies mixing the 3 metadata sources (global, per
stream, ICY). We assume these aren't mixed in a meaningful way. The old
code tried to be a bit more "exact". I didn't bother to look how the old
code did this, but the new code simply always "merges" with the previous
metadata, so if a newer tag removes a field, it's going to stick around
anyway.
I tried to keep it simple. Other approaches include making metadata a
special sh_stream with metadata packets. This would have been
conceptually clean, but the implementation would probably have been
unnatural (and doesn't match well with libavformat's API anyway). It
would have been nice to make the metadata updates chapter points (makes
a lot of sense for the intended use case, web radio current song
information), but I don't think it would have been a good idea to make
chapters suddenly so dynamic. (Still an idea to keep in mind; the new
code actually makes it easier to work towards this.)
You could mention how subtitles are timed metadata, and actually are
implemented as sparse packet streams in some formats. mp4 implements
chapters as special subtitle stream, AFAIK. (Ironically, this is very
not-ideal for files. It would be useful for streaming like web radio,
but mp4 is extremely bad for streaming by design for other reasons.)
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
Some OGG web radio streams use timestamp resets when a new song starts
(you can find those Xiph's directory - other streams there don't show
this behavior). Basically, the OGG stream behaves like concatenated OGG
files, and "of course" the timestamps will start at 0 again when the
song changes. This is very inconvenient, and breaks the seekable demuxer
cache. In fact, any kind of seeking will break
This is more time wasted in Xiph's bullshit. No, having timestamp resets
by design is not reasonable, and fuck you. I much prefer the awful
ICY/mp3 streaming mess, even if that's lower quality and awful. Maybe it
wouldn't be so bad if libavformat could tell us WHERE THE FUCK THE RESET
HAPPENS. But it doesn't, and the randomly changing timestamps is the
only thing we get from its API.
At this point, demux_lavf.c is like 90% hacks. But well, if libavformat
applies this strange mixture of being clever for us vs. giving us
unfiltered garbage (while pretending it abstracts everything, and hiding
_useful_ implementation/low level details), not much we can do.
This timestamp linearizing would, in general, probably be better done
after the decoder, because then we wouldn't need to deal with timestamp
resets. But the main purpose of this change is to fix seeking within the
demuxer cache, so we have to do it on the lowest level.
This can probably be applied to other containers and video streams too.
But that is untested. Some further caveats are explained in the manpage.
Probably doesn't change anything, other than looking slightly better. In
theory, the common function has some stuff that makes it more likely
that timestamps round-trip through conversions properly, but I didn't
confirm that.
This may call memmove() with size==0 and a NULL data pointer. In
addition to this being UB with memmove(), I think it's UB to do
arithmetic on a NULL pointer too. Of course, this doesn't matter in
practice at all, and is just stupidity to torture programmers.
Remove the duplicated creation of the first range. Explicitly destroy
ranges, including the last one on final deinit.
It looks like this also fixes a leak of removed range structs, which was
never noticed because they're so small, and were freed on final deinit
due to having the demuxer as talloc parent.
This improves upon the previous commit too (that change should have
been part of it I guess). Sub-demuxers (demux_timeline only) now
automatically don't use the cache (like it was intended by the previous
commit). The cache is "initialized" (or disabled) last in the recursive
call chain, which is messy, but this sub demuxer stuff FUCKING SUCKS, as
mentioned in the previous commit message. This would be no problem if
the caching layer and actual demuxer implementations were separate.
Most of this change has no purpose. Might make (de-)initialization of
further cache exerpiments simpler.
It seems the so called demuxer cache wasn't really disabled for
sub-demuxers (timeline stuff). This was relatively harmless, since the
actual packet data was shared anyway via refcounting. But with the
addition of a mmap cache backend, this may change a lot.
So strictly disable any caching for sub-demuxers. This assumes that
users of sub-demuxers (only demux_timeline.c by now?) strictly use
demux_read_any_packet(), since demux_read_packet_async() will require
some minor read-ahead if a low level packet read returned a packet for a
different stream.
This requires some awkward messing with this fucking heap of trash. The
thing that is really wrong here is that the demuxer API mixes different
concepts, and sub-demuxers get the same API as decoders, and use the
cache code.
The demuxer cache tries to track the number of bytes allocated for the
cache. In addition to the packet queue, the seek index is another data
structure that roughly depends on the amount of packets cached. So the
index size should somehow be part of the total number of bytes tracking.
Until now, this was handled with KF_SEEK_ENTRY_WORST_CASE, basically a
shitty heuristic. It was a guess (and probably rather an upper bound
than a lower bound). The implementation details made it annoying, and it
was conceptually inaccurate too.
Change this, and instead simply add the index size to the total cache
size. This essentially makes it part of the backbuffer. It's nice that
this cleanly decouples it from the packet size tracking itself.
Since it's part of the backbuffer number of bytes now, packet pruning
can't necessarily free enough space in the backbuffer anymore. Before
this commit, the backbuffer consisted of packets only, so it was
possible to reduce its size to 0 by pruning all packets until the
decoder reader position, at which point a packet was accounted as
forward buffered. Now the index is added to this, and it can't be
pruned. Replace the assert() because of this changed invariant.
The conversion to string as the pretty printer returns it is
sometimes used on OSD. I think it's pretty odd that quantities below 1
KB are shown as number without suffix. So use "B" for them.
For orthogonality, allow the same for parsing. (Although strictly
speaking, this is not a requirement of the option API. Option parsers
don't need to accept pretty-printed strings.)
Someone who rams a knife into his own hand just to see what happens is
normally put in a psychiatric ward. But in software, this is acceptable
behavior. Programs are not supposed to crash just because a user did
something unreasonably dumb.
Switching tracks during backward playback is such a thing. It triggered
an assertion because the newly enabled stream was not properly
initialized for backward playback. Fix this, and make it actually work
(mostly; it still takes a "while" until playback recovers fully).
This actually makes some aspects of initialization slightly cleaner.
Track switching doesn't run reset_playback_state(), so a track enabled
at runtime during backward playback would lead to a messed up state.
This commit just does a bad code monkey fix to this. It feels like there
needs to be a much better way to propagate this state.
Not sure if this is bug-free. You _always_ make bugs when writing a
binary search from scratch (and such is the curse of C, though if I did
this in C++ it would probably end in blood). It seems to work though,
checking against the normal linear search.
It's slightly faster. Not much.
I wonder if the termination condition can be written in a nicer/elegant
way. I guess the fact that it's not a == predicate makes this slightly
messier?
The purpose of the seek index is to avoid having to iterate over the
full linked list of cached packets, which should speed up seeking. Until
now, there was an excuse of a seek index, that didn't really work.
The idea of the old index was nice: have a fixed number of entries (no
need to worry about exceeding memory requirements), which are
"stretched" out as the cache gets bigger. The size of it was 16 entries,
which in theory should speed up seeking by the factor 16, given evenly
spaced out entries. To achieve this even spacing, it attempted to "thin
out" the index by half once the index was full (see e.g. index_distance
field). In my observations this didn't really work, and the distribution
of the index entries was very uneven. Effectively, this did nothing. It
probably worked once and I can't be assed to debug my own shit code.
Writing new shit code is more fun.
Write new shit code for fun. This time it's a complete index. It's kept
in a ringbuffer (for easier LIFO style appending/removing), which is
resized with realloc if it becomes too small.
Actually, the index is not completely completely; it's still "thinned
out" by a hardcoded time value (INDEX_STEP_SIZE). This should help with
things like audio or crazy subtitle tracks (do they still create
those?), where we can just iterate over a small part of the linked
packet list to get the exact seek position. For example, for AAC audio
tracks with typical samplerates/framesizes we'd iterate about 50 packets
in the linked list.
The results are good. Seeking in large caches is much faster now,
apparently at least 1 or 2 orders of magnitude. Part of this is because
we don't need to touch every damn packet in a huge linked list (bad
cache behavior - the index is a linear memory region instead), but
"thinning" out the search space also helps. Both aspects can be easily
tested (setting INDEX_STEP_SIZE to 0, and replacing e->pts with
e->pkt->kf_seek_pts in find_seek_target()).
This does use more memory of course. In theory, we could tolerate memory
allocation failures (the index is optional and only for performance),
but I didn't bother and inserted an apologetic comment instead, have fun
with the shit code). the memory usage doesn't seem to be that bad,
though. Due to INDEX_STEP_SIZE it's bounded by the file duration too.
Try to account for the additional memory usage with an approximation
(see KF_SEEK_ENTRY_WORST_CASE). It's still a bit different, because the
index needs a single, potentially large allocation.
The search was slightly more complicated and slow than it had to be. It
didn't assume that the packet list was sorted, which is responsible for
much of this. (I think the search code was borrowed from demux_mkv.c,
which does not sort index entries.)
There was a half-hearted attempt to make it exit early, but it was
mostly ineffective.
Simplify the code based on the assumption that the list is sorted. This
will exit the search loop once the worst case candidate entry was
checked.
Mostly about the packet queue and the subtitle handling of it.
(This mess sure sounds like a good argument to give up the separate
stream queues, and using a single packet queue per cached range.)
Until now, this usually passed a single audio frame to the decoder, and
then did a backstep operation (cache seek + frame search) again. This is
probably not very efficient, especially considering it has to search the
packet queue from the "start" every time again.
Also, with most audio codecs, an additional "preroll" frame was passed
first. In these cases, the preroll frame would make up 50% of audio
decoding time. Also not very efficient.
Attempt to fix this by returning multiple frames at once. This reduces
the number of backstep operations and the ratio the preoll frames. In
theory, this should help efficiency. I didn't test it though, why would
I do this? It's just a pain. Set it to unscientific 10 frames.
(Actually, these are 10 keyframes, so it's much more for codecs like
TrueHD. But I don't care about TrueHD.)
This commit changes some other implementation details. Since we can
return more than 1 non-preroll keyframe to the decoder, some new state
is needed to remember how much. The resume packet search is adjusted to
find N ("total") keyframe packets in general, not just preroll frames.
I'm removing the special case for 1 preroll packet; audio used this, but
doesn't anymore, and it's premature optimization anyway.
Expose the new mechanism with 2 new options. They're almost completely
pointless, since nobody will try them, and if they do, they won't
understand what these options truly do. And if they actually do, they
most likely would be capable of editing the source code, and we could
just hardcode the parameters. Just so you know that I know that the
added options are pointless.
The following two things are truly unrelated to this commit, and more
like general refactoring, but fortunately nobody can stop me.
Don't set back_seek_pos in dequeue_packet() anymore. This was sort of
pointless, since it was set in find_backward_restart_pos() anyway (using
some of the same packets). The latter function tries to restrict this to
the first keyframe range though, which is an optimization that in theory
might break with broken files (duh), but in these cases a lot of other
things would be broken anyway.
Don't set back_restart_* in dequeue_packet(). I think this is an
artifact of the old restart code (cf. ad9e473c555). It can be done
directly in find_backward_restart_pos() now. Although this adds another
shitty packet search loop, I prefer this, because clearer what's
actually happening.
The size of all forward buffered packets is used to control maximum
buffering.
Until now, this size was incrementally adjusted, but had to be
recomputed on seeks within the cache. Doing this was actually pretty
expensive. It iterates over a linked list of separate memory allocations
(which are probably spread all over the heap due to the allocation
behavior), and the demux_packet_estimate_total_size() call touches a lot
of further memory locations. I guess this affects the cache rather
negatively. In an unscientific test, the recompute_buffers() function
(which contained this loop) was responsible for roughly half of the time
seeking took.
Replace this with a way that computes the buffered size between 2
packets in constant times. The demux_packet.cum_pos field contains the
summed sizes of all previous packets, so subtracting cum_pos between two
packets yields the size of all packets in between. We can do this
because we never remove packets from the middle of the queue. We only
add packets to the end, or remove packets at the beginning.
The tail_cum_pos field is needed because we don't store the end position
of a packet, so the last packet's position would be unknown. We could
recompute the "estimated" packet size, or store the estimated size in
the packet struct, but I just didn't like this.
This also removes the cached fw_bytes fields. It's slightly nicer to
just recompute them when needed. Maintaining them incrementally was
annoying. total_size stays though, since recomputing it isn't that cheap
(would need to loop over all ranges every time).
I'm always using uint64_t for sizes. This is certainly needed (a stream
could easily burn through more than 4GB of data, even if much less of
that is cached). The actual cached amount should always fit into size_t,
so it's casted to size_t for printfs (yes, I hate the way you specify
stdint.h types in printfs, the less I have to use that crap, the
better).
In ancient times, the number of packets was used to limit excessive
read-ahead. This was completely replaced by tracking the size in bytes.
The number of packets was used in debugging output only.
In one case (packet got demuxed and is added to a queue), only log
whether there were packets on this stream before. (Unknown whether it's
useful.)
In another case (queue overflow), actually count the number of packets.
It's vaguely useful, and the message with the number of packets is shown
only once after a seek reset, so it doesn't matter whether it's slow.
Some state wasn't reset when decoding was started without a seek reset
before it. The code used to rely on reset_decoder() resetting this
state, but since the commit referenced below, reset_decoder() does less
than reset().
Fix this by explicitly calling reset() on initialization.
Fixes: "f_decoder_wrapper: avoid full reset on timeline switch etc."
Some files don't start with keyframe packets. Normally, this is not
sane, but the sample file which triggered this was a cut TV capture
transport stream. And this shouldn't happen anyway.
Introduce a further heuristic: if the last seek target was before the
start of the cached data, and the start of the cache is marked as BOF
(beginning of file), then we won't find anything better. This is
possibly a bit shaky, because both seek_start and back_seek_pos weren't
made for this purpose. But I can't come up with situations where this
would actually break. (Leave this to shitty broken files I hit later.)
I also considered finding the first packet in the cache that is marked
as keyframe, i.e. the first actual seek target, and comparing it to
"first", but I didn't like it much. Well whatever.
It's a bit silly that this caused a hard freeze (and similar issues
still will). The problem is that the demuxer holds the lock and has no
reason to release it. And in general, there's a single lock for the
entire demuxer cache. Finer grained locking would probably not make much
sense. In theory status of available data and maybe certain commands to
the demuxer could be moved to separate locks, but it would raise
complexity, and you'd probably still need to get the central lock in
some cases, which would deadlock you anyway.
It would still be nice if some minor corner case in the wonderfully
terrible and complex backward demuxer state machine couldn't lock up the
player. As a hack, unlock and then immediately lock again. Depending on
the OS mutex implementation, this may give other waiters a chance to
grab the lock. This is not a guarantee (some OSes may for example not
wake up other waiters until the next time slice or something), but works
well on Linux.
The step_backwards function set reader_head to the start of the current
cache range. This was completely unnecessary and made it _much_ slower.
Remove the code that adjusts reader_head. Merge the rest of the code
into the only caller and remove the function.
The comment on the removed code was quite right. It was "inefficient".
Removing it delegates going to an early position to the normal seek
code, triggered by find_backward_restart_pos() incremental back seek
logic. I suppose especially audio benefits from this, because this
happens for every single audio packet (except maybe freaky bullshit like
TrueHD, which has "keyframes").
The blabla about performance in the removed comments is still true, but
now applies to the seek code itself only.
Fixes "mpv file.mkv --cache --demuxer-cache-wait --play-dir=backward",
and other situations where the demuxer cache contains the entire file,
and playback is to start from the end. It also can be triggered when
starting playback normally with --cache, and once everything is in the
cache, enabling backward playback and seeking past EOF.
In all cases, the cache seek will set reader_head=NULL (because you
seeked to/past EOF). Then the code (the one modified by this commit)
sees that ds->queue->is_bof==true, and thinks we've reached BOF
(beginning of file) while searching for a useful packet, i.e. we found
nothing and playback really can only end.
Obviously this is nonsense, we've found only nothing if we actually
searched from the beginning, not some "random" reader_head (== first)
value that does not include the entire cache. That means the condition
should trigger only if the start of the search (first variable) points
to the beginning of the cache (ds->queue->head).
Not taking this if means we'll seek to an earlier position and retry.
Also, a seek before the beginning of the cache will always end up with
reader_head==ds->queue->head, i.e. we'll terminate properly.
That comment was quite right.
Before this commit, there was a single process_decoded_frame() function.
It handled various aspects of dealing with a newly decoded frame. Move
some of these to a separate process_output_frame() function.
This new function is called in the order the frames are returned to the
playback core. Some correct_audio_pts() (was process_audio_frame())
becomes slightly less awkward due to this, and the timestamp smoothing
can actually work in backward playback mode now (thus moving p->pts out
of reset_decoder()).
Behavior for normal playback also changes subtly. This shouldn't matter
in sane cases, but if you mix broken files, --no-correct-pts, and
timeline stuff, differences in behavior might be visible.
Timeline clipping (EDL/ordered chapters) works now, because it's done
before "transforming" the timestamps. Audio timestamp smoothing happens
after it, which is a behavior change, but should be more correct. This
still runs crazy_video_pts_stuff() before everything else. On the pther
hand, --no-correct-pts or missing timestamp processing is done last. But
these things didn't really work with timeline before.
Slightly cleaner. We don't need to awkwardly backup "some" state on
backwards playback. Due to not resetting last_format, normal timeline
switches don't unconditionally trigger recomputing of certain image
parameters. Also probably doesn't reset framedrop parameters, although I
don't care about that part.
This could lead to nonsense when backward playback is involved. Better
reduce the possible interactions. Besides, it's better to fully reset
things on seeks in general.
The only exception is has_broken_packet_pts, which enables hr-seek if
everything looks good. It's intended to trigger at the second hr-seek or
so if the file is normal, and to disable it if the file is broken. It
tries to avoid enabling the hr-seek logic before it can know about
whether things are "good", so resetting it on seeks would obviously
never enable it. Document it as explicit exception.
And add simpler aliases for the modes.
I'm not sure how to name things, and the option list is in general full
of different conventions. Some names are shortened, some are explicit
and long.
I guess options that have a chance to be used normally (i.e. not obscure
tuning or debugging) should have a short and convenient names.
In this specific case, play-direction is like a mixture of both. It
should be either playback-direction or play-dir, not shorten one word
but not the other.
The convenience aliases are because I got sick of typing out "backward".
I guess "back" would also do it, but there's no proper antonym (and
maybe it's "wrong" in the strict sense of the word).
Together with the previous commit, this seems to make backward playback
work in files with vorbis and mp3 audio codecs.
For Vorbis (with libavcodec's decoder, didn't test libvorbis), the first
packet was just always completely discarded. This happened even though
we tell libavcodec that we do discarding of padding manually. It simply
happened inside the codec, not libavcodec's general initial padding
handling. In addition, the first output decoded frame seems to contain
partial data. (Unlike the opus decoder, it doesn't report any padding at
all.)
The Opus decoder (again libavcodec only tested) reports an initial
padding, but it appears to be too small, and it sounds right only with 2
packets discarded. So its status doesn't change.
I'm not sure why I need 2 frames for mp3, but with that value I had
success on the samples I tested.
Some audio codecs will discard or cut the first frames when starting
decoding. While some of that works through well-defined mechanisms (like
initial padding), it's in general very codec/decoder specific, and not
really predictable. In addition, libavcodec isn't very good with
reporting "dropped" frames (and our internal interface reflects this).
It seems our only chance to handle this is through timestamps.
In theory, it would be best to discard frames that have timestamps
before the "resume" position. But since video has reordered timestamps,
we'd need to put some effort into finding this position. The first video
packet doesn't necessarily contain this timestamp. (In theory, we could
just do this in the demuxer with some trivial additional work, and set
it on the packet's kf_seek_pts field. Although this field is supposed to
contain just this value, the field is considered demuxer-internal, and I
didn't want to make matters worse by reusing it for the interface to the
decoder. With some more effort and buffering, we could calculate this
value within the decoder, but fuck that.)
The approach chosen in this commit is setting the timestamp to NOPTS.
This will break in some obscure situations, but backward playback is a
pretty obscure feature to begin with, so I considered this a reasonable
implementation choice.
Before passing a preroll packet to the decoder, its timestamps are set
to NOPTS. Frames that are returned from the decoder and have the NOPTS
timestamp are considered preroll and are discarded. This happens only
during "preroll" mode (preroll_discard==true), so it doesn't affect
normal forward playback. It's disabled on the first packet with a
timestamp, so it can tolerate some crap even in backward playback mode.
We don't check the dts fields out of laziness (decoded audio frames
don't even have this field).
I considered using an approach using the EDL clipping infrastructure (as
mentioned in the last two paragraphs in the commit message of commit
" demux_lavf: implement bad hack for backward playback of wav"). This
didn't work, and I blamed timestamp rounding within mpv for it. But the
problem was actually due to Matroska-rounded timestamps. Since the audio
frame size isn't exactly aligned to 1ms, there will be an overlap (or
gap) in the timestamps. This overlap is much smaller than 1ms, since
it's just the sub-millisecond remainder part of the audio frame size.
This makes the timestamps discontinuous and unreliable for the purpose
we wanted to use it. We can't just smooth the timestamps in the demuxer
either.
Matroska has this weird concept of "lacing", which are really sub-blocks
packed into a larger actual block. They are demuxed as individual
packets, because that's what the decoder needs. Basically they're a
Matroska hack to reduce per-packet muxing overhead.
One problem is that these sub-blocks don't have timestamps. The
timestamps need to be created from the "default duration". If this
default duration isn't in the file header (or if we drop it when it has
a known broken value), the resulting packets won't have a timestamp.
This is an usual and weird situation, that may confuse the demuxer layer
(and backward playback in particular). Fix this by not setting the
keyframe flag for these.
This affects only audio packets. Subtitle lacing is explicitly not
supported (in theory would at least lack a way to specify durations),
and video won't usually use it for real video codecs (not every frame is
a "keyframe", timestamp reordering).
Clarify existing semantics for the --start/--end/--length options.
De-emphasize the difference between absolute and relative timestamps,
since they've not been different by default since mpv 0.14.
Document a bug, that also happens to work as a feature: if the option
value begins with spaces, the code for checking for relative timestamps
is inactive, and they're always considered absolute. The check is done
on the first character of the string - so even a negative timestamp will
be treated as absolute.)
Yes, this is useful in extremely rare situations, such as when you
really want send a specific timestamp (even a negative one) to the
demuxer.
Another shitty obscure feature that usually nobody notices.
Unsurprisingly, it doesn't go well with backward playback mode.
If you use --keep-open in forward playback mode, and seek past the end
of the file, it tries to seek to the very last frame. The demuxer will
seek to the last "keyframe" before the end (i.e. some frames to go in
most cases), and trying to hr-seek to the file duration often won't cut
it, so this requires some special code. The function at hand seeks
"close" to the end, and then stops hr-seek when the last frame us
encountered (simple enough and very effective).
In backward playback mode, start and end are reversed, and we need to
seek "close" to the start of the file instead. Simple enough to do, and
it works.
One problem is that command.c has some weird logic to make going beyond
the last chapter either end playback (--keep-open=no), or jump to the
last frame. Now this will jump to the first frame, which is weird, but
let's ignore this.
Another problem is that seeking before playback start position hits EOF
in backward playback mode, which is a demuxer bug, and has nothing to do
with this code. But it triggers this code, so seeking before the start
will show the "last" frame. (My description is a mess with directions.
Figure it out yourself.)
Obviously should seek back to the end of the file when it loops.
Also remove some minor code duplication around start times. This isn't
the correct solution by the way. Rather than hoping we know a reasonable
start/end time, this stuff should instruct the demuxer to seek to the
exact location. It'll work with 99% of all normal files, but add an
appropriate comment (that basically says the function is bullshit) to
get_start_time() anyway.
This changes the behavior of the --ab-loop-a/b options. In addition, it
makes it work with backward playback mode.
The most obvious change is that the both the A and B point need to be
set now before any looping happens. Unlike before, unset points don't
implicitly use the start or end of the file. I think the old behavior
was a feature that was explicitly added/wanted. Well, it's gone now.
This is because of 2 reasons:
1. I never liked this feature, and it always got in my way (as user).
2. It's inherently annoying with backward playback mode.
In backward playback mode, the user wants to set A/B in the wrong order.
The ab-loop command will first set A, then B, so if you use this command
during backward playback, A will be set to a higher timestamps than B.
If you switch back to forward playback mode, the loop would stop
working. I want the loop to just continue to work, and the chosen
solution conflicts with the removed feature.
The order issue above _could_ be fixed by also switching the AB-loop
user option values around on direction switch. But there are no other
instances of option changes magically affecting other options, and doing
this would probably lead to unexpected misery (dying from corner cases
and such).
Another solution is sorting the A/B points by timestamps after copying
them from the user options. Then A/B options set in backward mode will
work in forward mode. This is the chosen solution. If you sort the
points, you don't know anymore whether the unset point is supposed to
signify the end or the start of the file.
The AB-loop code is slightly better abstracted now, so it should be easy
to restore the removed feature. It would still require coming up with a
solution for backwards playback, though.
A minor change is that if one point is set and the other is unset, I'm
rendering both the chapter markers and the marker for the set point.
Why? I don't know. My test file had chapters, and I guess I decided this
looked better.
This commit also fixes some subtle and obvious issues that I already
forgot about when I wrote this commit message. It cleans up some minor
code duplication and nonsense too.
Regarding backward playback, the code uses an unsanitary mix of internal
("transformed") and user timestamps. So the play_dir variable appears
more than usual.
To mention one unfixed issue: if you set an AB-loop that is completely
past the end of the file, it will get stuck in an infinite seeking loop
once playback reaches the end of the file. Fixing this reliably seemed
annoying, so the fix is "just don't do this". It's not a hard freeze
anyway.
This code attempts to seek to the last frame by seeking close to the
end, and then decoding until the last frame has been reached. To do so
it sets hrseek_lastframe, which for video enables some logic to "catch"
this last frame, and completely ignores hrseek_pts. But audio still may
use hrseek_pts
I don't know if the original author (me) was thinking, if anything, when
setting this variable to 1e99, essentially a random, number. It's very
large, and a timestamp like this will never happen, so it does its job.
But it's random.
Use INFINITY instead. It will skip all audio samples in the audio code
correctly. This change doesn't fix anything, but it does get rid of the
random looking number.
The get_play_start_pts() function was supposed to return "rebased"
(relative to 0) timestamps. This was roundabout, because one of 2
callers just added the offset back, and the other caller actually
expected an absolute timestamp.
Change rel_time_to_abs() (whose return value get_play_start_pts()
returns without further changes) to return absolute times.
This should fix that absolute and relative times passed to --start and
--end were treated the same, which can't be right. It probably also
fixes --end if --rebase-start-time=no is used (which can't have been
correct either).
All in all I'm not sure why --rebase-start-time=no or absolute vs.
relative times in --start/--end even exist, when they were incorrectly
implemented for years.
Untested, because no sample file and I don't care. However, if anyone
cares, and I got it wrong, I hope it's simple to fix.
Has been deprecated for almost 3 years. Manpage didn't mention the
deprecation, but CLI and release notes did. It wouldn't be much effort
to keep this option working, but I just don't see the damn point.
--start/--end can specify chapters using special syntax, which is
equivalent.
We need to transform the timestamp returned by get_play_end_pts().
I considered making it return the transformed timestamp directly. There
are 4 callers; 2 need a transformed timestamps, 2 don't. So I guess it
doesn't matter.
This adds the stops using the same logic get_play_end_pts() and
handle_loop_file(). It did that before, it just looks slightly different
now. It also won't try to add MP_NOPTS_VALUE as stop value.
Shitty ancient hack that wastes my time all the time.
demux.c: always return the coverart packet as soon as possible, and
don't let the backward demux state machine possibly stop it.
f_decoder_wrapper.c: mess with some shit until it somehow starts to
work. I think the old code tried to let it cleverly fall through so the
packet was processed "normally"; just make it run the "usual" code
instead.
This commit generally fixes backward playing in wav, at least in most
PCM cases.
libavformat's wav demuxer (and actually all other raw PCM based
demuxers) have a specific behavior that breaks backward demuxing. The
same thing also breaks persistent seek ranges in the demuxer cache,
although that's less critical (it just means some cached data gets
discarded). The backward demuxing issue is fatal, will log the message
"Demuxer not cooperating.", and then typically stop doing anything.
Unlike modern media formats, these formats don't organize media data in
packets, but just wrap a monolithic byte stream that is described by a
header. This is good enough for PCM, which uses fixed frames (a single
sample for all audio channels), and for which it would be too expensive
to have per frame headers.
libavformat (and mpv) is heavily packet based, and using a single packet
for each PCM frame causes too much overhead. So they typically "bundle"
multiple frames into a single packet. This packet size is obviously
arbitrary, and in libavformat's case hardcoded in its source code.
The problem is that seeking doesn't respect this arbitrary packet
boundary. Seeking is sample accurate. You can essentially seek inside a
packet. The resulting packets will not be aligned with previously
demuxed packets. This is normally OK.
Backward seeking (and some other demuxer layer features) expect that
demuxing an earlier demuxed file position eventually results in the same
packets, regardless of the seeks that were done to get there. I like to
call this "deterministic" demuxing. Backward demuxing in particular
requires this to avoid overlaps, which would make it rather hard to get
continuous output.
Fix this issue by detecting wav and hopefully other raw audio formats
with a heuristic (even PCM needs to be detected as heuristic). Then, if
a seek is requested, align the seek timestamps on the guessed number of
samples in the audio packets returned by the demuxer.
The heuristic excludes files with multiple streams. (Except "attachment"
video streams, which could be an ID3 tag. Yes, FFmpeg allows ID3 tags on
WAV files.) Such files will inherently use the packet concept in some
way.
We don't know how the demuxer chooses the internal packet size, but we
assume that it's fixed and aligned to PCM frame sizes. The frame size is
most likely given by block_align (the native wav frame size, according
to Microsoft). We possibly need to explicitly read and discard a packet
if the seek is done without reading anything before that. We ignore any
subsequent packet sizes; we need to avoid the very last packet, which
likely has a different size.
This hack should be rather benign. In the worst case, it will "round"
the seek target a little, but the maximum rounding amount is bounded.
Maybe we _could_ round up if SEEK_FORWARD is specified, but I didn't
bother.
An earlier commit fixed the same issue for mpv's demux_raw.
An alternative, and probably much better solution would be clipping
decoded data by timestamp. demux.c could allow the type of overlap the
wav demuxer introduces, and instruct the decoder to clip the output
against the last decoded timestamp. There's already an infrastructure
for this (demux_packet.end field) used by EDL/ordered chapters.
Although this sounds like a good solution, mpv unfortunately uses floats
for timestamps. The rounding errors break sample accuracy. Even if you
used integers, you'd need a timebase that is sample accurate (not always
easy, since EDL can merge tracks with different sample rates).
Yay, more subtle state on top of this nightmarish, fragile state
machine. But this is what happens when you subvert the laws of nature.
This simple checks where playback should "resume" from when no packets
were returned to the decoder yet after the seek that initiated backward
playback. The main purpose is to process the first returned keyframe
range in the same way like all other ranges. This ensures that things
like preroll are included properly.
Before this commit, it could for example have happened that the start of
the first audio frame was slightly broken, because no preroll was
included. Since the audio frame is reversed before sending it to the
audio output, it would have added an audible discontinuity before the
second frame was played; all subsequent frames would have been fine.
(Although I didn't test and confirm this particular issue.)
In future, this could be useful for certain other things.
At least the condition for delaying the backstep seek becomes simpler
and more explicit.
Move the code that attempts to start demuxing up in dequeue_packet.
Before, it was not called when the stream was in back_restarting state.
This commit makes streams be in back_restarting state at initialization,
so the demuxer would never have started reading.
Likewise, we need to call back_demux_see_packets() right after seek in
case the seek was within the cache. (We don't bother with checking
whether it was a cached seek; nothing happens if it was a normal one.)
There is nothing else that would process these cached packets
explicitly, although coincidences could sporadically trigger it.
The check for back_restart_next in find_backward_restart_pos() now
decides whether to use this EOF special code. Since the backward
playback start state also sets this variable, we don't need some of
the complex checks in dequeue_packet() anymore either.
As well as other filtering. I was writing this with the assumption that
timestamps go backwards (which I first planned to do). But in fact,
timestamps go forward, frame durations are positive, and adding a frame
duration to a timestamp yields the correct result. The only strange
thing is that timestamps are negative.
Also, media of course goes backwards. In other possible implementation,
filters would see normal forward playback, interrupted by seeks or
discontinuities. It turns out the current implementation of providing a
continuous backward media stream is probably better for filters.
Even deinterlacing seems to work. libavcodec always outputs fields in as
interleaved frames (i.e. fields are not reversed), and making up
timestamps for the new frames (when doubling the framerate) works
exactly like like in the forward case.
Actually the previous paragraph was a lie, and libavcodec does not
output fields as interleaved frames in rare cases. Sometimes AVFrame
contains single fields. In this case you'd need to inverse the field
dominance for deinterlacing filters to work correctly.
The way backward playback is implemented doesn't break basic assumptions
about timestamps after the decoder, so I guess all the encoding mode
needs to do is to adjust for the start offset, which it already does.
Though I might be wrong and my test was possibly flawed.
Stream recording on the other hand will fail immediately with
--record-file, and --stream-record will probably yield unexpected
results if any backstep seeks are done.
Make --audio-backward-overlap default to 2 for Opus. I have no idea why
this is needed. It seems to fix backward decoding though (going purely
by listening).
Normally, this should not be needed, since initial padding is completely
contained within the first packet (normally, and in the case I tested).
So the 2nd packet/frame should be fine, but for some unknown reason it
works only with the 3rd.
The only reasonable solution to this is probably to make discarding of
preroll frames based on timestmaps, instead of frame/packet count. But
then you get issues with video and its dumb timestamp reordering. So for
now, fuck it.
This seems more useful in general. This change also happens to fix a
miscounting of preroll packets when some of them were "rounded" away,
and which could make it stuck.
Also a simple intra-refresh encode with x264 (and muxed to mkv by it)
seems to work now. I guess I misinterpreted earlier results.
Backstepping still could get "stuck" if the demuxer didn't seek far back
enough. This commit fixes getting stuck if playing backwards from the
end, and audio has ended much earlier than the video.
In commit "demux: fix initial backward demuxing state in some cases",
I claimed that the backward seek semantics ("snapping" backward in
normal seeking, unrelated to backward playing) would take care of
this. Unfortunately, this is not always quite true.
In theory, a seek to any position (that does not use SEEK_FORWARD, i.e.
backward snapping) should return a packet for every stream. But I have a
mkv sample, where audio ends much earlier than video. Its mkvmerge
created index does not have entries for audio packets, so the video
index is used. This index has its last entry somewhere close after the
end of audio. So no audio packets will be returned. With a "too small"
back_seek_size, the demuxer will retry a seek target that ends up in
this place forever. (This does not happen if you use --index=recreate.
It also doesn't happen with libavformat, which always prefers its own
index, while mpv's internal mkv demuxer strictly prefers the index from
the file if it can be read.)
Fix this by adding the back_seek_size every time we fail to see enough
packets. This way the seek step can add up until it works.
To prevent that back_seek_pos just "runs away" towards negative infinity
by subtracting back_seek_size every time we back step to undo forward
reading (e.g. if --no-cache is used), readjust the back_seek_pos to the
lowest known resume position. (If the cache is active, kf_seek_pts can
be used, but to work in all situations, the code needs to grab the
minimum PTS in the keyframe range.)
Just rearranging shit. Setting SEEK_HR for backstep seeks actually
doesn't have much meaning, but disables the weird audio snapping for
"keyframe" seeks, and I don't know it's late.
This code used to be simpler, but now it's enough that it should be
factored into a single function.
Both uses of the new function are annoyingly different. The first use is
the special case when a decoder tries to read packets, but the demuxer
doesn't see any (like mp4 files with sparse video packets, which
actually turned out to be chapter thumbnail "tracks"). Then the other
stream queues will overflow, and the stream with no packets is marked
EOF to avoid stalling playback.
The second case is when the demxuer returns global EOF.
It would be more awkward to have the loop iterating the streams in the
function, because then you'd need a weird parameter to control the
behavior.
Just "mpv file.mkv --play-direction=backward" did not work, because
backward demuxing from the very end was not implemented. This is another
corner case, because the resume mechanism so far requires a packet
"position" (dts or pos) as reference. Now "EOF" is another possible
reference.
Also, the backstep mechanism could cause streams to find different
playback start positions, basically leading to random playback start
(instead of what you specified with --start). This happens only if
backstep seeks are involved (i.e. no cached data yet), but since this is
usually the case at playback start, it always happened. It was racy too,
because it depended on the order the decoders on other threads requested
new data. The comment below "resume_earlier" has some more blabla.
Some other details are changed.
I'm giving up on the "from_cache" parameter, and don't try to detect the
situation when the demuxer does not seek properly. Instead, always seek
back, hopefully some more.
Instead of trying to adjust the backstep seek target by a random value
of 1.0 seconds. Instead, always rely on the random value provided by the
user via --demuxer-backward-playback-step. If the demuxer should really
get "stuck" and somehow miss the seek target badly, or the user sets the
option value to 0, then the demuxer will not make any progress and just
eat CPU. (Although due to backward seek semantics used for backstep
seeks, even a very small seek step size will work. Just not 0.)
It seems this also fixes backstepping correctly when the initial seek
ended at the last keyframe range. (The explanation above was about the
case when it ends at EOF. These two cases are different. In the former,
you just need to step to the previous keyframe range, which was broken
because it didn't always react correctly to reaching EOF. In the latter,
you need to do a separate search for the last keyframe.)
Simple enough to do. May have mixed results. Typically, bitmap subtitles
will have a tight bounding box around the rendered text. But if for
example there is text on the top and bottom, it may be a single big
bitmap with a large transparent area between top and bottom. In
particular, DVD subtitles are really just a single screen-sized
RLE-encoded bitmap, though libavcodec will crop off transparent areas.
Like with sd_ass, you can't move subtitles _down_ if they are already in
their origin position. This could probably be improved, but I don't want
to deal with that right now.
Fixes the same thing as the previous commit did with demux_mkv. I'm not
sure if this is correct or a good idea (well, it works with my sample
file).
There are some shady things in this, but describing them would require
too many expletives.
In this scenario, the demuxer will output timestamps offset by the codec
delay (e.g. negative timestamps at the start; mkv simulates those), and
the trimming in the decoder (often libavcodec, but ad_lavc.c in our
case) will adjust the timestamps back (e.g. stream actually starts at
0).
This offset needs to be taken into account when seeking. This worked in
the uncached case. (demux_mkv.c is a bit tricky in that the index is
already in the offset space, so it compensates even though the seek call
does not reference codec_delay.) But in the cached case, seeks backwards
did not seek enough, and forward they seeked too much.
Fix this by adding the codec delay to the index search. We need to get
"earlier" packets, so e.g. seeking to position 0 really gets the initial
packets with negative timestamps.
This also adjusts the seek range start. This is also pretty obvious: if
the beginning of the file is cached, the seek range should start at 0,
not a negative value. We compare 0-based timestamps to it later on.
Not sure if this is the best approach. I also could have thought
about/checked some corner cases harder. But fuck this shit.
Not fixing duration (who cares) or end trimming, which would reduce the
seek range and duration (who cares).
This is a bad approach, and should be handled by a codec parameter field
(in mp_codec_params or AVCodecParameters).
It's bad because it's overly complicated, and has potential to break
demuxer cache assumptions: packets that were "intended" for seek
resuming may suddenly appear in the middle of a stream, when you seek
back and play a cached part again. (In general it was fine though,
because seek range joining tends to remove the first audio packet of the
next range when trying to find an overlap.)
demux_mkv.c does not try to export its codec_delay field through the
codec parameters mentioned above. In the only case I spotted this
element, the codec itself (opus) set this field within libavcodec. And I
think that's actually how it should be. On the other hand, a file could
in theory set this field via mkv headers if the codec is too stupid to
have such a field internally. But I don't really care until I see such a
file.
The end trimming is still sort of needed (though not sure if anything
uses it, other than the opus/mkv test sample I was using). The decoder
can't know whether something is the last packet, until it's too late.
The codec_delay field is still needed to offset timestamps.
Fixes stupid messages with a opus/mkv test file that had an absurdly
huge codec delay.
This file fully skips several frames at the start. ad_lavc.c trimmed
these frames to 0 samples and returned them. The next layer
(f_decoder_wrapper.c) saw discontinuous PTS values, because the PTS
values increased by a frame, but amounted to 0 audio samples. This was
harmless, but logged PTS discontinuity errors.
Not specifying a --start or using --start=100% with
--play-direction=backward usually does not work. The demuxer gets no
packets and immediately enters EOF state, which then hangs because
backward playback mode neither considers this mode, nor propagates the
EOF.
As far as demuxer implementations are concerned, this behavior is OK and
even wanted. Seeking near the end with SEEK_FORWARD set is allowed not
to return any packets (so a normal relative forward seek as done by the
user would end playback). Seeking exactly to the end or past it without
SEEK_FORWARD set is probably also sane.
Another vaguely related issue is that a backward seek during playback
start does not "establish" the demux position correctly: if stream A
hits the next keyframe and seeks back, while stream B has not had a
chance to read a packet yet, then stream B will never try to read from
the old position. The effect is that stream B (and thus playback) will
effectively miss the seek target. This is "random" because it depends on
the order and number of packet read calls made by the decoders.
Fixing this is probably hard, and requires extending the already complex
state machine with more states, so turn the manpage into a TODO list for
now.
Only timestamps that enter or leave the demuxer API should be adjusted
by ts_offset (which is usually the start time). queue_seek() is also
used by backward demux seeks, which uses an internal timestamp.
Raw audio formats can be accessed sample-wise, and logically audio
packets demuxed from it would contain only 1 sample. This is
inefficient, so raw audio demuxers typically "bundle" multiple samples
in one packet.
The problem for the demuxer cache and backward playback is that they
need properly aligned packets to make seeking "deterministic". The
requirement is that if you read some packets, and then seek back, you
eventually see the same packets again. demux_raw basically allowed to
seek into the middle of a previously returned packet, which makes it
impossible to make the transition seamless. (Unless you'd be aware of
the packet data format and cut them to make it seamless, which is too
complex for such a use case.)
Solve this by always aligning seeks to packet boundaries. This reduces
the seek accuracy to the arbitrarily chosen packet size. But you can use
hr-seek to fix this. The gain from not making raw audio an awful special
case pays in exchange for this "stupid" suggestion to use hr-seek.
It appears this also fixes that it could and did seek into the middle of
the frame (not sure if this code was ever tested - it goes back to
removing the code duplication between the former demux_rawaudio.c and
demux_rawvideo.c).
If you really cared, you could introduce a seek flag that controls
whether the seek is aligned or not. Then code which requires
"deterministic" demuxing could set it. But this isn't really useful for
us, and we'd always set the flag anyway, unless maybe the caching were
forced disabled.
libavformat's wav demuxer exhibits the same issue. We can't fix it (it
would require the unpleasant experience of contributing to FFmpeg), so
document this in otions.rst. In theory, this also affects seek range
joining, but the only bad effect should be that cached data is
discarded.
This is for uncompressed data, so every frame is a "keyframe". This is
part of making this demuxer work with the demuxer layer caching and
backward playback.
See manpage additions. This is a huge hack. You can bet there are shit
tons of bugs. It's literally forcing square pegs into round holes.
Hopefully, the manpage wall of text makes it clear enough that the whole
shit can easily crash and burn. (Although it shouldn't literally crash.
That would be a bug. It possibly _could_ start a fire by entering some
sort of endless loop, not a literal one, just something where it tries
to do work without making progress.)
(Some obvious bugs I simply ignored for this initial version, but
there's a number of potential bugs I can't even imagine. Normal playback
should remain completely unaffected, though.)
How this works is also described in the manpage. Basically, we demux in
reverse, then we decode in reverse, then we render in reverse.
The decoding part is the simplest: just reorder the decoder output. This
weirdly integrates with the timeline/ordered chapter code, which also
has special requirements on feeding the packets to the decoder in a
non-straightforward way (it doesn't conflict, although a bugmessmass
breaks correct slicing of segments, so EDL/ordered chapter playback is
broken in backward direction).
Backward demuxing is pretty involved. In theory, it could be much
easier: simply iterating the usual demuxer output backward. But this
just doesn't fit into our code, so there's a cthulhu nightmare of shit.
To be specific, each stream (audio, video) is reversed separately. At
least this means we can do backward playback within cached content (for
example, you could play backwards in a live stream; on that note, it
disables prefetching, which would lead to losing new live video, but
this could be avoided).
The fuckmess also meant that I didn't bother trying to support
subtitles. Subtitles are a problem because they're "sparse" streams.
They need to be "passively" demuxed: you don't try to read a subtitle
packet, you demux audio and video, and then look whether there was a
subtitle packet. This means to get subtitles for a time range, you need
to know that you demuxed video and audio over this range, which becomes
pretty messy when you demux audio and video backwards separately.
Backward display is the most weird (and potentially buggy) part. To
avoid that we need to touch a LOT of timing code, we negate all
timestamps. The basic idea is that due to the navigation, all
comparisons and subtractions of timestamps keep working, and you don't
need to touch every single of them to "reverse" them.
E.g.:
bool before = pts_a < pts_b;
would need to be:
bool before = forward
? pts_a < pts_b
: pts_a > pts_b;
or:
bool before = pts_a * dir < pts_b * dir;
or if you, as it's implemented now, just do this after decoding:
pts_a *= dir;
pts_b *= dir;
and then in the normal timing/renderer code:
bool before = pts_a < pts_b;
Consequently, we don't need many changes in the latter code. But some
assumptions inhererently true for forward playback may have been broken
anyway. What is mainly needed is fixing places where values are passed
between positive and negative "domains". For example, seeking and
timestamp user display always uses positive timestamps. The main mess is
that it's not obvious which domain a given variable should or does use.
Well, in my tests with a single file, it suddenly started to work when I
did this. I'm honestly surprised that it did, and that I didn't have to
change a single line in the timing code past decoder (just something
minor to make external/cached text subtitles display). I committed it
immediately while avoiding thinking about it. But there really likely
are subtle problems of all sorts.
As far as I'm aware, gstreamer also supports backward playback. When I
looked at this years ago, I couldn't find a way to actually try this,
and I didn't revisit it now. Back then I also read talk slides from the
person who implemented it, and I'm not sure if and which ideas I might
have taken from it. It's possible that the timestamp reversal is
inspired by it, but I didn't check. (I think it claimed that it could
avoid large changes by changing a sign?)
VapourSynth has some sort of reverse function, which provides a backward
view on a video. The function itself is trivial to implement, as
VapourSynth aims to provide random access to video by frame numbers (so
you just request decreasing frame numbers). From what I remember, it
wasn't exactly fluid, but it worked. It's implemented by creating an
index, and seeking to the target on demand, and a bunch of caching. mpv
could use it, but it would either require using VapourSynth as demuxer
and decoder for everything, or replacing the current file every time
something is supposed to be played backwards.
FFmpeg's libavfilter has reversal filters for audio and video. These
require buffering the entire media data of the file, and don't really
fit into mpv's architecture. It could be used by playing a libavfilter
graph that also demuxes, but that's like VapourSynth but worse.
The demuxer layer can start a thread to decouple the rest of the player
from blocking I/O (such as network accesses). But this particular
function does not support running with the thread enabled. The mutex use
within it is only since thread_work() may temporarily unlock the mutex,
and unlocking an unlocked mutex is not allowed. Most of the rest of the
code still does proper locking, even if it's pointless and effectively
single-threaded.
To make this look slightly cleaner, extend the mutex around the rest of
the code (like threaded code would have to do). This is mostly a
cosmetic change.
The demuxer cache benefits slightly from knowing where the current file
or stream begins. For example, seeking "left most" when the start is
cached would not trigger a low level seek (which would be followed by
messy range joining when it notices that the newly demuxed packets
overlap with an existing range).
Unfortunately, since multimedia is so crazy (or actually FFmpeg in its
quite imperfect attempt to be able to demux anything), it's hard to tell
where a file starts. There is no feedback whether a specific seek went
to the start of the file. Packets are not tagged with a flag indicating
they were demuxed from the start position. There is no index available
that could be used to cross-check this (even if the file contains a full
and "perfect" index, like mp4). You could go by the timestamps, but who
says streams start at 0? Streams can start somewhere at an extremely
high timestamps (transport streams like to do that), or they could start
at negative times (e.g. files with audio pre-padding will do that), and
maybe some file formats simply allow negative timestamps and could start
at any negative time. Even if the affected file formats don't allow it
in theory, they may in practice. In addition, FFmpeg exports a
start_time field, which may or may not be useful. (mpv's internal mkv
demuxer also exports such a field, but doesn't bother to set it for
efficiency and robustness reasons.)
Anyway, this is all a huge load of crap, so I decided that if the user
performs a seek command to time 0 or earlier, we consider the first
packet demuxed from each stream to be at the start of the file. In
addition, just trust the start_time field. This is the "shitty" part of
this commit.
One common case of negative timestamps is audio pre-padding. Demuxers
normally behave sanely, and will treat 0 as the start of the file, and
the first packets demuxed will have negative timestamps (since they
contain data to discard), which doesn't break our assumptions in this
commit. (Although, unfortunately, do break some other demuxer cache
assumptions, and the first cached range will be shown as starting at a
negative time.)
Implementation-wise, this is quite simple. Just split the existing
initial_state flag into two, since we want to deal with two separate
aspects. In addition, this avoids the refresh seek on track switching
when it happens right after a seek, instead of only after opening the
demuxer.
This is a minor benign hack that reorders the MPV_FORMAT_NODE output.
The order of members is not supposed to matter, but it's how the OSD
renders them as raw output. Normally this isn't used, but
demuxer-cache-state is a "prominent" case. Moving the seek ranges to the
end avoids that the more important other fields are not cut off by going
out of the screen on the bottom.
Also output the seek ranges in reverse. The order doesn't matter either
(as declared by input.rst). Currently, the demuxer orders them by least
recent use. Reversing it makes the most recently used range (the current
range) show up on top.
In other words, this commit does basically nothing but fudge stuff in a
cosmetic way to make debugging easier for me, and you've wasted your
time reading this commit message and the diff. Good.
This will properly notify observed properties if the player hasn't
started actual playback yet, such as with --demuxer-cache-wait.
This also happens to cause the main loop more often, which triggers
MPV_EVENT_IDLE, and fixes the OSC display. (See previous commit
message.)
The OSC's (osc.lua) event handling is fundamentally broken. It waits for
MPV_EVENT_TICK to update the UI, and MPV_EVENT_TICK has become entirely
meaningless, except as a hack for the OSC. There are many situations
where the OSC doesn't properly update because the TICK event it expects
isn't sent.
Fix one of them: it doesn't update the cache state if the VO window is
forced and --demuxer-cache-wait is used. Make it so that the tick event
is sent even if playback initialization is taking time.
This is still slightly broken, because it works only if the mainloop is
actually run, which depends on random circumstances (such as moving the
mouse over the VO window). The next commit will add another such
circumstance which will make it appear to work, although it's still
conceptually broken. If we "fixed" it and strictly woke up the player
if the idle timer ran out, we'd send tick events all the time, even
if nothing is going on, which we don't want. Fucking shitshow.
There were 3 packet reading functions: the "old" demux_read_packet()
that blocked (leftover from MPlayer times, but was still used until
recently by some obscure code), the "new" demux_read_packet_async(), and
the special demux_read_any_packet(), that is used by pseudo-demuxers
like demux_edl.
The first two could be used both in threaded and un-threaded mode. This
made 5 cases in total. Some bits of logic was spread across all of them.
Unify the logic. A recent commit made demux_read_packet() private, and
the code for it in threaded mode disappears. The difference between
threaded and un-threaded is minimized.
It's possible that this commit causes random regression. Enjoy.
There are 3 packet reading functions in the demux API, which all
function completely differently. One of them, demux_read_packet(), has
only 1 caller, which is in dec_sub.c. Change this caller to use
demux_read_packet_async() instead. Since it really wants to do a
blocking call, setup some proper waiting. This uses mp_dispatch_queue,
because even though it's overkill, it needs the least code.
In practice, waiting actually never happens. This code is only called on
code paths where everything is already read into memory (libavformat's
subtitle demuxers simply behave this way). It's still a bit of a
"coincidence", so implement it properly anyway.
If suubtitle decoder init fails, we still need to unset the demuxer
wakeup callback. Add a sub_destroy() call to the failure path. This also
happens to fix a missed pthread_mutex_destroy() call (in practice this
was a nop, or a memory leak on BSDs).
This is just redundant and slightly annoying, at least for normal
command line usage. If there are multiple entries, still print it
(because you want to know where you are). Also still print it if the
player was redirected (because you want to know where you got redirected
to).
I'm not sure about this, but it looks like a bug. If a stream didn't
have packets, but the joined range does, the stream should obviously
read the packets added by the joined range. Until now, due to
reader_head being NULL, reading was only resumed if a _new_ packet was
added by actual demuxing (in add_packet_locked()), which means the
stream would suddenly skip ahead, past the original end of the joined
range.
Change it so that it will pick up the new range.
Also, clear the skip_to_keyframe flag. Nothing useful can come from this
flag being set; in the first place, the first packet of a range (that
isn't the current range) should start with a keyframe. Some code
probably enforced it (although it's fuzzy).
Completely untested.
When doing a seek to the end of the cache, ds->skip_to_keyframe can be
set to true. Then some packets passed to add_packet_locked() may have to
be skipped. In some aspects, the skipped packet was still treated as if
it was going to be returned to the reader.
It almost doesn't matter though: it only caused a redundant wakeup_ds()
call, and could pass the packet to the stream recorder. Fix it anyway.
UB-sanitizer complains that we shift bits into the sign (when a is
used). Change it to unsigned, which in theory is more correct and
silences the warning.
Doesn't matter in practice, both the "bug" and the fix have 0 impact.
If a DASH-hack EDL has an init fragment is set, it opens the init
fragment as such to get the track layout (including codec etc.) and
avoids opening actual fragments until actual playback. It does not get
added to the source array, so it leaks on exit, which triggers an
obscure (but very justified) assertion in thread_tools.c:106. Fix the
leak by adding the additional demuxer instance to the sources arrays,
which gets it freed.
This is a regression from when I rewrote some of the timeline handling.
I decided that in order to make memory management slightly simpler,
freeing a timeline should only free elements in the sources array. That
is OK; I just didn't re-test with pseudo-DASH that has init fragments,
and just hit a video that uses that by accidents. These videos are
rather scarce (apparently) so it happened only now.
The real solution would probably be adding demuxer reference counting.
This EDL memory management is just too messy, and throwing refcounting
at such problems is an effective and popular fix. Then you'd get
debugging nightmares with incorrect refcounts too, though.
ytdl_hook.lua essentially uses these headers to implement parts of DASH.
Hopefully the FFmpeg DASH demuxer gets usable at some point, and/or mpv
gets a proper DASH demuxer. In any case, these EDL hacks could get
removed as soon as they get unnecessary and too annoying.
If you have a EDL stream with separate sources for audio and video
stream (like ytdl_hook now creates), you can get the problem that the
video stream seeks to a different position than audio due to different
key frame granularity.
In particular, if you seek backward, the video might undershoot the seek
target by a lot. Then video will resume from an earlier position than
audio, and the player plays silence. This is annoying.
Fix this by explicitly implementing a heuristic to detect separate
audio/video streams, determining where a video seek ends up, and then
seeking the audio stream to the video destination. This also makes sure
to not seek audio with SEEK_FORWARD, so it will always seek before the
video position. Non-precise seeks still skip audio to the video target,
so this helps with ensuring that audio is present at the final seek
target.
The implementation is very annoying, because the only way to determine
the seek target is to actually read a packet. Thus a 1-packet queue
needs to be added. In theory, we could get the seek target from the
index of the video file (especially if it's mp4), but libavformat does
not have public API that exports this index, so we're stuck with this
roundabout generic method.
Note that this is only for non-precise seeks. If precise seeks are done,
the problem is handled by the frontend by skipping unwanted video
frames. But non-precise seeking should still work. (Personally I prefer
non-precise seek mode by default because they're still significantly
faster.)
It also needs to be said that this is the 4th implementation of this
seek adjustment thing in mpv. The 1st implementation is in the frontend
(look for MPContext.seek_slave). This works only if the external audio
stream is known as such on the frontend level. The 2nd implementation is
in the demuxer level packet cache (top of execute_cache_seek()). This is
similar to code that any demuxer needs to handle non-precise seeks
sufficiently nicely. The 3rd is in demux_mkv.c. Since mkv is an
interleaved format, this implementation mostly consists on trying to
pick index entries for video packets if a video stream is selected.
Maybe these "redundant" implementations could be avoided by exposing
separate streams through the demuxer API (and making them individually
seekable) or something like this, but this is messy and not without
problems for multiple reasons. So for now this commit is the best way to
fix the observed behavior.
Instead of just using "edl/" for the file format, report mkv_oc if it's
generated from ordered chapters, "cue/" if from .cue, "multi/" if it's
from EDL but only for adding separate streams, "dash/" if it's from EDL
but only using the DASH hack, and "edl/" for everything else.
The EDL variants are mostly special-cased to the variants the ytdl
wrapper usually generates.
This has no effect other than what the command.c file-format property
returns.
Remove the singly linked list hack, replace it with a slightly more
proper data structure. This probably gets rid of a few minor bugs along
the way, caused by the awkward nonsensical sharing/duplication of some
fields.
Another change (because I'm touching everything related to timeline
anyway) is that I'm removing the special semantics for parts[num_parts].
This is now strictly out of bounds, and instead of using the start time
of the next/beyond-last part, there is an end time field now.
Unfortunately, this also requires touching the code for cue and mkv
ordered chapters. From some superficial testing, they still seem to
mostly work.
One observable change is that the "no_chapters" header is per-stream
now, which is arguably more correct, and getting the old behavior would
require adding code to handle it as special-case, so just adjust
ytdl_hook.lua to the new behavior.
I noticed that some ytdl streams have a start time other than 0. There's
currently no mechanism inside of the EDL stuff that determines this
start time correctly, so it can happen that if the start time is high,
demux_timeline.c tries to clip off the entire video and audio, resulting
in failure of playback.
As a counter measure, use the no_clip header, which entirely disables
clipping against time ranges in demux_timeline.c. (It's basically a
hack.)
Used by the next commit. It mostly exposes part of mp4_dash
functionality. It actually makes little sense other than for ytdl
special-use. See next commit.
E.g. "mpv null:// --demuxer=rawvideo" will "hang" by waiting for video
EOF forever. It's not signalled correctly because of the last-frame
corner case, which attempts to wait until the current frame is finally
displayed (which is signalled by whether a new frame can be queued, see
commit 1a339fa09d for some details). If no frame was ever queued, the VO
is not configured, and vo_is_ready_for_frame() never returns true.
Fix this by using vo_has_frame(), which seems to be exactly the correct
thing we need.
Init fragments are not a necessity for DASH, but this code assumed so.
Maybe the check was to prevent worse. But using normal EDL here leads to
very shitty behavior where it tries to open hundreds or thousands of
fragments, each with its own demuxer and HTTP connection. (This behavior
is fine for normal uses of EDLs, but completely unacceptable when
emulating fragmented streaming protocols. I'm not sure why the normal
EDL code is needed here, but I think someone claimed some obscure sites
just need it.)
This happens in the same situation as the one described in the previous
commit.
Otherwise we'd just use the base URL as media URL, which would fail with
a 404 error.
Not sure if there's a deeper reason why the audio path was explicitly
different from the video one. But this actually works now for a video
that returned fragmented DASH audio with the default format selection.
(This affects streams on that well known site of a big evil Silicon
Valley company. Typically happens after live stream gets converted to a
normal video, though after some time passes, this fragmented version is
deleted, and replaced by a non-fragmented one. I've observed this
several times and this seems to be the "normal" behavior.)
Normal EDL needs to clip packets coming from the underlying demuxer to
the segment range (including complicated stuff due to frame reordering).
This is unwanted In pseudo-DASH mode. A broken or subtly incorrect
manifest would lead to "bad stuff" happening. The intention of the
pseudo-DASH mode is to literally concatenate fragments.
This fixes that there were weird delay ("buffering") when seeking into
the last part of a seekable range. The exact case which triggers it if
SEEK_FORWARD is used, and the seek pts is after the second-last
keyframe, but before the end of the range. In that case,
find_seek_target() returned NULL, and the cache layer waited until the
_next_ keyframe the underlying demuxer returned until resuming playback.
find_seek_target() returned NULL, because the last keyframe had
kf_seek_pts unset. This field contains the lowest PTS in the packet
range from the keyframe until the next keyframe (or EOF). For normal
seeks, this is needed because keyframes don't necessarily have the
minimum PTS in the packet range, so it needs to be computed by waiting
for all packets until the next keyframe (or EOF).
Strictly speaking, this behavior was correct, but it meant that the
caller would set ds->skip_to_keyframe, which waits for the next newly
demuxed keyframe. No packets were returned to the decoder until this
happened, usually resulting in the frontend entering "buffering" mode.
What it really needs to do is returning the last keyframe in the cache.
In this situation, the seek target points in the middle of the last
completely cached packet range (as delimited by keyframes), and
SEEK_FORWARD is supposed to skip to the next keyframe. This is in line
with the basic assumptions the packet cache makes (e.g. the keyframe
flag means it's possible to start decoding, and the frames decoded from
it and following packets will strictly have PTS values above the
previous keyframe range). This means in this situation the kf_seek_pts
value doesn't matter either.
So fix this situation by explicitly detecting it and then returning the
last cached keyframe.
Should the search loop look at all packets, instead of only keyframe
ones? This would mean it can know that it's within the last keyframe
range (without looking at queue->seek_end). Maybe this would be a bit
more natural for the SEEK_FORWARD case, but due to PTS reordering it
doesn't sound like a useful thing to do.
Should skip_to_keyframe be checked by the code that sets kf_seek_pts to
a known value? This wouldn't help too much; the frontend would still go
into "buffering" mode for no reason until the packet range is completed,
although it would resume from the correct range.
Should a NULL return always unconditionally use keyframe_latest? This
makes sense because the seek PTS is usually already in the cached range,
so this is the only case that should happen. But there are scary special
cases, like sparse subtitle streams, or other uses of find_seek_target()
which could be out of range now or in future. Basically, don't "risk"
it.
One other potential problem with this is that the "adjust seek target"
code will be disabled in this case. It checks kf_seek_pts, and if it's
unset, the adjustment is not done. Maybe this could be changed to use
the queue's seek_end time, but I'm not sure if this is fully kosher. On
the other hand, I think the main use for this adjustment is with
backwards seeks, so this shouldn't matter.
A previous commit dealing with audio/video stream merging mentioned how
seeking forward entered "buffering" mode for unknown reasons; this
commit fixes this issue.
demux_timeline doesn't do any transport accesses itself. The slave
demuxers do this (these will actually access the stream layer and
perform e.g. network accesses). As a consequence, demux_timeline always
reported 0 bytes read, and network speed display didn't work.
Fix this by awkwardly reporting the amount of read bytes upwards. This
is not very nice, and requires explicit calls whenever the slave "might"
have read data.
Due to the way the reporting is done, it only works if the slaves do not
run demuxer threads, which makes things even less nice. (Fortunately
they don't anyway, because it would be a waste of resources.) Some
identifiers contain the word "hack" as a warning.
Some of the stupidity comes from the fact that demux.c itself resets the
stats randomly in order to calculate the bytes_per_second value, which
is useless for a slave, but of course is still done, because demux.c
itself is not aware of whether it's on the slave or top-level layer.
Unfortunately, this must do.
In theory, the demuxer thread/cache layer should be separated from
demuxer implementations. This would get rid of all the awkwardness and
nonsense. For example, the only threading involved would be the caching
layer, completely separate from demuxers themselves. It'd be the only
thing calculates speed rates for the player frontend, too (instead of
doing it for each demuxer, even if unused).
It was an ugly hack, and the next commit will make it even uglier.
Slightly reduce the ugliness to prevent death of too many brain cells,
though it's still an ugly hack.
The cleanup is really minor, but I guess the following commit would be
much worse otherwise. In particular, this commit checks accesses
(instead of having a public field with evil access rules), which should
avoid misunderstandings and incorrect use. Strictly speaking, the added
field is redundant, but the next commit complicates it a bit.
I think this is better. On the other hand, this is a behavior change.
The EDL "spec" says that unknown fields are igored. But strictly
speaking, unknown headers are not "fields", but unknown entities.
EDL "headers" were always an afterthought, and kind of hacked on top of
the existing code. Improve it slightly, and make it follow the
conventions of the normal parsing. Basically use the same code structure
for them, just that they use different field names.
This merges separate audio and video tracks into one virtual stream,
which helps the mpv caching layer. See previous EDL commit for details.
It's apparently active for most of evil Silicon Valley giant's streaming
videos.
Initial tests seem to work fine, except it happens pretty often that
playback goes into buffering immediately even when seeking within a
cached range, because there is not enough forward cache data yet to
fully restart playback. (Or something like this.)
The audio stream title used to be derived from track.format_note; this
commit stops doing so. It seemed pointless anyway. If really necessary,
it could be restored by adding new EDL headers.
Note that we explicitly don't do this with subtitle tracks. Subtitle
tracks still have a chance with on-demand loading or loading in the
background while video is already playing; merging them with EDL would
prevent this. Currently, subtitles are still added in a "blocking"
manner, but in theory this could be loosened. For example, the Lua API
already provides a way to run processes asynchronously, which could be
used to add subtitles during playback. EDL will probably be never
flexible enough to provide this. Also, subtitles are downloaded at
once, rather than streamed like audio and video.
Still missing: disabling EDL's pointless chapter generation, and
propagating download speed statistics through the EDL wrapper.
The ytdl wrapper can resolve web links to playlists. This playlist is
passed as big memory:// blob, and will contain further quite normal web
links. When playback of one of these playlist entries starts, ytdl is
called again and will resolve the web link to a media URL again.
This didn't work if playlist entries resolved to EDL URLs. Playback was
rejected with a "potentially unsafe URL from playlist" error. This was
completely weird and unexpected: using the playlist entry directly on
the command line worked fine, and there isn't a reason why it should be
different for a playlist entry (both are resolved by the ytdl wrapper
anyway). Also, if the only EDL URL was added via audio-add or sub-add,
the URL was accessed successfully.
The reason this happened is because the playlist entries were marked as
STREAM_SAFE_ONLY, and edl:// is not marked as "safe". Playlist entries
passed via command line directly are not marked, so resolving them to
EDL worked.
Fix this by making the ytdl hook set load-unsafe-playlists while the
playlist is parsed. (After the playlist is parsed, and before the first
playlist entry is played, file-local options are reset again.) Further,
extend the load-unsafe-playlists option so that the playlist entries are
not marked while the playlist is loaded.
Since playlist entries are already verified, this should change nothing
about the actual security situation.
There are now 2 locations which check load_unsafe_playlists. The old one
is a bit redundant now. In theory, the playlist loading code might not
be the only code which sets these flags, so keeping the old code is
somewhat justified (and in any case it doesn't hurt to keep it).
In general, the security concept sucks (and always did). I can for
example not answer the question whether you can "break" this mechanism
with various combinations of archives, EDL files, playlists files,
compromised sites, and so on. You probably can, and I'm fully aware that
it's probably possible, so don't blame me.
This commit adds an extension to mpv EDL, which basically allows you to
do the same as --audio-file, --external-file, etc. in a single EDL file.
This is a relatively quick & dirty implementation. The dirty part lies
in the fact that several shortcuts are taken. For example, struct
timeline now forms a singly linked list, which is really weird, but also
means the other timeline using demuxers (cue, mkv) don't need to be
touched. Also, memory management becomes even worse (weird object
ownership rules that are just fragile WTFs). There are some other
dubious small changes, mostly related to the weird representation of
separate streams.
demux_timeline.c contains the actual implementation of the separate
stream handling. For the most part, most things that used to be on the
top level are now in struct virtual_source, of which one for each
separate stream exists. This is basically like running multiple
demux_edl.c in parallel. Some changes could strictly speaking be split
into a separate commit, such as the stream_map type change.
Mostly untested. Seems to work for the intended purpose. Potential for
regressions for other timeline uses (like ordered chapters) is probably
low. One thing which could definitely break and which I didn't test is
the pseudo-DASH fragmented EDL code, of which ytdl can trigger various
forms in obscure situations. (Uh why don't we have a test suite.)
Background:
The intention is to use this for the ytdl wrapper. A certain streaming
site from a particularly brain damaged and plain evil Silicon Valley
company usually provides streams as separate audio and video streams.
The ytdl wrapper simply does use audio-add (i.e. adding it as external
track, like with --audio-file), which works mostly fine. Unfortunately,
mpv manages caching completely separately for external files. This has
the following potential problems:
1. Seek ranges are rendered incorrectly. They always use the "main"
stream, in this case the video stream. E.g. clicking into a cached range
on the OSC could trigger a low level seek if the audio stream is
actually not cached at the target position.
2. The stream cache bloats unnecessarily. Each stream may allocate the
full configured maximum cache size, which is not what the user intends
to do. Cached ranges are not pruned the same way, which creates disjoint
cache ranges, which only use memory and won't help with fast seeking or
playback.
3. mpv will try to aggressively read from both streams. This is done
from different threads, with no regard which stream is more important.
So it might happen that one stream starves the other one, especially if
they have different bitrates.
4. Every stream will use a separate thread, which is an unnecessary
waste of system resources.
In theory, the following solutions are available (this commit works
towards D):
A. Centrally manage reading and caching of all streams. A single thread
would do all I/O, and decide from which stream it should read next. As
long as the total TCP/socket buffering is not too high, this should be
effective to avoid starvation issues. This can also manage the cached
ranges better. It would also get rid of the quite useless additional
demuxer threads. This solution is conceptually simple, but requires
refactoring the entire demuxer middle layer.
B. Attempt to coordinate the demuxer threads. This would maintain a
shared cache and readahead state to solve the mentioned problems
explicitly. While this sounds simple and like an incremental change,
it's probably hard to implement, creates more messy special cases,
solution A. seems just a better and simpler variant of this. (On the
other hand, A. requires refactoring more code.)
C. Render an intersection of the seek ranges across all streams. This
fixes only problem 1.
D. Merge all streams in a dedicated wrapper demuxer. The general demuxer
layer remains unchanged, and reading from separate streams is handled as
special case. This effectively achieves the same as A. In particular,
caching is simply handled by the usual demuxer cache layer, which sees
the wrapper demuxer as a single stream of interleaved packets. One
implementation variant of this is to reuse the EDL infrastructure, which
this commit does.
All in all, solution A would be preferable, because it's cleaner and
works for all external streams in general.
Some previous commit tried to prepare for implementing solution A. This
could still happen. But it could take years until this is finally
seriously started and finished. In any case, this commit doesn't block
or complicate such attempts, which is also why it's the way to go.
It's worth mentioning that original mplayer handles external files by
creating a wrapper demuxer. This is like a less ideal mixture of A. and
D. (The similarity with A. is that extending the mplayer approach to be
fully dynamic and without certain disadvantages caused by the wrapper
would end up with A. anyway. The similarity with D. is that due to the
wrapper, no higher level code needs to be changed.)
This warns about player/audio.c:253 with gcc 8.2.0. Although this
warning could be useful to check the worst case estimation, the compiler
doesn't explain how it gets its dumb, bogus result, so this is useless.
You'd just end up trying to make the compiler happy for no reason.
struct stream used to include the stream buffer, including peek buffer,
inline in the struct. It could not be resized, which means the maximum
peek size was set in stone. This meant demux_lavf.c could peek only so
much data.
Change it to use a dynamic buffer. Because it's possible, keep the
inline buffer for default buffer sizes (which are basically always used
outside of file opening). It's unknown whether it really helps with
anything. Probably not.
This is also the fallback plan in case we need something like the old
stream cache in order to deal with mp4 + unseekable http: the code can
now be easily changed to use any buffer size.
The only thing left is the notification for track switching. Just get
rid of that.
There's probably no real reason to get rid of control(), but why not. I
think I was actually trying to do some real work but fuck that.
Subtitles (and a few other file types, like playlists) are not streamed,
but fully read on opening. This means keeping the file handle or network
socket open is a waste of resources and could cause other weird
behavior. This is why there's a hack to close them after opening.
Change this hack to make the demuxer itself do this, which is less
weird. (Until recently, demuxer->stream ownership was more complex,
which is why it was done this way.)
There is some evil shit due to a huge ownership/lifetime mess of various
objects. Especially EDL (the currently only nested demuxer case)
requires being careful about mp_cancel and passing down stream pointers.
As one defensive programming measure, stop accessing the "stream"
variable in open_given_type(), even where it would still work. This
includes removing a redundant line of code, and removing the peak call,
which should not be needed anymore, as the remaining demuxers do this
mostly correctly.
I always wanted to get rid of this, because it makes the ownership rules
for the stream pointer really awkward. demux_edl.c was the only
remaining user of this. Replace it with a semi-clever idea: the init
segment shit can be used to pass the "file" contents as memory block,
and "memory://" itself provides an empty stream. I have no idea if this
actually works, because I didn't immediately find a test stream (would
have to be some youtube DASH shit).
Instead of going through those weird DEMUXER_CTRLs, query this
information directly. I'm not sure which kind of brain damage made me
use CTRLs for these. Since there are no other DEMUXER_CTRLs that make
sense for the frontend, remove the remaining infrastructure for them
too.
The stream size return was the only thing that still required doing
STREAM_CTRLs from frontend through the demuxer layer. This can be done
much easier, so rip it out. Also rip out the now unused infrastructure
for STREAM_CTRLs via demuxer layer.
Apparently this was so that when playing a video file from a .rar file,
it would load external subtitles with the same name (instead of looking
for mpv's rar:// mangled URL). This was requested on github almost 5
years ago. Seems like a weird feature, and I don't care. Drop it,
because it complicates some in progress change.
This code set pkt->stream to a value which I'm not sure whether it's
correct. A recent commit overwrote it with a value that is definitely
correct.
There appears to be an off by one error. No fucking clue whether this
was somehow correct, but applying an apparent fix does not seem to break
anything, so whatever.
EDLs can be provided either as external file, or "inline" as a big
edl:// URL. There is no difference between them, except if it's loaded
from an external file, there is some weird filename sanitation going on
(see fix_filenames() in demux_edl.c). It seems this is intended to be a
security mechanism, but probably makes no sense at all.
Note that playlists are allowed to access anything locally. One
difference to playlists is that the EDL code lacks the "security"
mechanism when accessing playlist entries (see handling of the
playlist_entry.stream_flags field - EDL would need something similar),
so don't remove that, as I'm unaware of the exact consequences.
libarchive (sometimes affectionately called libcve) has this annoying
behavior that if after a "fatal" error, you do any operation on the
archive context other than querying the error and closing the context,
you get a free CVE. So we close the archive context in these situations.
This can set p->mpa to NULL, so code accessing this field needs to be
careful.
This was not considered in a certain code path, and a simple truncated
.rar file made it crash. Part of the problem was that the file inside
the rar was a mkv file, which triggered seeking when the demux_mkv
resync code encountered bogus data.
This is probably a regression from a relatively recent change to this
code (in any case mpv 0.29.1 doesn't crash).
Fix this by adding the check.
There's also a mechanism to reopen an archive context used to emulate
seeking, since most libarchive format handlers don't support this
natively. Add a reopen call to the codepath, because obviously it should
always be possible to seek back into a "working" area of the file.
There is a second bug with this: if reopening fails, we don't adjust the
current position back to 0, which in some cases means we accidentally
return bogus data to the reader when we shouldn't. Fix this by always
resetting the position on reopening.
It would always autodetect it based on the passed style block,
but as we are defining it - we might as well define it always.
(As far as I can see all decoders in libavcodec utilize 4+ style
blocks)
This one is probably not terribly obvious from just the valgrind log,
but a wayland dev explained it to me just a second ago. Whenever mpv
sends events to the screen with wl_display_dispatch, wayland internally
allocates memory to a struct wl_proxy object if a new id is found. Quite
a few more things happen to that proxy object, but eventually mpv stores
the data on the client-side in a wrapper type of struct (struct
wl_data_offer). mpv's data_device_listener keeps track of those proxies
and frees the memory when appropriate. Of course, mpv is constantly
sending events to the screen and does so until the user quits the
player. What happens here is that one final wl_display_dispatch is called
right before the user quits the player and before mpv's
data_device_listener can handle that object. So the result is that you
always have one extra dangling proxy that doesn't get properly freed.
The solution is to just simply call wl_data_offer_destroy before closing
the wl_display to free that final dangling wl_proxy.
Extending the client-allocated mpv_opengl_drm_params struct
constituted a break of ABI that could cause UB.
Create a clean break by deprecating "drm_params" and related structs
and enum values, and replacing it with "drm_params_v2".
Also fix some comments and code that wrongly assumed that open could
return any other negative number than -1 for failure.
This commit updates the libmpv version to 1.104
The intent of the line above the modified one code was raising the
amount of read data, so that many stream_peek() calls with small len
values would not degrade performance by effectively turning every
stream_peak() into an unbuffered read call to the stream implementation.
So this confusing looking MPMAX() was correct, but "chunk" could still
get beyond the buffer.
So just fix that and limit "chunk" correctly.
I'm not sure whether the commit referenced below accidentally removed
some intricate guarantee that this couldn't happen, since the code was
around since 2013. It could have relied on TOTAL_BUFFER_SIZE >
STREAM_BUFFER_SIZE. But not sure. I've rewritten all this code in my own
branch a year ago, so who knows.
Fixes: 162e0f5ad9Fixes: #6948
Like hwdec_cuda, you get a big #ifdef mess if you try and keep the
OpenGL and Vulkan interops in the same file. So, I've refactored
them into separate files in a similar way.
Originally, vo_gpu/vo_opengl considered the case of Nvidia proprietary
drivers, which required vdpau/GLX, and Intel open source drivers, which
require vaapi/EGL. Since window creation and GPU context creation are
inseparable in mpv's internal API, it had to pick the correct API very
early, or hardware decoding wouldn't work. "x11probe" was introduced for
this reason. It created a GLX context (without showing the window yet),
and checked whether vdpau was available. If yes, it used GLX, if not, it
continued probing x11/EGL. (Obviously it couldn't always fail on GLX
without vdpau, which is why it was a separate "probe" backend.)
Years passed, and now the situation is different. Vdpau is dead. Nvidia
drivers and libavcodec now provide CUDA interop, which requires EGL, and
fixes some of the vdpau problems. AMD drivers now provide vaapi, which
generally works better than vdpau. Intel didn't change.
In particular, vaapi provides working HEVC Main10 support. In theory, it
should work on vdpau too, with quality reduction (no 10 bit surfaces),
but I couldn't get it to work.
So always prefer EGL. And suddenly hardware decoding works. This is
actually rather important, because HEVC is unfortunately on the rise,
despite shitty encoders and unoptimized decoders. The latter may mean
that hardware decoding works better than libavcodec.
This should have been done a long, long time ago.
In some cases, src.sig_peak remains undefined as 0, which was definitely
the case when using the OSD, since it never got passed through the usual
color space normalization process. Most robust work-around is to simply
force the normalization at the site where it's needed. This ensures this
value is always valid and defined, to make the peak-dependent logic in
these two functions always work.
Fixes 4b25ec3a9dFixes#6917Fixes#6918
The write functionality is almost unused (only encoding 2-pass mode uses
it to write the log file). Moreover, it almost makes no sense to use
this in a not local scenario. This change is just to prevent people from
duplicating the short write logic across all streams that happen to
support writing. Mostly untested; local log file writing still works.
The "program" property could switch between TS programs. It was rather
complex and rather obscure (even if you deal with TS captures, you
usually don't need it). If anyone actually needs it (did anyone ever
attempt to even use it?), it should be rewritten. The demuxer should
export a program list, and the frontend should handle the "cycling"
logic.
Linux analog TV support (via tv://) was excessively complex, and
whenever I attempted to use it (cameras or loopback devices), it didn't
work well, or would have required some major work to update it. It's
very much stuck in the analog past (my favorite are the frequency tables
in frequencies.c for analog TV channels which don't exist anymore).
Especially cameras and such work fine with libavdevice and better than
tv://, for example:
mpv av://v4l2:/dev/video0
(adding --profile=low-latency --untimed even makes it mostly realtime)
Adding a new input layer that targets such "modern" uses would be
acceptable, if anyone is interested in it. The old TV code is just too
focused on actual analog TV.
DVB is rather obscure, but has an active maintainer, so don't remove it.
However, the demux/stream ctrl layer must go, so remove controls for
channel switching. Most of these could be reimplemented by using the
normal method for option runtime changes.
This was possibly needed by libdvdread, and/or old CD drivers on some
system. It still works with on-filesystem DVD and BD test images, so
this can go.
This removes anything related to DVD/BD/CD that negatively affected the
core code. It includes trying to rewrite timestamps (since DVDs and
Blurays do not set packet stream timestamps to playback time, and can
even have resets mid-stream), export of chapters, stream languages,
export of title/track lists, and all that.
Only basic seeking is supported. It is very much possible that seeking
completely fails on some discs (on some parts of the timeline), because
timestamp rewriting was removed.
Note that I don't give a shit about optical media. If you want to watch
them, rip them. Keeping some bare support for DVD/BD is the most I'm
going to do to appease the type of lazy, obnoxious users who will care.
There are other players which are better at optical discs.
stream_dvd.c contained large amounts of ancient, unmaintained code,
which has been historically moved to libdvdnav. Basically, it's full of
low level parsing of DVD on-disc structures.
Kill it for good. Users can use the remaining dvdnav support (which
basically operates in non-menu mode). Users have reported that
libdvdread sometimes works better, but this is just libdvdnav's problem
and not ours.
Semantics changes are the same as at 548ef078 .
Also, the previous C implementation returnd a string for the `stdout`
value, but stdout of the subprocess command is MPV_FORMAT_BYTE_ARRAY
which js previously didn't support, so support it too (at pushnode)
by returning it as a string - the same as the lua code does.
There were some cases where a js number (double) was blindly casted to
int or uint64, but that can be undefined behavior (out of range to int)
or wrong (negative to uint).
Now the code throws a js error if the value is out of range.
Additionally, commit ec625266 added these checks for the new hooks API,
but incorrectly tested int64 range rather than uint64. Fix this too.
Mesa supports the EGL_CHROMIUM_sync_control extension, and it's
available out of the box with AMD drivers. In practice, this is exactly
the same as GLX_OML_sync_control, but for EGL. The extension
specification is separate from the GLX one though, and buried somewhere
in the Chromium code.
This appears to work, although I don't know if it really works.
In theory, this could be useful for other EGL targets. Support code for
it could have been added to egl_helpers.c to avoid some minor duplicated
glue code if another EGL target were to provide this extension. I didn't
bother with that. ANGLE on Windows can't support it, because the
extension spec. explicitly requires POSIX timers. ANGLE on Linux/OSX is
actively harmful for mpv and hopefully won't ever use it. Wayland uses
EGL, but has its own fancy presentation feedback stuff (and besides, I
don't think basic video player functionality works on Wayland at all).
context_drm_egl maybe? But I think DRM has its own stuff.
So the next commit can make EGL use it. EGL has a quite similar
function, that practically works the same. Although it's relatively
trivial, it's still tricky, and probably shouldn't end up as duplicated
code.
There are no functional changes, except initialization, and how failure
of the glXGetSyncValues call is handled. Also, some comments mention the
EGL extension.
Note that there's no intention for this code to handle anything else
than the very specific OML sync extension (and its EGL equivalent). This
is just too weirdly specific to the weird idiosyncrasies of the
extension, and it makes no sense to extend it to handle anything else.
(Such as Wayland or DXGI presentation feedback.)
In the past, src peak was always equal to or higher than dst peak. But
since `--target-peak` got introduced, this could no longer be the case.
This leads to an incorrect result (scaling for peak mismatch in gamma
light) unless some other option (CMS, --linear-scaling, etc.) forces the
linearization.
Fixes#6533
While they accept the frequency field with MHz for DVB-S,
for DVB-C and DVB-T, it may be in Hz, kHz or MHz.
The official rule is to multiply whatever is in the channels.conf
by 1000 until a value > 1000000 is reached to get correct units for tuning.
* Adds a script to clone and build FFmpeg as well as
to configure and build mpv itself. Currently only used
for macOS and contain hard-coded macOS specific options.
* Still works with the Linux containers.
* Moves our language back to "c" from "generic"
* Defines our Linux distribution as "bionic" to get the latest
Ubuntu base distribution to be the runner for our containers.
* Adds the homebrew add-on for macOS package installation for
dependencies. Installs everything required but FFmpeg, as we want
to have our own FFmpeg snapshots.
We collect a 'vulkan-device' option today but then don't actually
pass it on, so it's useless. Once that's fixed, it can be used
to select a specific vulkan device by name.
Tested with the new nvidia offload feature to select between the
nvidia and intel GPUs.
Somehow I got the idea that compound literals had function-scoped
lifetime. Instead, like all other objects with automatic storage
duration, compound literals are block-scoped, so they become invalid
after exiting the block they were declared in. It seems like a recent
change to GCC actually reuses the memory that the compound literals
used to occupy, which was causing a few bugs.
The pattern of conditionally assigning a pointer to a compound literal
was used in a few places in ra_d3d11 where the Direct3D API expects
either a pointer to an initialised struct or NULL. Change these to
ensure the lifetime of the struct includes the API call.
Should fix#6775.
This is documented as required (although we did not do it in
the old GL codepath, with no visible problems) and I have seen
transient artifacts after seeking which _appear_ to have gone
away after introducing this.
this migrates our current swift code to version 5 and 4. building is
support from 10.12.6 and xcode 9.1 onwards.
dynamic linking is the new default, since Apple removed static libs
from their new toolchains and it's the recommended way.
additionally the found macOS SDK version is printed since it's an
important information for finding possible errors now.
Fixes#6470
this is in preparation for the upcoming swift 5 transition, where static
linking was replaced by dynamic linking the swift libraries as the
preferred way, by Apple. furthermore Apple removed the static swift libs
from their dev Tools starting with xcode 10.2/swift 5.
because of ABI incompatibility dynamic linking for swift versions prior
to 5 doesn't use the system lib path for the dynamic swift libs.
for now static linking is still the default, but that will be changed
when swift 5 support is added and swift 3 support is dropped.
Fixes#6232
the xcode-select tool only properly works with Apple provided toolchains
but not with third party ones from swift.org. in the latter case the
swift compiler executable is found in the proper path but the swift libs
from the xcode or command line tools will be picked. this leads to a
not wanted discrepancy of the swift compiler and libs and possible
errors.
instead of relying on the xcode-select tool search for the libs relative
to the swift executable. that relative path seems to be the same for all
toolchains. if for any reasons a swift executable is not found in the
relative path, fall back to the old xcode-select method.
furthermore, both static and dynamic libs will be searched for but only
the former will be used for now. this is a preparation for the upcoming
swift 5 migration.
We saw a segfault when trying to use the intel-media-driver (iHD)
rather than the normal intel va driver. This happened because the
iHD driver reports P010 (and maybe other formats) with multiple
layers to represent the interleaved UV plane. The normal va driver
reports one UV layer to match the plane.
This threw off my logic which assumed that the number of layers
could not exceed the number of planes.
There's a way one could fix this in a fully generalised form, but
I'm just going to do what the EGL path does and assume that:
* Layer 'n' is on Plane 'n' for n < total number of planes
* These layers always start at offset 0 on the plane
You can imagine ways that these assumptions are violated, but at
least the failure will look the same for both EGL and Vulkan
paths.
Today, we normally see a format error when probing because yuyv422
cannot be used, but it's in the normal set of probed formats.
This error is distracting and confusing, so only log probing errors
at the VERBOSE level.
Fixes#6411
This change introduces a vulkan interop path for the vaapi hwdec.
The basic principles are mostly the same as for EGL, with the
exported dma_buf being imported by Vukan. The biggest difference
is that we cannot reuse the texture as we do with OpenGL - there's
no way to rebind a VkImage to a different piece of memory, as far
as I can see. So, a new texture is created on each map call.
I did not bother implementing a code path for the old libva API as
I think it's safe to assume any system with a working vulkan driver
will have access to a newer libva.
Note that we are using separate layers for the vaapi surface, just
as is done for EGL. This is because libplacebo doesn't support
multiplane images.
This change does not include format negotiation because no driver
implements the vk_ext_image_drm_format_modifier extension that
would be required to do that. In practice, the two formats we care
about (nv12, p010) work correctly, so we are not blocked. A separate
change had to be made in libplacebo to filter out non-fatal validation
errors related to surface sizes due to the lack of format negotiation.
New releases of VDPAU support decoding 4:4:4 content, and that comes
back as NV24 when using 'direct mode' in OpenGL Interop. That means we
need to be a little bit smarter about how we set up the OpenGL
textures.
If the compositor sends a configure event before the surface is initially
mapped, resize gets called before the egl_window gets created, resulting
in a crash in wl_egl_window_resize.
This was fixed back in 618361c697, but was reintroduced when the wayland
code was rewritten in 68f9ee7e0b.
While `ra` supports the concept of a texture as a storage
destination, it does not support the concept of a texture format
being usable for a storage texture. This can lead to us attempting
to create a texture from an incompatible format, with undefined
results.
So, let's introduce an explicit format flag for storage and use
it. In `ra_pl` we can simply reflect the `storable` flag. For
GL and D3D, we'll need to write some new code to do the compatibility
checks. I'm not going to do it here because it's not a regression;
we were already implicitly assuming all formats were storable.
Fixes#6657
Obvious mistake. This reported 44 bytes more data than what was
available. Could cause out of bounds reads. Security researchers would
claim a major victory if they found something like this in more popular
software, and would create a website for it.
Support for Ada and Objective-C was removed from MSYS2, which made
pacman refuse to update GCC while the gcc-ada and gcc-objc packages were
installed. Remove those packages before updating the others. Also remove
ANGLE, which has been removed from MSYS2, and add libplacebo, which is
now needed for the Vulkan VO.
This started as a desperate attempt to lower the memory requirement
of error diffusion, but later it turns out that this change also
improved the rendering performance a lot (by 40% as I tested).
Errors was stored in three uint before this change, each with 24bit
precision. This change encoded them into a single uint, each with 8bit
precision. This reduced the shared memory usage, as well as number of
atomic operations, all by three times.
Before this change, with the minimum required 32kb shared memory, only
the `simple` kernel can be used to render 1080p video, which is mostly
useless compare to `--dither=fruit`. After this change, 32kb can
handle `burkes` kernel for 1080p, or `sierra-lite` for 4K resolution.
error diffusion requires two texture rendering pass. The existing code
reuses `screen_tex` and creates another for such purpose. This works
generally well for opengl, but could potentially be problematic for
vulkan, due to its async natural.
This is a straightforward parallel implementation of error diffusion
algorithms in compute shader. Basically we use single work group with
maximal possible size to process the whole image. After a shift
mapping we are able to process all pixels column by column.
A large ring buffer are allocated in shared memory to speed things up.
However the size of required shared memory depends linearly on the
height of video window (or screen height in fullscreen mode). In case
there is no enough shared memory, it will fallback to `--dither=fruit`.
The maximal allowed work group size is hardcoded as 1024. Ideally we
could query `GL_MAX_COMPUTE_WORK_GROUP_INVOCATIONS`. But for whatever
reason, it seems most high end card from nvidia and amd support only
the minimal required value, so I guess we can stick to it for now.
When the D3D11 backend was first written, SPIRV-Cross only had a C++ API
and no guarantee of API or ABI stability, so instead of using
SPIRV-Cross directly, mpv used an unofficial C wrapper called crossc.
Now that KhronosGroup/SPIRV-Cross#611 is resolved, SPIRV-Cross has an
official C API that can be used instead, so remove crossc and use
SPIRV-Cross directly.
The calculation of scale factor involves 32-bit float, and a strict
equality test will effectively ignore `--scaler-resizes-only` option
for some non-integer scale factor.
Fix this by using non-strict equality check.
since the none native fs is a special legacy case it needs a special
quit routine. it indefinitely waited for an exit fs screen event to
shutdown properly, though that event only fires for the native fs.
now we check if we really are using a native fullscreen and if not
shutdown immediately.
Fixes#6704
These were unnecessary for a couple of reasons, but it seems like the
old code went through a lot of effort to avoid duplicating the code to
print a RECT, even though the windowrc gets printed anyway at the end of
the function.
Avoid printing the same windowrc twice by only printing it when it gets
changed (in the w32->current_fs branch.)
I assume (but cannot confirm) that VA-AP-API is in fact a typo, because
most if not all search engine results related to it are from mpv's manual
page.
By changing this to VA-API and clarifying that this requires VA-API support
on a system to use it, we can hopefully make it clear to unsuspecting
Windows users that this is not the filter they're looking for.
Concerns #6690.
This allows to select the drm mode using a string specification. You
can either select the the preferred mode, the mode with the highest
resolution, by specifying WxH[@R] or by its index in the list of modes
as before.
This was implemented by using OPT_STRING_VALIDATE for drm-mode,
instead of OPT_INT. Using a string here also prepares for future
additions to drm-mode that aim to allow specifying a mode by its
resolution.
It is useful when debugging to be able to force atomic off, or as a
workaround if atomic breaks for some user. Legacy modesetting is less
likely to break by virtue of being a less complex API.
The amount of code now present that's specific to Vulkan or OpenGL
has reached the point where we really want to split it out to
avoid a mess of #ifdefs.
At the same time, I'm moving the code to an api neutral location.
the force unwrapping of optionals caused many unpredictable segfaults
instead of gracefully exiting or falling back. besides that, it is bad
practice and the code is a lot more stable now.
This change updates the vulkan interop code to work with the
libplacebo based ra_vk, but also introduces direct VkImage
sharing to avoid the use of the intermediate buffer.
It is also necessary and desirable to introduce explicit
semaphore bsed synchronisation for operations on the shared
images.
Synchronisation means we can safely reuse the same VkImage for every
mapped frame, by ensuring the frame is copied to the VkImage before
mapping the next frame.
This functionality requires a 417.xx or newer nvidia driver, due to
bugs in the VkImage interop in the earlier 411 and 415 drivers.
It's definitely worth the effort, as the raw throughput is about
twice that of implementation using an intermediate buffer.
When interacting directly with libplacebo, we may need to pass a
pl_fmt based on an ra_format. Although the mapping is currently
trivial, it's worth wrapping to make it easy to adapt if this
changes in the future.
This commit rips out the entire mpv vulkan implementation in favor of
exposing lightweight wrappers on top of libplacebo instead, which
provides much of the same except in a more up-to-date and polished form.
This (finally) unifies the code base between mpv and libplacebo, which
is something I've been hoping to do for a long time.
Note: The ra_pl wrappers are abstract enough from the actual libplacebo
device type that we can in theory re-use them for other devices like
d3d11 or even opengl in the future, so I moved them to a separate
directory for the time being. However, the rest of the code is still
vulkan-specific, so I've kept the "vulkan" naming and file paths, rather
than introducing a new `--gpu-api` type. (Which would have been ended up
with significantly more code duplicaiton)
Plus, the code and functionality is similar enough that for most users
this should just be a straight-up drop-in replacement.
Note: This commit excludes some changes; specifically, the updates to
context_win and hwdec_cuda are deferred to separate commits for
authorship reasons.
This helps with compatibility and/or performance in particular for
oddly-sized formats like rgb24. We use a loop to avoid having to
calculate the lcm (or waste bytes in the extremely common case of the
byte size and the stride align having shared factors).
Shaderc comes with a Python script that automatically fetches
"known-good" versions of its dependencies. Use that instead of manually
cloning dependencies to third-party.
half of the materials we used were deprecated with macOS 10.14, broken
and not supported by run time changes of the macOS theme. furthermore
our styling names were completely inconsistent with the actually look
since macOS 10.14, eg ultradark got a lot brighter and couldn't be
considered ultradark anymore.
i decided to drop the old option --macos-title-bar-style and rework
the whole mechanism to allow more freedom. now materials and appearance
can be set separately. even if apple changes the look or semantics in
the future the new options can be easily adapted.
setting the appearance of the window to nil will always use the system
setting and changes on run time switches too. this is only the case for
macOS 10.14.
quite a lot of the title bar functionality and logic was within our
window. since we recently added a custom title bar class to our window
i decided to move all that functionality into that class and in its
own file.
this is also a preparation for the next commits.
i found the old pixel format creation a bit too messy. pixel format
attribute arrays and look ups were all over the place, the actual logic
what kind of format was created was inscrutable, the software pixel
format was hardcoded and no probing was done.
i split the attributes into mandatory and optional ones, one mandatory
for a hardware and software pixel format each, and moved those to the
top of the class. that way new attributes can be easily added to either
the mandatory or optional attributes and they don't mess up the actual
pixel creation logic any more. furthermore both hardware and software
pixel formats are being probed the same way now. to minimise code
duplications the probing was moved into its own function.
only the dragged types NSFilenamesPboardType and NSURLPboardType were
supported to be dropped on the window, which was inconsistent with the
dragged types the dock icon supports. the dock icon additional supports
strings that represents an URL or a path. the system takes care of
validating the strings properly in the case of the dock icon, but in the
case of dropping on the window it needs to be done manually.
support for strings is added by also allowing the NSPasteboardTypeString
type and manually validating the strings. strings are split by new lines
and trimmed, to also support a list of URLs and paths. every new element
is checked if being an URL or path and only then being added to the
playlist.
on init an NSWindow can't be placed on a none Main screen NSScreen
outside the Main screen's frame bounds.
To fix this we just check if the target screen is different from the
main screen and if that is the case we check the actual position with
the expect one. if they are different we try to reposition the window
to the wanted position.
Fixes#6453
when quitting mpv in fullscreen the System always switchs to Space 1
regardless of which Space mpv was on previous to switching to fs. to fix
this we close the window before quitting fs. that way the System
switches back the the Space mpv was previous on before fs.
new events were added but not fetched by the vo, because we didn't
signal the vo that new events were available.
actually wakeup the vo when new events are available.
#6299 reported problems with earlier 3.0.x swift versions. i tested with
3.0.2/SDK 10.12.2 and just assumed it also works with the older 3.0.x
swift and 10.12.x SDK versions. due to the unstable nature of swift
there were slight API differences that caused build problems.
since swift is bundled with the SDK we just bump the minimum swift
version.
Regression from 8e3308d687.
Broken cases were:
* --no-cursor-autohide acted like --cursor-autohide=always.
* --cursor-autohide-fs-only always hid the cursor if starting
non-fullscreen; entering fullscreen at least once fixed it.
The old size limit was chosen before LUT texture was supported in user
shader. At that time, the whole user shader will be compiled and run
on GPU, which makes large user shader impractical to be used.
With the introduction of LUT texture, the old size limit doesn't make
any sense. For example, a 1024x1024 rgba16f LUT will cost 32MB shader
size.
Fix this by increasing the size limit to a value that's unlikely be
reached.
this was meant to be fixed by 546f038, but with --disable-cplugins the
do_the_symbol_stuff function was never called and the handle_add_object
function was again always called before the actual linking task was
created.
to fix this we explicitly call handle_add_object only after all the
tasks the do_the_symbol_stuff function is called after too.
Fixes#6028
Adds the negation missed in 8816e1117e
when moving from a positive-is-active to positive-is-idle variable.
This leads to proper updates to properties such as "eof-reached",
as well as fixes screensaver state updates.
Separately found and fixed by avih and wnoun.
Co-authored-by: wnoun <wnoun@outlook.com>
Manual changes done:
* Merged the interface-changes under the already master'd changes.
* Moved the hwdec-related option changes to video/decode/vd_lavc.c.
modulo operator could be used to check if size is multiple of a
certain number.
equal operator could be used to verify if size of different textures
aligns.
Before this commit, texture offset is set after all source textures
are finalized. Which means CHROMA hooks won't be able to align with
luma planes. This could be problematic for chroma prescalers utilizing
information from luma plane.
Fix this by find the reference texture early, and set global texture
offset early.
Only printable ASCII characters were considered to be valid texts. Make
it possible that UTF-8 contents are also considered valid.
This does not make the SDH subtitle filter support non-English
languages. This just prevents the filter from blindly marking lines that
have only UTF-8 characters as empty.
Fixes#6502
The seek range update was to early and did not take the removed head
packets into account. And therefore missed that the queue was not
BOF anymore.
This led to not be able to backward seek before the first packet of
the first seek range.
Fix it by moving the seek range update after the possible removal and
the change of the BOF flag.
Fixes: #6522
As stated in the original commit message, if the demuxer set the start
time to the first subtitle packet, the subtitles would be shifted
incorrectly. It appears that it is the case for external PGS subtitles.
This reverts commit 520fc74036.
Fixes#5485
This allows context_drm_egl to use as many buffers as libgbm or the
swapchain_depth setting allows (whichever is smaller).
On pause and on still images (cover art etc.) to make sure that output does not
lag behind user input, the swapchain is drained and reverts to working in a dual
buffered (equivalent to swapchain-depth=1) manner.
When possible (swapchain-depth>=2), the wait on the page flip event is now not
done immediately after queueing, but is deferred to the next invocation of
swap_buffers. Which should give us more CPU time between invocations.
Although, since gbm_surface_has_free_buffers() can only tell us a boolean value
and not how many buffers we have left, we are forced to do this contortionist
dance where we first overshoot until gbm_surface_has_free_buffers() reports 0,
followed by immediately waiting so we can free a buffer, to be able to get the
deferred wait on page flip rolling.
With this commit we do not rely on the default vsync fences/latency emulation of
video/out/opengl/context.c, but supply our own, since the places we create and
wait for the fences needs to be somewhat different for best performance.
Minor fixes:
* According to GBM documentation all BO:s gotten with
gbm_surface_lock_front_buffer must be released before gbm_surface_destroy is
called on the surface.
* We let the page flip handler function handle the waiting_for_flip flag.
This solves some edge cases when using files with very weird metadata
(e.g. MaxCLL 10k and so forth). Instead of just blindly seeding it with
the tagged metadata, forcibly set the initial state from the detected
values.
Rather than the linear cd/m^2 units, these (relative) logarithmic units
lend themselves much better to actually detecting scene changes,
especially since the scene averaging was changed to also work
logarithmically.
Gamut mapping can take very bright out-of-gamut colors into the
negatives, which completely destroys the color balance (which tone
mapping tries its best to preserve).
This change switches to a logarithmic mean to estimate the average
signal brightness. This handles dark scenes with isolated highlights
much more faithfully than the linear mean did, since the log of the
signal roughly corresponds to the perceptual brightness.
In theory our "eye adaptation" algorithm works in both ways, both
darkening bright scenes and brightening dark scenes. But I've always
just prevented the latter with a hard clamp, since I wanted to avoid
blowing up dark scenes into looking funny (and full of noise).
But allowing a tiny bit of over-exposure might be a good thing. I won't
change the default just yet (better let users test), but a moderate
value of 1.2 might be better than the current 1.0 limit. Needs testing
especially on dark scenes.
The previous approach of using an FIR with tunable hard threshold for
scene changes had several problems:
- the FIR involved annoying hard-coded buffer sizes, high VRAM usage,
and the FIR sum was prone to numerical overflow which limited the
number of frames we could average over. We also totally redesign the
scene change detection.
- the hard scene change detection was prone to both false positives and
false negatives, each with their own (annoying) issues.
Scrap this entirely and switch to a dual approach of using a simple
single-pole IIR low pass filter to smooth out noise, while using a
softer scene change curve (with tunable low and high thresholds), based
on `smoothstep`. The IIR filter is extremely simple in its
implementation and has an arbitrarily user-tunable cutoff frequency,
while the smoothstep-based scene change curve provides a good, tunable
tradeoff between adaptation speed and stability - without exhibiting
either of the traditional issues associated with the hard cutoff.
Another way to think about the new options is that the "low threshold"
provides a margin of error within which we don't care about small
fluctuations in the scene (which will therefore be smoothed out by the
IIR filter).
Instead of desaturating towards luma, we desaturate towards the
per-channel tone mapped version. This essentially proves a smooth
roll-off towards the "hollywood"-style (non-chromatic) tone mapping
algorithm, which works better for bright content, while continuing to
use the "linear" style (chromatic) tone mapping algorithm for primarily
in-gamut content.
We also split up the desaturation algorithm into strength and exponent,
which allows users to use less aggressive desaturation settings without
affecting the overall curve.
This is the naming xdg-shell stable adopted, it doesn’t make much sense
to keep using “shell” everywhere with all functions calling it
“wm_base”.
Finishes what 76211609e3 started.
Too many broken hardware decoders. Noticed wrong decoding of a video
file encoded with x262 on RX Vega when using VAAPI (Mesa 18.3.2).
Looks fine with swdec and a cheap hardware BD player.
Reverts 017f3d0674
some safety mechanism for the async fs animation aren't needed anymore,
due to possible improved logic and slightly different behaviour on new
macOS versions. that safety fallback prevented the Split View because
it always returned a rectangle of the whole screen, instead of just
part/half of it.
Fixes#6443
Add "auto" the possible values of target-peak. The default value
for target_peak is to calculate the target using mp_trc_nom_peak.
Unfortunately, this default was outside the acceptable range of
10-10000 nits, which prevented its later reassignment. So add an
"auto" choice to target-peak which lets clients and scripts go back
to using the trc default after assigning a value.
this lead to an unexpected videotoolbox-copy hwdec name due to the last
two chars being cut off. since selection is also done by that name one
had to use "videotoolbox-co" to explicitly use the copy mode of
videotoolbox.
Merge file-size/file-format and audio channel-count/format into one line
respectively. This fixes stats overflowing the screen in larger than
19:6 aspect ratios. In this case a problem was reported for ~21:9 which
should be common enough for us to "support" it.
Idle handlers used to not be executed when timers were active
Now they are executed:
* After all expired timers have been executed
* After all events have been processed (same as when there are no timers)
Commit e392d6610d modified the native
demuxer to use track gain as a fallback for album gain if the latter is
not present. This commit makes functionally equivalent changes in the
libavformat demuxer.
If the number of chapters is 0, the chapter list can be NULL. clang
complains that we pass NULL to qsort(). This is yet another pointless UB
that exists for no reason other than wasting your time.
Seems to happen often with ytdl pseudo-DASH streams, so whatever. I
couldn't reproduce it and check what triggers it, I just remember seeing
the error message and found it annoying.
I misunderstood how this extension works. If I understand it correctly
now, it's worse than I thought. They key thing is that the (ust, msc,
sbc) tripple is not for a single swap event. Instead, (ust, msc) run
independently from sbc. Assuming a CFR display/compositor, this means
you can at best know the vsync phase and frequency, but not the exact
time a sbc changed value.
There is GLX_INTEL_swap_event, which might work as expected, but it has
no EGL equivalent (while GLX_OML_sync_control does, in theory).
Redo the context_glx sync code. Now it's either more correct or less
correct. I wanted to add proper skip detection (if a vsync gets skipped
due to rendering taking too long and other problems), but it turned out
to be too complex, so only some unused fields in vo.h are left of it.
The "generic" skip detection has to do.
The vsync_duration field is also unused by vo.c.
Actually this seems to be an improvement. In cases where the flip call
timing is off, but the real driver-level timing apparently still works,
this will not report vsync skips or higher vsync jitter anymore. I could
observe this with screenshots and fullscreen switching. On the other
hand, maybe it just introduces an A/V offset or so.
Why the fuck can't there be a proper API for retrieving these
statistics? I'm not even asking for much.
The redundancy here always annoyed me. Back then I didn't change it
because it's hard to test and I just had fixed something. This doesn't
matter anymore, so simplify it, without testing and with the risk that
something breaks (why care).
--record-file is nice, but only sometimes. If you watch some sort of
livestream which you want to record, it's actually much nicer not to
record what you're currently "seeing", but anything you're receiving.
In theory, this could be easily done with custom I/O. In practice, all
the halfassed garbage in FFmpeg shits itself and fucks up like there's
no tomorrow. There are several problems:
1. FFmpeg pretends you can do custom I/O, but in reality there's a lot
that custom I/O can do. hls.c even contains explicit checks to disable
important things if custom I/O is used! In particular, you can't use the
HTTP keepalive functionality (needed for somewhat decent HLS
performance), because some cranky asshole in the cursed FFmpeg dev.
community blocked it.
2. The implementation of nested I/O callbacks (io_open/io_close) is
bogus and halfassed (like everything in FFmpeg, really). It will call
io_open on some URLs without ever calling io_close. Instead, it'll call
avio_close() on the context directly. From what I can tell, avio_close()
is incompable to custom I/O anyway (overwhelmed by their own garbage,
the fFmpeg devs created the io_close callback for this reason, because
they couldn't fix their own fucking garbage). This commit adds some
shitty workaround for this (technically triggers UB, but with that
garbage heap of a library we depend on it's not like it matters).
3. Even then, you can't proxy I/O contexts (see 1.), but we can just
keep track of the opened nested I/O contexts. The bytes_read is
documented as not public, but reading it is literally the only way to
get what we want.
A more reasonable approach would probably be using curl. It could
transparently handle the keep-alive thing, as well as propagating
cookies etc. (which doesn't work with the FFmpeg approach if you use
custom I/O). Of course even better if there were an independent HLS
implementation anywhere. FFmpeg's HLS support is so embarrassing
pathetic and just goes to show that they belong into the past
(multimedia from 2000-2010) and should either modernize or fuck off.
With FFmpeg's shit-crusted structures, todic communities, and retarded
assholes denying progress, probably the latter. Did I already mention
that FFmpeg is a shit fucked steaming pile of garbage shit?
And all just to get some basic I/O stats, that any proper HLS consumer
requires in order to implement adaptive streaming correctly (i.e.
browser based players, and nothing FFmshit based).
Use the extension to compute the (hopefully correct) video delay and
vsync phase.
This is very fuzzy, because the latency will suddenly be applied after
some frames have already been shown. This means there _will_ be "jumps"
in the time accounting, which can lead to strange effects at start of
playback (such as making initial "dropped" etc. frames worse). The only
reasonable way to fix this would be running a few dummy frame swaps at
start of playback until the latency is known. The same happens when
unpausing.
This only affects display-sync mode.
Correct function was not confirmed. It only "looks right". I don't have
the equipment to make scientifically correct measurements.
A potentially bad thing is that we trust the timestamps we're receiving.
Out of bounds timestamps could wreak havoc. On the other hand, this will
probably cause the higher level code to panic and just disable DS.
As a further caveat, this makes a bunch of assumptions about UST
timestamps. If there are delayed frames (i.e. we skipped one or more
vsyncs), the latency logic is mostly reset. There is no attempt to make
the vo.c skipped vsync logic to use this. Also, the latency computation
determines a vsync duration, and there's no effort to reconcile or share
the vo.c logic for determining vsync duration.
This option has been deprecated upstream for a long time, probably
doesn't even work anymore, and won't work moving forwards as we replace
the vulkan code by libplacebo wrappers.
I haven't removed the option completely yet since in theory we could
still add support for e.g. a native glslang wrapper in the future. But
most likely the future of this code is deletion.
As an aside, fix an issue where the man page didn't mention d3d11.
This commit bumps the libmpv version to 1.102
drm-osd-plane -> drm-draw-plane
drm-video-plane -> drm-drmprime-video-plane
drm-osd-size -> drm-draw-surface-size
"draw plane", as in the plane that OpenGL draws to, whether it be
video + OSD or just OSD.
"drmprime video plane", as in the plane used for hwdec video imported
via drmprime.
"draw surface size", as in the size of the surface used for the draw plane
The new names are invariant whether or not hwdec_drmprime_drm is being
used or not. The original naming was very confusing, as when doing
regular rendering (swdec or vaapi) the video would be displayed on the
"OSD plane", and the "Video plane" would remain unused.
Add general primary/overlay plane option to drm-osd-plane-id and
drm-video-plane-id, so that the user can just request any usable
primary or overlay plane for either of these two options. This should
be somewhat more user-friendly (especially as neither of these two
options currently have a useful help function), as usually you would
only be interested in the type of the plane, and not exactly which
plane gets picked.
By design, some vulkan implementations block until vsync during
vkAcquireNextImageKHR. Since mpv only considers the time that
`swap_buffers` spent blocking as constituting part of the vsync, we can
help it out a bit by pre-emptively calling this function here in order
to improve the accuracy of vsync jitter measurements on vulkan.
(If it fails, we just ignore the error and have the user call it a
second time later - maybe it will work then)
On my system this drops vsync-jitter from ~0.030 to ~0.007, an accuracy
of +/- 100μs. (Which *might* have something to do with the fact that
this is the polling interval for command polling)
Makes performance slightly better when using multiple queues by avoiding
unnecessary semaphores due to bad queue selection.
Also remove an aeons-old workaround for an nvidia bug that only ever
existed in the earliest beta vulkan drivers anyway.
Historically, there's been no way to offer deinterlacing with nvdec,
and for cuviddec, it required a command line flag, with no way to
toggle while playing.
Now that we have a cuda deinterlacing filter in ffmpeg, we can hook
it up hook it up as the cuda auto-deinterlacer. In practice, this
isn't going to be present very often, due to the licensing mess with
the cuda sdk, but we can support it when it is there.
We are currently unnecessarily including vulkan headers even when
not building with vulkan support. I also guarded the GL header
inclusion even though this doesn't appear to break anything today.
Fixes#6330.
This makes the default fit on screen, autofit and window-scale
changing behavior to use the screen working area, instead of
the whole screen area.
As a result mpv window doesn't cover the taskbar now when opening
videos with size larger than the screen size.
The actual behavior now is the same as expected behavior for
usecases 1-4 from #4363.
This commit also removes the screenrc from w32 struct.
The screen rect can now be retrieved via `get_screen_area` function,
which was renamed from `update_screen_rect`.
On a multi-monitor system, if the user moved the window between
monitors, this function will return the current screen area under
the window, and not the screen area from monitor specified by
`--screen` option. The `--screen` option sets the initial monitor
the mpv window is displayed on.
Returning -1 in a function with return type bool is the same as
returning true. In the error paths, false should be returned to
indicate that something went wrong.
depending on the capabilities of the system and testing of various
attributes the resulting CGL pixel format can change. due to that
probing it can be helpful to know which pixel format is used to create
the CGL context. added some verbose logging that outputs final pixel
format.
this adds support for GPU rendered screenshots, DR (theoretically) and
possible other advanced functions in the future that need to be executed
from the rendering thread.
additionally frames that would be off screen or not be displayed when on
screen are being dropped now.
`vkGetPhysicalDeviceProperties2KHR` was added in Vulkan 1.0.39,
but 1.0.61 from Ubuntu Xenial (16.04) was the lowest anyone tried
to build mpv with as of late, so we are marking that as the minimum
required version.
This fixes issues arising during build time from having too old
version of vulkan available on a system, instead of causing a failure
to build.
I was inconsistent about this originally, as the functionality was
moved into the core spec in 1.1 and so both suffixed and unsuffixed
versions of everything exist and can be mixed together.
There's no reason to fail to build with 1.0.39+ so I'm fixing the
names.
On my system, when trying to use mpv with a read-only fd created by
python 3.7, `send` triggers ENOTSOCK, not EBADF. Extend the logic to
detect non-writable fds by this `errno`.
The buffer is written to in `audio_config_to_str_buf` using `snprintf`
with a `%s` formatting of a 128-byte buffer. This can overflow the
target buffer of 80 causing a truncated output.
The `strncpy` function will not always place a terminating `NUL` if the
source is longer than the destination buffer. Instead, let `strncpy`
truncate as it normally would, but leave the last byte alone (it is
zero-initialized when declared).
Currently, the error paths in init() are a bit confusing, and we can
end up trying to pop the current context when there is no context,
which leads to distracting error messages.
I also added an explicit path to return early if the GPU backend is
not OpenGL or Vulkan. It's pointless to do any other cuda init
after that point. (Of course, someone could write more interops.)
Fixes#6256
Apparently, for bit streaming DTS-HD MA is specified to be handled as an
eight channel (7.1) bit stream, while DTS-HD HRA is specified to be
handled as a stereo bit stream.
Define a variable for this, and utilize it to set the correct values
for both the DTS-HD bit streaming rate, as well as the channel count
for the SPDIF encoder.
Fixes#6148
when passed as a string check_cc tries to split that string, since it
assumes that several include paths can be passed to it. instead we just
use a list to make it unambiguous.
Since 810acf32d6 video_plane can be NULL
under some circumstances. While there is a check in init, init treats
this as an error condition and would call uninit, which in turn calls
disable_video_plane, which would then segfault. Fix this by including
a NULL check inside disable_video_plane, so that it doesn't try to
disable what isnt' there.
Despite their place in the tree, hwdecs can be loaded and used just
fine by the vulkan GPU backend.
In this change we add Vulkan interop support to the cuda/nvdec hwdec.
The overall process is mostly straight forward, so the main observation
here is that I had to implement it using an intermediate Vulkan buffer
because the direct VkImage usage is blocked by a bug in the nvidia
driver. When that gets fixed, I will revist this.
Nevertheless, the intermediate buffer copy is very cheap as it's all
device memory from start to finish. Overall CPU utilisiation is pretty
much the same as with the OpenGL GPU backend.
Note that we cannot use a single intermediate buffer - rather there
is a pool of them. This is done because the cuda memcpys are not
explicitly synchronised with the texture uploads.
In the basic case, this doesn't matter because the hwdec is not
asked to map and copy the next frame until after the previous one
is rendered. In the interpolation case, we need extra future frames
available immediately, so we'll be asked to map/copy those frames
and vulkan will be asked to render them. So far, harmless right? No.
All the vulkan rendering, including the upload steps, are batched
together and end up running very asynchronously from the CUDA copies.
The end result is that all the copies happen one after another, and
only then do the uploads happen, which means all textures are uploaded
the same, final, frame data. Whoops. Unsurprisingly this results in
the jerky motion because every 3/4 frames are identical.
The buffer pool ensures that we do not overwrite a buffer that is
still waiting to be uploaded. The ra_buf_pool implementation
automatically checks if existing buffers are available for use and
only creates a new one if it really has to. It's hard to say for sure
what the maximum number of buffers might be but we believe it won't
be so large as to make this strategy unusable. The highest I've seen
is 12 when using interpolation with tscale=bicubic.
A future optimisation here is to synchronise the CUDA copies with
respect to the vulkan uploads. This can be done with shared semaphores
that would ensure the copy of the second frames only happens after the
upload of the first frame, and so on. This isn't trivial to implement
as I'd have to first adjust the hwdec code to use asynchronous cuda;
without that, there's no way to use the semaphore for synchronisation.
This should result in fewer intermediate buffers being required.
This is arguably a little contrived, but in the case of CUDA interop,
we have to track additional state on the cuda side for each exported
buffer. If we want to be able to manage buffers with an ra_buf_pool,
we need some way to keep that CUDA state associated with each created
buffer. The easiest way to do that is to attach it directly to the
buffers.
The CUDA/Vulkan interop works on the basis of memory being exported
from Vulkan and then imported by CUDA. To enable this, we add a way
to declare a buffer as being intended for export, and then add a
function to do the export.
For now, we support the fd and Handle based exports on Linux and
Windows respectively. There are others, which we can support when
a need arises.
Also note that this is just for exporting buffers, rather than
textures (VkImages). Image import on the CUDA side is supposed to
work, but it is currently buggy and waiting for a new driver release.
Finally, at least with my nvidia hardware and drivers, everything
seems to work even if we don't initialise the buffer with the right
exportability options. Nevertheless I'm enforcing it so that we're
following the spec.
Since the code just broke out of the loop on a match rather than jumping
straight to the end of the function body, it ended up hitting the code
path for when the end of the list was reached.
since we draw our own title bar we lose the standard functionality of
the system provided title bar. because of that we have to reimplement
the functionality of double clicking the title bar. depending on the
system preferences we want to minimize, zoom or do nothing.
Fixes#6223
On a multi monitor setup, when the center of the window was going off
screen, the icc profile would always switch to the profile of the first
screen.
This fixes the issue by defaulting the value to the current screen.
This was pased on the texture height, which was a mistake. In some cases
it could exceed the actual size of the buffer, leading to a vulkan API
error. This didn't seem to cause any problems in practice, since a
too-large synchronization is just bad for performance and shouldn't do
any harm internally, but either way, it was still undefined behavior to
submit a barrier outside of the buffer size.
Fix the calculation, thus fixing this issue.
Since linear downscaling makes sense to handle independently from
linear/sigmoid upscaling, we split this option up. Now,
linear-downscaling is its own option that only controls linearization
when downscaling and nothing more. Likewise, linear-upscaling /
sigmoid-upscaling are two mutually exclusive options (the latter
overriding the former) that apply only to upscaling and no longer
implicitly enable linear light downscaling as well.
The old behavior was very confusing, as evidenced by issues such
as #6213. The current behavior should make much more sense, and only
minimally breaks backwards compatibility (since using linear-scaling
directly was very uncommon - most users got this for free as part of
gpu-hq and relied only on that).
Closes#6213.
Someone on IRC pointed out that the default stats bindings weren't
documented in the interactive control section of the manual, so
let's add them with a short mention and a reference to the STATS
section of the manual.
when entering a Split View a windowDidEnterFullScreen event happens
without a previous toggleFullScreen call. in that case it tries to stop
an animation that was never initiated by us and basically breaks the
system initiated fullscreen, or in this case the Split View. immediately
after entering the fullscreen it tries top stop the animation and
resizes the window, which causes the window to exit fullscreen. only
try to stop an animation that was initiated by us and is safe to stop.
without assistive-device permissions the event tap can't be create on
10.14 any more which lead to an assertion.
System Preferences > Security & Privacy > Privacy > Accessibility and
add mpv or your terminal App to the list.
Since this function is called with packets on the stack, trying to free
them makes no sense. Instead, it should unref (which is what
`av_interleaved_write_frame` does anyway, rather than freeing).
Also, the calling code tried unreffing the packet a second time, even
after it was "freed" by the callee in the failure case - and after
ownership was taken over by `av_interleaved_write_frame` in the
successful case. Both of these cases were wrong.
I encountered a stream that fails with "Could not demux init fragment.".
It turns out this is a regression from the recent change to that code.
The assumption was that demux_lavf.c would treat this as concatenated
stream - which it does, but not for probing.
Doing this transparently is hard without doing it properly. Doing it
properly would mean creating some sort of stream_concat (reminiscent of
that FFmpeg security bug). I probably don't want to go there, and I
think libavformat should just support this directly, so whatever.
Hack-fix this with the knowledge that the init segment will always
contain the headers.
This happened with a .flac file inside an archive. It tried to seek
beyond the end of the archive entry in a format where seeking isn't
supported. stream_libarchive handles these situations by skipping data.
But when the end of the archive is reached, archive_read_data() returns
0. While libarchive didn't bother to fucking document this, they do say
it's supposed to work like read(), so I guess a return value of 0 really
means EOF. So change the "< 0" to "<= 0". Also add some error logging.
The same file actually worked without out of bounds reads when
extracted, so there still might be something very wrong.
FFmpeg is retarded enough not to give us any indication whether it is
(unless we query fields not in the ABI/API). I bet FFmpeg developers
love it when library users have to litter their code with duplicated
information.
by default the pixel format creation falls back to software renderer
when everything fails. this is mostly needed for VMs. additionally one
can directly request an sw renderer or exclude it entirely.
moved the retrieval of the macOS specific options from the backend
initialisation to the initialisation of the CocoaCB class, the earliest
point possible. this way macOS specific options can be used for the
opengl context creation for example.
This is to improve the experience when running with default settings
on a driver that doesn't have any overlay planes (or indeed only one
plane), but still supports DRM atomic. Since the drmprime video plane
is set to pick an overlay plane by default it would fail on these
drivers due to not being able to create any atomic context. Users with
such cards had to specify --drm-video-plane-id manually to some bogus
value (it's not used after all).
The "video" plane is only ever used by the drmprime-drm hwdec interop,
which is not used at all in the typical usecase where everything is
actually rendered on to the "OSD" plane using EGL, so having an atomic
context without the "video" plane should be fine most of the time.
For vec3, the alignment and size differ. The current code will pack a
struct like { vec3; float; vec2 } into 8 machine words, whereas the spec
would only use 6.
This actually fixes a real bug: The only place in the code I could find
where it was conceivably possible that a vec3 is followed by a float was
when using --gpu-dumb-mode in combination with --gamma-factor, and only
when --gpu-api=vulkan. So it's no surprised nobody ran into it yet.
These used to be unsupported long ago, but it seems glslang added
support in the meantime. (I don't know which version, but I'm guessing
it was long enough ago that we don't have to add a feature check)
Should hopefully help make push constant layouts more robust against
possible bugs either in our code or in the driver.
According to ALSA doxy, EPIPE is a synonym to SND_PCM_STATE_XRUN,
and that is a state that we should attempt to automatically recover
from. In case recovery fails, log an error and return zero.
A warning message will still be output for each XRUN since those
are not something we should generally be receiving.
This has been way too long coming, and for me to notice that a
whole lot of ao_alsa functions do an early return if the AO is
paused.
For the STATE_SETUP case, I had this reproduced once, and never
since. Still, seems like we can start calling this function before
the ALSA device has been fully initialized so we might as well
early exit in that case.
Certain low-end Mali GPUs have a rather low precision and overflow
during the PRNG calculations, thereby breaking e.g. deband-grain.
Modify the permute() to avoid this, this does not impact the
quality of PRNG output (noticeably).
This problem was observed on:
GL_VENDOR='ARM', GL_RENDERER='Mali-T720'
GL_VERSION='OpenGL ES 3.1 v1.r15p0-00rel0.bdd9e62cdc8c88e0610a16b5901161e9'
mpctx->current_track[0][STREAM_VIDEO] (and STREAM_AUDIO) are empty when
using --lavfi-complex. Moving the muxer stream hinting after audio/video chain
initialization and checking if the chains exist fixes encoding with --lavfi-complex.
Previously, the output audio/video streams did not get prepared and the encode
would fail due to unexpected stream addition.
Python 2 may not be present in the CI images in the future,
but waf’s shebang line still uses its executable name.
Explicitly call the right major version of the interpreter.
Upstream has this now. Didn't really make any different for me (except
making the polar compute shader 2%-3% faster), but maybe it does for
somebody else.
The demuxer cache is the only cache now. Might need another change to
combat seeking failures in mp4 etc. The only bad thing is the loss of
cache-speed, which was sort of nice to have.
When using multiple compute shaders as part of the same pass, there can
be a conflict in the block sizes. In the problematic case, the HDR
detection shader can collide with the polar sampling shader. In this
case, the solution is clear - the passes that can handle any size should
"give in" and not overwrite the block sizes.
Fixes#6083.
FFmpeg is retarded enough not to give us any indication whether it is
(unless we query fields not in the ABI/API). I bet FFmpeg developers
love it when library users have to litter their code with duplicated
information.
With this option, the script will check that the expected waf version is
present, but will simply fail if it's not, rather than trying to
download it. This allows a package build script to avoid compile-time
network access but still ensure it's using the right waf version.
Quitting is slightly asynchronous, so the status line can be updated
again. Normally, that's fine, but if quitting comes with a message (such
as with quit_watch_later), it will print the status line again after the
message, which looks annoying. So flush and clear the status message if
it's updated during quitting.
instead of force unwrapping and chaining the optional vars in our
containsMouseLocation function, safely unwrap and guard the resulting
var.
Fixes#6062
duration is parsed as an integer, and the default value is used if ```-1``` is passed. Passing ```-``` as described here causes a parameter value error.
ao->device_buffer will only affect the enqueue size if the latter
is not specified. In other word, its intended purpose will solely
be setting/guarding the soft buffer size.
This guarantees that the soft buffer size will be consistent no
matter a specific enqueue size is set or not. (In the past it
would drop to the default of the generic audio-buffer option.)
opensles-frames-per-buffer has been renamed to opensles-frames-per
-enqueue, as it was never purposed to set the soft buffer size. It
will only make sure the size is never smaller than itself, just as
before.
opensles-buffer-size-in-ms is introduced to allow easy tuning of
the relative (i.e. in time) soft buffer size (and enqueue size,
unless the aforementioned option is set). As "device buffer" never
really made sense in this AO, this option OVERRIDES audio-buffer
whenever its value (including the default) is larger than 0.
Setting opensl-buffer-size-in-ms to 1 allows you to equate the soft
buffer size to the absolute enqueue size set with opensl-frames-per
-enqueue conveniently (unless it is less than 1ms).
When both are set to 0, audio-buffer will be the ultimate fallback.
If audio-buffer is also 0, the AO errors out.
Functions `write` and `smbc_write` are given a diminishing buffer of
incorrect constant size. After partial writes, the code would do another
write of the full original length, failing to subtract the amount
already written.
mpv.com fails to build when cross-compiling with a multilib version
of GCC, because the -m32 flags aren't getting passed to the build
process for osdep/win32-console-wrapper.c or the link phase for
mpv.com itself.
This started breaking with newer (2.0.x) waf versions, and it was
noticed that a built-in waf module was providing very similar
functionality.
APPNAME definition was required to have `gnu_dirs`' PACKAGE
variable to be defined in order to have f.ex. documentation
installed to the correct directory.
Currently unused options added by `gnu_dirs` were removed to clean
up the output of the help command.
Effective changes to behavior:
- `gnu_dirs` will attempt to figure out if it needs to use lib64
instead of lib within your installation prefix. If you would
like it to not do that, set `--libdir` during configuration.
The way it tries to figure lib/lib64 out is if there's a
`/usr/lib64` and no `/usr/lib32`.
- `--incdir` is now `--includedir` as per standard `gnu_dirs`
behavior.
Before, `do_the_symbol_stuff` would implicitly come before
`handle_add_object`, which adds object files to the linking task.
With newer (2.0.x) versions of waf, the ordering seems to get more
optimized, and thus we have to declare that the function that creates
the linking task should come before the task that adds object files
to the task.
Add another parameter to mpv_opengl_drm_params to hold the FD to the
render node, so that the fd can be passed to hwdec_vaegl.
The render node is opened in context_drm_egl and inferred from the
primary device fd using drmGetRenderDeviceNameFromFd.
The previous code did not save enough information about the old state,
and could end up changing what plane the fbcon:s FB got attached to,
or in worse case causing a blank screen (observed in some multi-screen
setups on Sandy Bridge).
In addition refactor the handling of drmModeModeInfo property blobs to
not leak, as well as enable reuse of already created blobs.
this was caused by commit 2e7a4f7. the LAST_LINKFLAGS were not added to
the linking of libmpv and that caused a linking error. manually add the
link flags the same way it's done when linking mpv.
Fixes#5968
Temporary solution. For now, this builds using a container image
based on openSUSE Tumbleweed with the current FFmpeg release.
More containers will be added (at least with git snapshots of FFmpeg
and libass), and Travis will eventually be replaced with something
we have more control over.
init is a reserved keyword and Swift 4.2 got a bit stricter about using
it. this could be fixed by adding apostrophes around init but makes the
code uglier. hence i just renamed init to initialized and for
consistency uninit to uninitialized.
Fixes#5899
this reverts commit a174566 since the actually reason for failing has
been found. the isysroot flag overwrites the framework and library
search paths. though we only need to overwrite the former and there is
no way to just overwrite that one. we manually add the standard library
search paths to the very end of the linking command, so it won't
interfere with the search paths extracted by waf.
Fixes#5791
the pre-allocation was needed because the layer allocated a opengl
context async itself and we couldn't influence that. so we had to start
the core after the context was actually allocated. furthermore a window,
view and layer hierarchy had to be created so the layer would create
a context.
now, instead of relying on the layer to create a context we do this
manually and re-use that context later when the layer wants to create
one async itself.
The currently-displayed tags make sense for music files, but similar
information for video is more commonly - or at least should be - put
under other tags, while the audio-related tags are often used for
other information on video files (particularly with youtube-dl's
output).
When seeking near the end of the file and the next file loads, seeking
continues on the next file at the same position and then immediately
the file after that. This patch stops slider seeking when a new file is
loaded, which is the standard behavior of many other players.
Fixes a bug with alsa dmix on Fedora 29. After several minutes,
audio suddenly becomes bad and muted.
Actually, I don't know what causes this. Probably this is a bug in alsa.
In any case, as snd_pcm_status() returns not only 'avail', but also other
fields such as tstamp, htstamp, etc, this could be considered a good
simplification, as only avail is required for this function.
This sacrifices some dynamic range for well-behaved sources, but
prevents catastrophic desaturation on badly mastered / too bright
sources. I think that's the better trade-off. This makes the
desaturation algorithm much "safer" to deploy by default, as well. One
could even argue going up to strength 1.0, which works better for some
sources but worse for others. But I think the current strength is the
best trade-off even after this change.
For some reason shadow_offset is a int16_t variable (to save some space
or something), which means the static part of the entire option list
must be below 32KB. This is fine, but still add a check against
overflows. (Currently it's 3.6KB. This does not include dynamic
allocations like strings.)
The player fully restarts playback when the edition or disk title is
changed. Before this, the player tried to reinitialized playback
partially. For example, it did not print a new "Playing: <file>"
message, and did not send playback end to libmpv users (scripts or
applications).
This playback restart code was a bit messy and could have unforeseen
interactions with various state. There have been bugs before. Since it's
a mostly cosmetic thing for an obscure feature, just change it to a full
restart. This works well, though since it may have consequences for
scripts or client API users, mention it in interface-changes.rst.
This flag is used only by the command parser. Its value overlapped with
some of the existing m_option flags, but only flags that did not matter
for the command parser (i.e. the flag bits used had mostly private uses
in each component). It's still a bit unclean and dangerous to use an
essentially random value, so reuse M_OPT_OPTIONAL_PARAM for it.
Since M_OPT_OPTIONAL_PARAM has a slightly longer name than
MP_CMD_OPT_ARG, I'm going to keep the old name.
When changing video filters during initialization, there was a small
time window where video was initialized, but playback restart was not
complete yet. In this time window, playback_pts is not set. But since
issue_refresh_seek() was using this, it could lead to no refresh being
done _if_ the "video" had only 1 frame (such as cover art).
Fix this by using get_current_time() instead, which is the current time
with corner cases such as ongoing loading or seeks taken into account.
See also the previous commit. Without that, get_current_time() could
return NOPTS during init.
Fixes#5831.
There is some code that checks a FD for whether it is a regular file or
not. If it's not a regular file, it e.g. enables use of poll() to avoid
blocking forever.
But this was done only for FDs that were open()ed by us, not from stdin
special handling or fd://. Consequently, " | mpv -" could block the
player. Fix this by moving the code and running for it on all FDs.
Also, set p->regular_file even on mingw.
When the current packet queue was completely empty, and EOF was reached,
the queue->is_eof flag was not correctly set to true. Change this by
reading ds->eof to check whether the stream is considered EOF. We also
need to make sure update_seek_ranges() is called in this case, so change
the code to simply call it when queue->is_eof changes.
Also, read_packet() needs to call adjust_seek_range_on_packet() if
ds->eof changes. In that case, the decoder also needs to be notified
about EOF. So both of these should be called when ds->eof changes to
true. (Other code outside of this function deals with the case when
ds->eof is changed to false.)
In addition, this code was kind of shoddy about calling wakeup_ds()
correctly. It looks like there was an inverted condition, and sent a
wakeup to the decoder only when ds->eof was already true, which is
obviously bogus. The final EOF case tried to be somehow clever about
checking in->last_eof for notifying the codec, which is sort of OK, but
seems to be strictly worse than just checking whether ds->eof changed.
Fix these things.
The only effective difference is that the former explicitly checks
whether the JSON value type is string, and errors out if not. The rest
is exactly the same (mpv_set_property_string is mpv_set_property with
MPV_FORMAT_STRING).
It seems silly to keep this, so just remove it.
The default get_format does exactly do this, so we don't need to
duplicate it.
The only potential problem with this is that the logic doesn't entirely
prevent that the avcodec_default_get_format hw_device_ctx path is
triggered, which would probably work, but has unknown consequences and
interactions. But the way the logic currently works it can't happen,
provided the hwaccel metadata libavcodec provides is correct.
Avoids 100% CPU usage due to terminal code retrying read(). Seems like
this was "forgotten" (or there was somehow the assumption poll() would
not signal POLLIN anymore).
Fixes#5842.
For some reason, the X default modifier map binds shift+tab to
ISO_Left_Tab instead of the regular Tab. So to get Shift+TAB recognized
by mpv, we also need to accept ISO_Left_Tab.
This patch matches what other programs like e.g. Qt do, which treat Tab
and ISO_Left_Tab as the same thing.
God only knows why the distinction exists, and why X decides to mix up
its bindings like that.
Fixes#5849
If anyone happened to build with GL disabled, this could lead to option
changes not always refreshing the screen. Since vo_gpu is always enabled
now (just not necessarily any backend for it), we can drop the #if
completely.
(The way this works is a bit idiotic - the option cache exists only to
grab the change notification, which will trigger a redraw and make
vo_gpu update its own second copy of them. But at least it avoids some
layering issues for now.)
Avoids 100% CPU usage due to terminal code retrying read(). Seems like
this was "forgotten" (or there was somehow the assumption poll() would
not signal POLLIN anymore).
Fixes#5842.
This is working towards a change intended in the future: nothing should
write to the option struct directly, but use functions that raise proper
notifications. Until this is complete it will take a while, and this
commit does not change all cases of direct access, just some simple
ones.
In all of these 3 changes, the actual write access is done by the
generic property-option bridge.
C11 can access atomic variables normally (in which case they use the
strictest memory access semantics). But the mpv stdatomic wrapper for
C99 compilers does not allow it, because it couldn't give any
guarantees. This means we always need to access them with atomic macros.
While we're at, use relaxed semantics for the m_config_cache field,
since because it's accessed from a single thread only (essentially
used in a non-atomic way). Switch the comparison arguments to make the
formatting look slightly less weird.
Although the new code actually fires update notifications only when
needed, m_config_cache_update() itself returned a rather coarse change
value, which could indicate change even if none of the cached options
were changed. On top of that, some code (like vo_gpu) calls the update
function on every frame, which would reconfigure the renderer even on
unrelated option changes.
This was always a legacy thing. Remove it by applying an orgy of
mp_get_config_group() calls, and sometimes m_config_cache_alloc() or
mp_read_option_raw().
win32 changes untested.
The --hwdec* options are a good fit for the vd_lavc local option
struct. This annoyingly requires manual prefixing of most of these
options with --vd-lavc (could be avoided by using more sub-struct
craziness, but let's not).
The path functions need to access the option that forces non-default
config directories. Just add it as a field to mpv_global - it seems
justified. The accessed options were always enforced as immutable after
init, so there's not much of a change.
Passing NULL to mp_get_config_group() returns the main option struct.
This is just a dumb hack to deal with inconsistencies caused by legacy
things (as I'll claim), and will probably be changed in the future. So
before littering the whole code base with hard to find NULL parameters,
require using callers an easy to find separate define.
Options with dynamic memory allocations (such as strings) require some
care. They need to be fully copied on initialization, and if a static
default value was declared, we must not free that value either.
Instead of going through the entire thing even for simple types like
integers, really run it only for options with dynamic allocations. To
distinguish types which use dynamic allocations, we can use the fact
that they require a free callback (otherwise they would leak). As a
result initialization of simple types becomes chaper, and the init
function does nothing at all if src==dst for a simple type.
(It's funny how mplayer had M_OPT_TYPE_DYNAMIC since 2002, until we
replaced it by the same heuristic as used here in commit 3bb134969e.
It's also funny how the new check was used only for some asserts, and
finally removed in commit 7539928c1c. I guess at this time I felt like
having uniform code was more important than pointless
micro-optimizations.)
The src==NULL case is removed because it can't happen.
C99 still works, but in theory we're using C11 features already, such as
stdatomic.h. gcc/clang let us use it in C99 mode too, but using C11 is
at least more proper.
Actually rewrite most of the option management code. This affects how
options are allocated, and how thread-safe access to them is done.
One thing that is nicer is that creating m_config_cache does not need to
ridiculously recreate and store the entire option list again. Instead,
option metadata and option storage are now separated. m_config contains
the metadata, and m_config_data all or parts of the actual option
values. (m_config_cache simply uses the metadata part of m_config, which
is immutable after creation.)
The mentioned hack was introduced in commit 1a2319f3e4, and is the
global state around g_group_mutex. Although it was "benign" global
state, it's good that it's finally removed.
For some reason shadow_offset is a int16_t variable (to save some space
or something), which means the static part of the entire option list
must be below 32KB. This is fine, but still add a check against
overflows. (Currently it's 3.6KB. This does not include dynamic
allocations like strings.)
Until now, stopping playback aborted the demuxer and I/O layer violently
by signaling mp_cancel (bound to libavformat's AVIOInterruptCB
mechanism). Change it to try closing them gracefully.
The main purpose is to silence those libavformat errors that happen when
you request termination. Most of libavformat barely cares about the
termination mechanism (AVIOInterruptCB), and essentially it's like the
network connection is abruptly severed, or file I/O suddenly returns I/O
errors. There were issues with dumb TLS warnings, parsers complaining
about incomplete data, and some special protocols that require server
communication to gracefully disconnect.
We still want to abort it forcefully if it refuses to terminate on its
own, so a timeout is required. Users can set the timeout to 0, which
should give them the old behavior.
This also removes the old mechanism that treats certain commands (like
"quit") specially, and tries to terminate the demuxers even if the core
is currently frozen. This is for situations where the core synchronized
to the demuxer or stream layer while network is unresponsive. This in
turn can only happen due to the "program" or "cache-size" properties in
the current code (see one of the previous commits). Also, the old
mechanism doesn't fit particularly well with the new one. We wouldn't
want to abort playback immediately on a "quit" command - the new code is
all about giving it a chance to end it gracefully. We'd need some sort
of watchdog thread or something equally complicated to handle this. So
just remove it.
The change in osd.c is to prevent that it clears the status line while
waiting for termination. The normal status line code doesn't output
anything useful at this point, and the code path taken clears it, both
of which is an annoying behavior change, so just let it show the old
one.
Before this, mpctx->playing was often used to determine whether certain
new state could be added to the playback state. In particular this
affected external files (which added tracks and demuxers). The variable
was checked to prevent that they were added before the corresponding
uninit code. We want to make a small part of uninit asynchronous, but
mpctx->playing needs to stay in the place where it is. It can't be used
for this purpose anymore.
Use mpctx->stop_play instead. Make it never have the value 0 outside of
loading/playback. On unloading, it obviously has to be non-0.
Change some other code in playloop.c to use this, because it seems
slightly more correct. But mostly this is preparation for the following
commit.
This will enable the player core to terminate the demuxers in a "nicer"
way without having to block on network. If it just used demux_free(), it
would either have to block on network, or like currently, essentially
kill all I/O forcefully.
The API is slightly awkward, because demuxer lifetime is bound to its
allocation. On the other hand, changing that would also be awkward, and
introduce weird in-between states that would have to be handled in tons
of places.
Currently unused, to be user later.
Alway give each demuxer its own mp_cancel instance. This makes
management of the mp_cancel things much easier. Also, instead of having
add/remove functions for mp_cancel slaves, replace them with a simpler
to use set_parent function. Remove cancel_and_free_demuxer(), which had
mpctx as parameter only to check an assumption. With this commit,
demuxers have their own mp_cancel, so add demux_cancel_and_free() which
makes use of it.
Them being separate is just dumb. Replace them with a single
demux_free() function, and free its stream by default. Not freeing the
stream is only needed in 1 special case (demux_disc.c), use a special
flag to not free the stream in this case.
The player fully restarts playback when the edition or disk title is
changed. Before this, the player tried to reinitialized playback
partially. For example, it did not print a new "Playing: <file>"
message, and did not send playback end to libmpv users (scripts or
applications).
This playback restart code was a bit messy and could have unforeseen
interactions with various state. There have been bugs before. Since it's
a mostly cosmetic thing for an obscure feature, just change it to a full
restart. This works well, though since it may have consequences for
scripts or client API users, mention it in interface-changes.rst.
The properties/commands touched in this commit are all for obscure
special inputs (BD/DVD/DVB/TV), and they all block on the demuxer/stream
layer. For network streams, this blocking is very unwelcome. They will
affect playback and probably introduce pauses and frame drops. The
player can even freeze fully, and the logic that tries to make playback
abortable even if frozen complicates the player.
Since the mentioned accesses are not needed for network streams, but
they will block on network streams even though they're going to fail,
add a flag that coarsely enables/disables these accesses. Essentially it
establishes a whitelist of demuxers/streams which support them.
In theory you could to access BD/DVD images over network (or add such
support, I don't think it's a thing in mpv). In these cases these
controls still can block and could even "freeze" the player completely.
Writing to the "program" and "cache-size" properties still can block
even for network streams. Just don't use them if you don't want freezes.
Don't allow it to freeze everything when loading a playlist from network
(although you definitely shouldn't do that, but whatever).
This also affects the really obscure --ordered-chapters-files option.
The --playlist option on the other hand has no choice but to freeze the
shit, because there's no concept of aborting the player during command
line parsing.
Until now, they could be aborted only by ending playback, and calling
mpv_abort_async_command didn't do anything.
This requires furthering the mess how playback abort is done. The main
reason why mp_cancel exists at all is to avoid that a "frozen" demuxer
(blocked on network I/O or whatever) cannot freeze the core. The core
should always get its way. Previously, there was a single mp_cancel
handle, that could be signaled, and all demuxers would unfreeze. With
external files, we might want to abort loading of a certain external
file, which automatically means they need a separate mp_cancel. So give
every demuxer its own mp_cancel, and "slave" it to whatever parent
mp_cancel handles aborting.
Since the mpv demuxer API conflates creating the demuxer and reading the
file headers, mp_cancel strictly need to be created before the demuxer
is created (or we couldn't abort loading). Although we give every
demuxer its own mp_cancel (as "enforced" by cancel_and_free_demuxer),
it's still rather messy to create/destroy it along with the demuxer.
The intention is to avoid that the parent mp_cancel retains the
internally allocated wakeup pipe. File FDs are a relatively scarce
resource, so try to avoid having too many. This might matter for
subtitle files, for which it is relatively likely that they are loaded
in large quantities.
demux_lavf.c will close the underlying stream for most subtitle files,
and now it will free the wakeup pipe too. Actually, there are currently
only 1 or 2 mp_cancel objects per mpv core, but this could change if
every external subtitle track gets its own mp_cancel in later commits.
The OS specifics are merged because the resulting ifdeffery is not much
worse than the old ifdeffery, but the logic that is now shared is
becoming more complex.
Create all objects lazily. The intention is to make mp_cancel instances
cheaper. POSIX pipes and win32 Events are pretty heavy weight, and are
only needed in special situations.
Add a mechanism to "chain" mp_cancel instances. Needed by the later
commits for whatever reasons.
Untested on win32.
This provides macros for managing intrusive doubly linked lists.
There are many ways how to do those in a "generic" way in C. For example
Solaris style lists are pretty nice:
https://github.com/illumos/illumos-gate/blob/master/usr/src/uts/common/sys/list.hhttps://github.com/illumos/illumos-gate/blob/master/usr/src/common/list/list.c
I even have an independent implementation of this, which could be ISC
licensed. But I think it's easier to vomit ~100 lines of preprocessor
garbage, which has a lower footprint, and I think it wins slightly on
the side of type safety, simplicity, and ease of use, even if it doesn't
look as magically nice.
It seems a bit inappropriate to have dumped this into stream.c, even if
it's roughly speaking its main user. At least it made its way somewhat
unfortunately to other components not related to the stream or demuxer
layer at all.
I'm too greedy to give this weird helper its own file, so dump it into
thread_tools.c.
Probably a somewhat pointless change.
This flag is used only by the command parser. Its value overlapped with
some of the existing m_option flags, but only flags that did not matter
for the command parser (i.e. the flag bits used had mostly private uses
in each component). It's still a bit unclean and dangerous to use an
essentially random value, so reuse M_OPT_OPTIONAL_PARAM for it.
Since M_OPT_OPTIONAL_PARAM has a slightly longer name than
MP_CMD_OPT_ARG, I'm going to keep the old name.
Before this change, only 1 command or so had named arguments. There is
no reason why other commands can't have them, except that it's a bit of
work to add them.
Commands with variable number of arguments are inherently incompatible
to named arguments, such as the "run" command. They still have dummy
names, but obviously you can't assign multiple values to a single named
argument (unless the argument has an array type, which would be
something different). For now, disallow using named argument APIs with
these commands. This might change later.
2 commands are adjusted to not need a separate default value by changing
flag constants. (The numeric values are C only and can't be set by
users.)
Make the command syntax in the manpage more consistent. Now none of the
allowed choice/flag names are in the command header, and all arguments
are shown with their proper name and quoted with <...>.
Some places in the manpage and the client.h doxygen are updated to
reflect that most commands support named arguments. In addition, try to
improve the documentation of the syntax and need for escaping etc. as
well.
(Or actually most uses of the word "argument" should be "parameter".)
When changing video filters during initialization, there was a small
time window where video was initialized, but playback restart was not
complete yet. In this time window, playback_pts is not set. But since
issue_refresh_seek() was using this, it could lead to no refresh being
done _if_ the "video" had only 1 frame (such as cover art).
Fix this by using get_current_time() instead, which is the current time
with corner cases such as ongoing loading or seeks taken into account.
See also the previous commit. Without that, get_current_time() could
return NOPTS during init.
Fixes#5831.
This is nonsense. Didn't matter in most situations, because seeking
itself set this after it was cleared. But some callers don't do this,
see e.g. commit ed73ba8964. There is no need to clear it at all, and
it causes issues with the next commit. It only needs to be reset on
loading.
Also move the initialization on loading up, which doesn't change
behavior, but makes the intention clearer.
There is some code that checks a FD for whether it is a regular file or
not. If it's not a regular file, it e.g. enables use of poll() to avoid
blocking forever.
But this was done only for FDs that were open()ed by us, not from stdin
special handling or fd://. Consequently, " | mpv -" could block the
player. Fix this by moving the code and running for it on all FDs.
Also, set p->regular_file even on mingw.
The default get_format does exactly do this, so we don't need to
duplicate it.
The only potential problem with this is that the logic doesn't entirely
prevent that the avcodec_default_get_format hw_device_ctx path is
triggered, which would probably work, but has unknown consequences and
interactions. But the way the logic currently works it can't happen,
provided the hwaccel metadata libavcodec provides is correct.
If a stream starts later than the others at the start of the file, it
shouldn't restrict the seek range to the time stamp where it begins.
This is similar to the previous commit, just for the other end.
Normally, the seek range is the minimum overlap of the cached ranges of
each stream. But if one of the streams ends earlier, this leads to the
seek range getting cut off, even if you could seek there.
Change it so that EOF streams cannot restrict the end of the seek range.
They can only extend it. This is the opposite from not-EOF streams, so
they need to be handled separately. In particular, they get exluded from
normal end range calculation, but when full EOF is reached, all streams
are EOF, and the maximum end time can be used to set the seek end time.
(In theory we could also take the max with the demuxer signaled total
file duration, but let's not for now.)
Also, if a stream is completely empty, essentially skip it, instead of
considering the range unseekable. (Also, we don't need to mess with
seek_start in this case, because it will be NOPTS and is skipped
anyway.)
When the current packet queue was completely empty, and EOF was reached,
the queue->is_eof flag was not correctly set to true. Change this by
reading ds->eof to check whether the stream is considered EOF. We also
need to make sure update_seek_ranges() is called in this case, so change
the code to simply call it when queue->is_eof changes.
Also, read_packet() needs to call adjust_seek_range_on_packet() if
ds->eof changes. In that case, the decoder also needs to be notified
about EOF. So both of these should be called when ds->eof changes to
true. (Other code outside of this function deals with the case when
ds->eof is changed to false.)
In addition, this code was kind of shoddy about calling wakeup_ds()
correctly. It looks like there was an inverted condition, and sent a
wakeup to the decoder only when ds->eof was already true, which is
obviously bogus. The final EOF case tried to be somehow clever about
checking in->last_eof for notifying the codec, which is sort of OK, but
seems to be strictly worse than just checking whether ds->eof changed.
Fix these things.
This affects async commands started by client API, commands with async
capability run in a sync way by client API (think mpv_command_node()
with "subprocess"), and detached async work.
Since scripts might want to do some cleanup work (that might involve
launching processes, don't ask), we don't unconditionally kill
everything on exit, but apply an arbitrary timeout of 2 seconds until
async commands are aborted.
I wanted to put all commands through mpv_command_node_async() instead of
mpv_command_node(). Using synchronous commands over a synchronous
transport doesn't make sense anyway.
This would have used the request_id field in IPC requests as reply ID
for the async commands. But the latter need to be [u]int64, while the
former can be any type. To avoid that we need an extra lookup table for
mapping reply IDs to request_id values, we now require that request_id
fields are integers.
Since this would be an incompatible change, just deprecate non-integers
for now, and plan the change for a later time.
The only effective difference is that the former explicitly checks
whether the JSON value type is string, and errors out if not. The rest
is exactly the same (mpv_set_property_string is mpv_set_property with
MPV_FORMAT_STRING).
It seems silly to keep this, so just remove it.
Many asynchronous commands are potentially long running operations, such
as loading something from network or running a foreign process.
Obviously it shouldn't just be possible for them to freeze the player if
they don't terminate as expected. Also, there will be situations where
you want to explicitly stop some of those operations explicitly. So add
an infrastructure for this.
Commands have to support this explicitly. The next commit uses this to
actually add support to a command.
If a struct as large as MPContext contains a field named "lock", it
creates the impression that it is the primary lock for MPContext. This
is wrong, the lock just protects a single field.
The "run" command is old. I'm not sure why the separate Lua
implementation was added. But maybe it as because the "run" command used
to be limited to a small number of arguments. This limit has been
removed a while ago. In any case, the old implementation is not needed
anymore.
We keep mp.subprocess() with roughly the same semantics for
compatibility with scripts (including the internal ytdl script).
Seems to work with rhe ytdl wrapper. Not tested further.
This supports named arguments. It benefits from the infrastructure of
async commands.
The plan is to reimplement Lua's utils.subprocess() on top of it.
Named arguments should make it easier to have long time compatibility,
even if command arguments get added or removed. They're also much nicer
for commands with a large number of arguments, especially if many
arguments are optional.
As of this commit, this can not be used, because there is no command yet
which supports them. See the following commit.
Both asynchronous and synchronous calls used to be put into the core's
dispatch queue. Also, asynchronous calls were actually synchronous, just
without forcing a wait on the client's thread. This meant that both
kinds of calls were always strictly ordered.
A longer time ago, synchronous calls were changed to simply lock the
core. This could possibly lead to reordering. Recently, some commands
were changed to run on worker threads, which made the order even looser.
Also remove another now incorrect doxygen comment regarding async
commands.
Basically reimplement the async behavior on top of the async command
code. With this, all screenshot commands are async, and the "async"
prefix basically does nothing. The prefix now behaves exactly like with
other commands that use spawn_thread.
This also means using the prefix in the preset input.conf is pointless
(without effect) and misleading, so remove that.
The each_frame mode was actually particularly painful in making this
change, since the player wants to block for it when writing a
screenshot, and generally doesn't fit into the new infrastructure. It
was still relatively easy to reimplement by copying the original command
and then repeating it on each frame. The waiting is reentrant now, so
move the call in video.c to a "safer" spot.
One way to observe how the new semantics interact with everything is
using the mpv repl script and sending a screenshot command through it.
Without async flag, the script will freeze while writing the screenshot
(while playback continues), while with async flag it continues.
Commands are not a monolithic giant switch() statement anymore, but
individual functions. There's no reason to have the command handlers
themselves in command.c, with a weird under-defined API in between.
(In the future, I'd like to split up command.c further, and when I do
that, scrrenshot.c will probably gets its own mp_cmd_def[] array, and
define the commands locally instead of exporting the raw handlers.)
Basically, the ytdl_hook script will not terminate the script, even if
you change to a new playlist entry. This happens because ytdl_hook keeps
the player core in an early loading stage, and the forceful playback
abort is done only in the ermination code.
This does not handle the "stop" and "quit" commands, which can still
take longer than expected, but on the other hand have some weird special
handling (see below). I'm not doing this out of laziness. Playback
stopping will have to be somewhat redone anyway. Basically we want to
give everything a chance to terminate, and if it doesn't work, we want
to stop loading or playback forcefully after a small timeout. We also
want to remove the mess with input.c's special handling of "quit" and
some other commands (see abort_playback_cb stuff).
It seems the ytdl script like to continue loading external tracks even
if loading was aborted. Trying to do so will still quickly fail, but not
without a load of log noise. So check and error out early.
Pretty trivial, since commands can be async now, and the common code
even provides convenience like running commands on a worker thread.
The only ugly thing is that mp_add_external_file() needs an extra flag
for locking. This is because there's still some code which calls this
synchronously from the main thread, and unlocking the core makes no
sense there.
This enables two types of command behavior:
1. Plain async behavior, like "loadfile" not completing until the file
is fully loaded.
2. Running parts of the command on worker threads, e.g. for I/O, such as
"sub-add" doing network accesses on a thread while the core
continues.
Both have no implementation yet, and most new code is actually inactive.
The plan is to implement a number of useful cases in the following
commits.
The most tricky part is handling internal keybindings (input.conf) and
the multi-command feature (concatenating commands with ";"). It requires
a bunch of roundabout code to make it do the expected thing in
combination with async commands.
There is the question how commands should be handled that come in at a
higher rate than what can be handled by the core. Currently, it will
simply queue up input.conf commands as long as memory lasts. The client
API is limited by the size of the reply queue per client. For commands
which require a worker thread, the thread pool is limited to 30 threads,
and then will queue up work in memory. The number is completely
arbitrary.
The existing thread pool code is the most primitive thread pool
possible. That's fine, but one annoying thing was that it used a static
number of threads. Make it dynamic, so we don't need to "waste" idle
threads.
This tries to add threads as needed. If threads are idle for some time,
destroy them again until a minimum number of threads is reached.
Also change the license to ISC.
This has some tricky interactions. In particular, it requires the core
to be locked due to reading outstanding_async, which is documented on
the only caller only. It's probably better to merge it with its only
caller.
The new code should be strictly equivalent, other than the fact that it
doesn't temporarily unlock+lock when entering the loop for the first
time (which doesn't matter here).
This is almost like rendezvous(), except it allows async wakeup, and
does not require global state. It will be used by a later commit.
struct mp_waiter is intended to be allocated on the stack, and uses an
initializer including PTHREAD_MUTEX_INITIALIZER. This is the first case
in mpv that it uses PTHREAD_MUTEX_INITIALIZER for stack-allocated
mutexes. It seems POSIX still does not allow this formally, but since
POSIX is worth less than used toilet paper, I don't really care. Modern
OSes use futexes, which means you can make _every_ memory location a
lock, and this code tries to make use of it, without using OS specific
code.
The name of the source file is rather generic, because I intend to dump
further small helpers there (or maybe move mp_rendezvous() to it).
Fixes several issues playing back mpegts with video streams marked
as having "still images". For example, see this video which has
frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts
Changes include:
- start playback right away, without waiting for first video frame
- do not consider the sparse video stream in demuxer underrun detection
- do not require multiple video frames for the VO
- use audio as the master stream for demuxer metadata events
- use audio stream for playback time
Signed-off-by: Aman Gupta <aman@tmm1.net>
With the advent of actual HDR devices, my real measured ICC profile has
an "infinite" contrast, since the display is completely off on pure
black inputs. 100k:1 might not be enough, so let's just bump it up to
1m:1 to be safe.
Also, improve the logging in the case that the detected contrast is too
high by default.
This was there originally to detect too-old versions of ffmpeg. We now
only support >= 4.0, so it's not relevant. We just need the dependencies
to be present.
First fix a memory leak when skipping cursor planes by inverting the
check and putting everything, but the free, in the body.
Then fix a missed drmModeFreePlane by simply copying the fields of the
drmModePlane we are interested in and freeing the drmModePlane struct
early.
Fixes `--ytdl-format="dash-fastly_skyfire-video-363357330+dash-fastly_skyfire_sep-audio-363357330" https://vimeo.com/108650530`
This happened because the video track also had audio available and after
adding it expecting an audio-only track, there were no more tracks with video.
Could make it behave differently (and leak memory) in certain cases.
Basically, m_option_parse() randomly returns 0 or 1, but most time 1,
with the difference due to legacy reasons that don't matter anymore.
The "if (prop.name)" check is redundant, because an assert above it
implies that it never can be NULL.
Deduplicate some code for initializing the "prop" variable.
This gets rid of run_command() and its big switch statement, which was
an idiotically big function of almost 1000 lines.
The switch is replaced with a callback per command, and each command is
now implemented in its own function. Command IDs are not needed anymore,
so the mp_command_type enum disappears.
There should be no functional changes, but since this refactors 64
commands, regressions are possible.
The handler() parameter is void*, because in theory the input code is
supposed to be independent of the player core code. For example, you
should be able to reuse the command parser code for some other part of
mpv. In practice, the variable containing command list is defined in the
player core anyway, so you could say this doesn't work. But I'm still
trying to hold onto this idea, so I went with void*.
The plan is to remove the command ID enum. This will happen by replacing
the big switch statement in command.c with dispatching to per-command
callbacks. As preparation, remove uses of the command IDs outside of the
actual dispatching mechanism.
Also remove some instances of checking cmd->def for NULL. We now require
this always to be set.
These are old MPlayer commands that were redundant since 2007 or so. In
2013, mpv explicitly deprecated them (actually removed them, but left
this wrapper, which translated them to modern commands). The list was
not extended since 2013, and mpv always warned on the terminal when a
legacy command was used. So it's time to remove it.
Until recently, ao_lavc and vo_lavc started encoding whenever the core
happened to send them data. Since audio and video are not initialized at
the same time, and the muxer was not necessarily opened when the first
encoder started to produce data, the resulting packets were put into a
queue. As soon as the muxer was opened, the queue was flushed.
Change this to make the core wait with sending data until all encoders
are initialized. This has the advantage that we don't need to queue up
the packets.
The user won't want to have those in the video (I think). The core can
sporadically issue redraws, which is what you want for actual playback,
but not in encode mode. vo_lavc can explicitly detect those and skip
them. It only requires switching to a more advanced internal VO API.
The comments in vo.h are because vo_lavc draws to one of the images in
order to render OSD. This is OK, but might come as a surprise to whoever
calls draw_frame, so document it. (Current callers are OK with it.)
The video timing code could just decide that EOF was reached before it
was displayed. This is not really a problem for normal playback (if you
use something like --keep-open it'd show the last frame anyway,
otherwise it'd at best flash it on screen before destroying the window).
But in encode mode, it really matters, and makes the difference between
having one frame more or less in the output file.
Fix this by waiting for the VO before starting the real EOF.
vo_is_ready_for_frame() is normally used to determine when the VO frame
queue has enough space to send a new frame. Since the VO frame queue is
currently at most 1 frame, it being signaled means the remaining frame
was consumed and thus sent to the VO driver. If it returns false, it
will wake up the playloop as soon as the state changes.
I also considered using vo_still_displaying(), but it's not reliable,
because it checks the realtime of the frame end display time.
While I'm not sure whether it really works, at least it writes the pass1
log correctly now.
How 2-pass stat output is supposed to interact with the new decode API
is rather fishy. ffmpeg.c does the same, and before this change, the
log was not written on EOF (when at least libvpx actually outputs its
stats).
With the internal change from stringlist to keyvaluelist, these
sub-options stop working. I don't really care enough to bring them
back. (Order doesn't matter, -del always seemed annoying.)
Inspired by kmscube, first try to pick the Encoder and CRTC already
associated with the selected Connector, if any. Otherwise try to find
the first matching encoder & CRTC like before.
The previous behavior had problems when using atomic
modesetting (crtc_setup_atomic) when we picked an Encoder & CRTC that
was currently being used by the fbcon together with another Encoder.
drmModeSetCrtc was able to "steal" the CRTC in this case, but using
atomic modesetting we do not seem to get this behavior automatically.
This should also improve behavior somewhat when run on a multi screen
setup with regards to deinit and VT switching (still sometimes you end
up with a blank screen where you previously had a cloned display of
your fbcon)
Add some properties which where forgotten in crtc_setup_atomic.
In both change to not use DRM_MODE_PAGE_FLIP_EVENT | DRM_MODE_ATOMIC_NONBLOCK
flags. This should make it more similar to the drmSetCrtc which it aims to
replace (take effect directly, and blocking call). This also saves us the
trouble of having to set up a poll to wait for pageflip, which would've been
neccesary with DRM_MODE_PAGE_FLIP_EVENT, in both crtc_setup_atomic and
crtc_release_atomic.
This patch will make sure that the video plane is hidden when unused.
When using high resolution modes, typically UHD, and embedding mpv,
having the video plane sitting in the back when you don't play any video
is eating a lot of memory bandwidth for compositing.
That patch makes sure that the video layer is just disabled before and
after playback.
This commit allows to add atomic modesetting when using the atomic renderer.
This is actually needed when using and osd with a smaller size than screen resolution.
It will also make the drm atomic path more consistent
We are currently using primary / overlay planes drm objects, assuming that primary plane is osd and overlay plane is video.
This commit is doing two things :
- replace the primary / overlay planes members with osd and video planes member without the assumption
- Add two more options to determine which one of the primary / overlay is associated to osd / video.
- It will default osd to overlay and video to primary if unspecified
This patch adds
- DRM connector object to atomic context.
- fd property to the drm atomic object as well as a method to read blob type properties.
This allows to ensure that the proper connector is picked up, especially when specifying it
from the commandline, and also allows to make sure we're using the right one when embedding
with interop into an application.
That new API was introduced and allows to have several native resources.
Thisuses that mechanisma for drm resources rather than the deprecated
opengl-cb structs.
This patch therefore add two structs that can be used with the drm atomic interop.
- mpv_opengl_drm_params : which will hold all the drm handles
- mpv_opengl_drm_osd_size : which will hold osd layer size
This commit adds a drm-osd-size=WxH parameter to commandline which
allows to define the OSD plane dimension. OSD can be upscaled to
screen resolution when having OSD at video resolution is too heavy.
This is especially useful for UHD modes on embedded devices where
the GPU cannot handle UHD modes at a decent framerate.
It was documented for --screenshot-template, but apparently never
implemented.
This value is not explicitly rounded, other than for printf() %f default
formatting (which rounds to 6 digits).
This means that a time of 4.5678 is displayed as "00:00:04.568" when the
format string is "%H:%M:%S.%T". Likewise, 59.99999 will be displayed as
"00:01:00.000". Before this change, the sub-ms times were just
truncated.
Requested by TheAMM.
Define a hard-coded value for gl_NumWorkGroups if it is not available.
This adds an additional requirement of needing a shader recompile for
all window size changes.
This was considered a worthwhile compromise as currently f.ex. d3d11
completely lacked any peak computation - this is a major quality of
life upgrade.
Instead of using an internal counter to keep track of the value that was
set last, attempt to find the current value of the property/option in
the value list, and then set the next value in the list.
There are some potential problems. If a property refuses to accept a
specific value, the cycle-values command will fail, and start from the
same position again. It can't know that it's supposed to skip the next
value. The same can happen to properties which behave "strangely", such
as the "aspect" property, which will return the current aspect if you
write "-1" to it. As a consequence, cycle-values can appear to get
"stuck".
I still think the new behavior is what users expect more, and which is
generally more useful. We won't restore the ability to get the old
behavior, unless we decide to revert this commit entirely.
Fixes#5772, and hopefully other complaints.
If playback has not been initialized yet (decoders not initialized
etc.), or if in idle mode, let the track properties cycle through "no"
and "auto". This should be slightly more helpful than making it simply
exit.
Depending on the stage of loading, more could be done. For example, if
youtube-dl loads additional subtitle files, it can happen that these get
added before the main file, and this could be cycled through to an
extent. This is probably too clever, and also sort of dangerous
(unintended interactions with messy in-loading state), so don't do it.
AVFilterContext instances support some additional AVOptions over the
actual filter. This includes useful options like "threads". We didn't
support setting those for the "direct" wrapper (--vf=yadif:threads=1
failed). Change this. It requires setting options on the AVFilterContext
directly, except the code for positional parameters still needs to
access the actual filter's AVOptions.
Strictly speaking redundant, but probably helpful.
In particular I want to push MPV_RENDER_PARAM_ADVANCED_CONTROL. Not
enabling this parameter is actually not very sane.
This is for working around bugs in certain Android devices. At least one
device fails to sort EGLConfigs by size, so eglChooseConfig() ends up
choosing a config with 5/6/5 bits per r/g/b component. The other
attributes in the affected EGLConfigs did not look like they should
affect the sorting process as specified by the EGL 1.4 standard.
The device was reported as:
Sony Xperia Z3 Tablet Compact
Firmware 6.0.1 build number 23.5.A.1.291
GL_VERSION='OpenGL ES 3.0 V@140.0 AU@ (GIT@I741a3d36ca)'
GL_VENDOR='Qualcomm'
GL_RENDERER='Adreno (TM) 330'
Other Qualcom/Adreno devices have been reported as unaffected by this
(including some with same GL_RENDERER string).
"Fix" this by always requiring at least 8 bit. This means it would fail
on devices which cannot provide this. We're fine with this.
mpv-android/mpv-android#112
This was supposed to be a replacement for encode_lavc_discontinuity()
(so we don't need to store last_video_in_pts in a way which requires
synchronization). Unfortunately, VOCTRL_RESET is also called before
termination, and even though it shouldn't matter as far as the VO API is
concerned, it does. It's because vo_lavc.c buffers a frame to compute
the frame duration.
Drop this code. The consequence is that it appears to encode 2 frames
with the same PTS if multiple files are encoded into one. Before this,
it merely dropped a frame (maybe the first of every subsequent file, not
sure).
With -v -v ("debug" level), which is the default for --log-file, this
would log every damn Matroska EBML element and some other uninteresting
things, which was very noisy.
Adjust the log levels to make them less noisy. Also, change some log
calls to MP_ERR for things which are actually errors.
Sometimes this hints that there's a bug, but sometimes it's normal.
Since the code for --end/--frames puts frames that should not be shown
anymore back into the pin, using those options will show this warning
when playback ends. This is a minor annoyance. We could change how it's
done (e.g. set an explicit flag somewhere), but that seems bothersome,
so just change the message from warning to verbose.
The main change is that we wait with opening the muxer ("writing
headers") until we have data from all streams. This fixes race
conditions at init due to broken assumptions in the old code.
This also changes a lot of other stuff. I found and fixed a few API
violations (often things for which better mechanisms were invented, and
the old ones are not valid anymore). I try to get away from the public
mutex and shared fields in encode_lavc_context. For now it's still
needed for some timestamp-related fields, but most are gone. It also
removes some bad code duplication between audio and video paths.
1. I want to get away from mp_image_params (maybe).
2. For encoding mode, it's convenient to get the nominal_fps, which is
a mp_image field, and not in mp_image_params.
Removes a good hunk of weird code.
This loses qscale "emulation", some logging, and the fact that duplicate
keys for values starting with +/- were added with AV_DICT_APPEND. I
don't assign those any importance, even if they are user-visible
changes.
The new M_OPT_ flag is just so that nothing weird happens for other
key-value options, which do not interpret a "help" key specially.
Also rename stereo3d to stereo_in. The only real change is that the
vo_gpu OSD code now uses the actual stereo 3D mode, instead of the
--video-steroe-mode value. (Why does this vo_gpu code even exist?)
Going by ISO 639.2, "und" means "Undetermined". Whatever it's supposed
to mean, in practice it's user for "unset". We prefer if the language
tag remains simply unset in this case.
This removes an ugliness with mp4 in partricular, because libavformat
will export unset languages as such, which affects most mp4 files.
I think this is more intuitive. This requires a dedicated "out" dummy
filter. But keep the "in" dummy filter for symmetry, like in the old
filter code. (We could remove the "in" dummy filter, because the first
actual filter would still show the real input format.)
Attempts to enable the following things:
- let a render API user do "proper" audio-sync video timing itself
- make it possible to not re-render repeated frames if the API user has
better mechanisms available (e.g. waiting for a DisplayLink cycle
instead)
- allow the user to delay or skip redraws if it makes sense
Basically this information will be needed by API users who want to be
"clever" about optimizing timing and rendering.
In MPV_RENDER_PARAM_ADVANCED_CONTROL mode, a simple update callback does
not necessarily make the API user redraw. So handle it differently.
For one, setting vo->want_redraw already uses the "normal" redraw path,
which will call draw_frame() and set next_frame.
Then there are redraws trigered by mpv_render_context_set_parameter(),
which are on the render thread, and would require a separate mechanism.
I decided this is not really a good idea, since it's not even clear that
setting an arbitrary parameter should redraw. Also this could trigger an
unbounded number of redraws. The user can trigger redraws manually if
really needed, depending on the parameter that's being set. If we really
wanted vo_libmpv to do this, we could add a new flag like need_redraw,
which would be 4 lines of code or so.
update() used to require the lock, but now it doesn't matter. It's
slightly better to do it outside of the lock now, in case the update
callback reschedules before returning, and the user render thread tries
to acquire the still held lock (which would require 2 more context
switches).
DR (letting the decoder allocate texture memory) requires running the
allocation on the render thread. This is rather hard with the render
API, because the user controls this thread and when it's entered. It was
not possible until now.
This commit adds a bunch of infrastructure to make this possible. We add
a new optional mode (MPV_RENDER_PARAM_ADVANCED_CONTROL) which basically
lets the user's render thread and libmpv agree how this should be done.
Misuse would lead to deadlocks. To make this less likely, strictly
document thread safety/locking issues. In particular, document which
libmpv functions can be called without issues. (The rest has to be
assumed unsafe.)
The worst issue is destruction of the render context while video is
still active. To avoid certain unintended recursive locks (i.e.
deadlocks, unless we'd make the locks recursive), make the update
callback lock separate. Make "killing" the video chain asynchronous, so
we can do extra work while video is being destroyed.
Because losing wakeups is a big deal, setting the update callback now
triggers a wakeup. (It would have been better if the wakeup callback
were a parameter to mpv_render_context_create(), but too late.)
This commit does not add DR yet; the following commit does this.
This means vf_vapoursynth doesn't need a hack to work around the filter
code, and libavfilter filters now actually get the frame_rate field on
input pads set.
The libavfilter doxygen says the frame_rate field is only to be set if
the frame rate is known to be constant, and uses the word "must" (which
probably means they really mean it?) - but ffmpeg.c sets the field to
mere guesses anyway, and it looks like this normally won't lead to
problems.
This makes ICY title changes show up at approximately the correct time,
even if the demuxer buffer is huge. (It'll still be wrong if the stream
byte cache contains a meaningful amount of data.)
It should have the same effect for mid-stream metadata changes in e.g.
OGG (untested).
This is still somewhat fishy, but in parts due to ICY being fishy, and
FFmpeg's metadata change API being somewhat fishy. For example, what
happens if you seek? With FFmpeg AVFMT_EVENT_FLAG_METADATA_UPDATED and
AVSTREAM_EVENT_FLAG_METADATA_UPDATED we hope that FFmpeg will correctly
restore the correct metadata when the first packet is returned.
If you seke with ICY, we're out of luck, and some audio will be
associated with the wrong tag until we get a new title through ICY
metadata update at an essentially random point (it's mostly inherent to
ICY). Then the tags will switch back and forth, and this behavior will
stick with the data stored in the demuxer cache. Fortunately, this can
happen only if the HTTP stream is actually seekable, which it usually is
not for ICY things. Seeking doesn't even make sense with ICY, since you
can't know the exact metadata location. Basically ICY metsdata sucks.
Some complexity is due to a microoptimization: I didn't want additional
atomic accesses for each packet if no timed metadata is used. (It
probably doesn't matter at all.)
Recursive invocation was needed up until the previous commit. Drop this
feature, and simplify the code. It's more logical, and easier to detect
miuses of the API.
This partially reverts commit 3878a59e. The original reason for it was
removed.
I suppose this doesn't matter in practice, i.e. even if calls relayed
over the dispatch queue will cause WndProc to be invoked, WndProc will
never run for a longer time.
Preparation for removing recursion support from the dispatch queue code.
Fundamentally, scripts are loaded asynchronously, but as a feature,
there was code to wait until a script is loaded (for a certain arbitrary
definition of "loaded"). This was done in scripting.c with the
wait_loaded() function.
This called mp_idle(), and since there are commands to load/unload
scripts, it meant the player core loop could be entered recursively. I
think this is a major complication and has some problems. For example,
if you had a script that does 'os.execute("sleep inf")', then every time
you ran a command to load an instance of the script would add a new
stack frame of mp_idle(). This would lead to some sort of reentrancy
horror that is hard to debug. Also misc/dispatch.c contains a somewhat
tricky mess to support such recursive invocations. There were also some
bugs due to this and due to unforeseen interactions with other messes.
This scripting stuff was the only thing making use of that reentrancy,
and future commands that have "logical" waiting for something should be
implemented differently. So get rid of it.
Change the code to wait only in the player initialization phase: the
only place where it really has to wait is before playback is started,
because scripts might want to set options or hooks that interact with
playback initialization. Unloading of builtin scripts (can happen with
e.g. "set osc no") is left asynchronous; the unloading wasn't too robust
anyway, and this change won't make a difference if someone is trying to
break it intentionally. Note that this is not in mp_initialize(),
because mpv_initialize() uses this by locking the core, which would have
the same problem.
In the future, commands which logically wait should use different
mechanisms. Originally I thought the current approach (that is removed
with this commit) should be used, but it's too much of a mess and can't
even be used in some cases. Examples are:
- "loadfile" should be made blocking (needs to run the normal player
code and manually unblock the thread issuing the command)
- "add-sub" should not freeze the player until the URL is opened (needs
to run opening on a separate thread)
Possibly the current scripting behavior could be restored once new
mechanisms exist, and if it turns out that anyone needs it.
With this commit there should be no further instances of recursive
playloop invocations (other than the case in the following commit),
since all mp_idle()/mp_wait_events() calls are done strictly from the
main thread (and not commands/properties or libmpv client API that
"lock" the main thread).
the icc profile data is mutated to an UnsafeMutablePointer and could
possibly changed. therefore the size of it should be accessed before a
possible change.
This fixes an issue where captions stop rendering after an
in-demuxer-cache seek, because the demuxer keeps waiting to find
a keyframe (ds->skip_to_keyframe set to true in execute_cache_seek).
ffmpeg marks audio tracks which are not meant to be played standalone
as DEPENDENT. these are typically used in DVB broadcasts for audio
descriptions, and are meant to be mixed into the main audio track during
playback.
This was slightly broken: since mp_initialize() did not necessarily
interrupt core_thread() (which is waiting for initialization), it did
not enter mp_play_files(), which would have sent an IDLE event.
I suppose that in some cases (like with mpv-android), the initial IDLE
event was never actually sent, because the first wakeup of the core
thread happens with the "loadfile" command, which will disallow the core
thread an IDLE event.
I changed avio_flush() and introduced avformat_flush() exactly for this
reason.
Used with DVD/BD only (on seeks and when setting the "angle" property).
Seems to work, but wasn't tested too thoroughly (I don't care about
optical discs, I only want this ugly stuff gone that might even violate
the API/ABI).
Although this was never observed to be happening, it seems definitely
possible: we first tell the main thread to exit, and then we ask the
main thread to do some work for us (with mp_dispatch_run()). Obviously
this is racy, and the main thread could exit without doing this, which
would block mp_dispatch_run() forever.
Fix this by changing the order of operation, so that it makes sense.
We could also just store the pthread_t of the main thread in some
variable, but the fact that pthread_create() might set the pthread_t
argument _after_ starting the thread makes this a bit messy (at least it
doesn't seem to be guaranteed on a superficial look at the manpage).
Normally, MPV_RENDER_PARAM* arguments are copied, unless documented
otherwise. Of course we can't copy X11 Display or Wayland wl_display
types, but for arguments that are "summarized" in a struct (like
MPV_RENDER_PARAM_OPENGL_FBO), a copy is expected.
Also add some unused infrastructure to make this explicit, and to make
it easier to add parameter types that require a copy.
Untested.
The first change is about spdif - I mostly ignore spdif issues these
days, but it seems like the recent changes made handling of it slightly
better (but I didn't really test).
The second change is about broken libavfilter filters. We won't restore
the old behavior, because people were complaining about the old behavior
in the past. Possibly we could make libavfilter export this was metadata
and use the old behavior if we know they're broken - but it doesn't
exist yet.
Normally we don't even try this, but in corner cases it can happen. For
example when inserting lavcac3enc at runtime, and display-sync-resample
was active.
Until recently, the AO was reinitialized strictly only on decoder format
changes. But the commit for simplifying audio format negotiation removed
this. Now the AO is recreated for any format change.
This is sort of annoying if you change playback speed. The
insertion/removal of af_scaletempo can change the sample format. For
example, the acompressor filter will convert output to double, so
toggling scaletempo will force the format back to float. This recreates
the AO under the --gapless-audio=weak default. This likely affects a lot
of other filters too.
Work this around by allowing sample format changes, and keeping the
current AO format in these cases. This is probably not a big problem.
Most audio APIs force the output format to float anyway.
This means you actually have to worry about what the default gapless
mode does to your audio. If you start with a file that uses 8 bit per
sample, and then continue playing a 24 bit FLAC, it will be converted
down to 8 bit per sample. (Assuming they are played in a way that uses
the gapless logic.)
Print them as a warning.
Note that there may be some cases where it underruns, without being a
bad condition. This could possibly happen e.g. if the last chunk is
written, and then it resumes playback some time after that. Eventually I
want to add more code to avoid such spurious warnings.
The audio format neogitation code was pretty complicated, although the
idea was simple: when the format changes (or on the first audio frame),
filter only the new frame through the entire filter chain, discard the
resulting frame, but use the format to initialize the AO.
This was useful for "fudging" the channel remix behavior (upmix or
downmix), and moving it before other filters. Apparently this was useful
for things like DRC filters, which might work better in stereo, and
which also can only achieve the desired volume levels by doing it before
a downmix, which would modify the volume. This mechanism was introduced
in commit 60048b7eb9 (which the commit message also describes as
"idiotic heuristic"). Knowing the output format is inherently necessary
for this, because otherwise we can't know what the hell the user defined
filters will do.
There were problems with robustness. Some filters needed more than one
frame. Resampling in particular would discard initial audio at high
resampling ratios. Some filters might drop audio intentionally (like
clipping data on timestamp ranges). There were also allegations that
some decoders output 0 length frames (although that is invalid in
libavcodec). The state machine was excessively complex and hard to
understand too.
There are 3 things that could have been done:
1. Fix robustness problems by doing more heuristics, like repeating
audio frames or simply decoding several frames. Since filters can
behave differently, this would have added lots of complexity.
2. Make use of libavfilter's format negotiation, and add the same to
mpv builtin filters. This is sort of annoying, because the format
negotiation in libavfilter changes the state of the filters. It also
reports only some parameters (mostly all for audio, but a lot of
holes for video). It would remove some of the state machine, but not
all.
3. Drop the channel remix fudging, and do the same as the video chain.
This would not require format negotiation, but instead you can just
filter the audio frames, and look what comes out of it. If nothing
comes out, simply never create an AO.
This commit selects option 3. It removes the remix fudging, which means
the loss of a feature. Users can instead add "--af=format=channels=2"
before their DRC filter, or something. I'm also considering changing the
default for --audio-channels back to stereo, and downmix in the decoder
or at the start of the filter chain, which would give the same results,
except requiring more configuration.
Implementation-wise, this is still a bit different from the video path.
The VO always remains the same instance, while the AO might have to be
recreated on configuration changes. This still requires explicit format
change handling + draining old data, but by putting it into
f_autoconvert, not much new code is needed.
This tried to avoid running the audio/video functions depending on
whether any of the audio or video related format restrictions were
called (so the filter would show an error if a mismatching media type
was passed in). It was a shit idea anyway, so fuck it.
There is a dedicated thread for feeding audio to the ALSA API from a
buffer with a larger size. There is little reason to have such a large
device buffer.
Some shittily muxed files (by a certain HandBrake+libavformat combo)
contain a SeekHead pointing to a SeekHead at the end of the file, which
in turn points to track headers (also at the end of the file). This
failed because the demuxer didn't bother to actually read the elements
listed by the second SeekHead, so no track headers were read, and
playback broke.
Somehow commit 6fe75c38 broke this for no reason. It adds a "needed"
field, which seems completely pointless and replaced the "parsed" flag
in an incomplete way. In particular, the "needed" field was not set when
a _recursive_ SeekHead was read, so those elements were not read. Just
get rid of the field and use "parsed" instead.
The CUDA dynamic loader was broken out of ffmpeg into its own repo
and package. This gives us an opportunity to re-use it in mpv and
remove our custom loader logic.
One can now set the number of buffers and the buffer size.
This can reduce the CPU usage and the total latency stays mostly the same.
As there are sync mechanisms the A/V sync continue intact and working.
It also modifies 6.1 channel order, as per OpenAL spec
and add AOPLAY_FINAL_CHUNK support
OpenAL Soft's AL_SOFT_source_latency extension allows one to correctly
get the device output latency, facilitating the syncronization with
video.
Also added a simpler generic fallback that does not take into account
latency of the device.
Uses OpenAL Soft's AL_DIRECT_CHANNELS_SOFT extension and can be controlled through
a new CLI option, --openal-direct-channels.
This allows one to send the audio data direrctly to the desired channel without
effects applied.
Quickly tested by a person who had FFmpeg linked with libaom.
Seems as simple as the VP9 mappings, where there is no extradata/
initialization data off-band, and just stuff in the packets
themselves.
Do note that the AV1 video format itself at this point is still
not frozen, so what you might produce one day might not be
decodable the following day.
When dump's argument is an array, it was displaying <VISITED> for all
the array's object elements (objects, arrays, etc), regardless if they're
actually visited or not.
The reason is that we try to stringify twice: once normally which may
throw (on cycles), and a second time while excluding visited items which
is indicated by binding the replacer to an empty array - in which we hold
the visited items, where the replacer tests if its 'this' is an array or
not and acts accordingly.
However, its "this" may also be an array even if we don't bind it to one,
because its "normal" this is the main stringified object, so the test of
Array.isArray(this) is true when the top object is an array, and the object
items are indeed are in it - so the replacer considers them visited.
Fix by binding to null on the first attempt such that "this" is an array
only when we want it to test for visited items and not when the argument
itself is an array.
Due to earlier misinterpretation of the Lua docs as if mp.register_idle
registers a one-shot callback, the JS docs suggested to use setTimeout.
But the behavior and Lua docs are such that it's a repeating callback
which fires just before the script thread goes to sleep.
Implement it for JS too.
Although half (non-fast track on sink rate) or one-third (non-fast track not on sink rate) of the buffer size of the created AudioTrack instance as the SL Enqueue buffer size is basically enough for dropout-free playback, only using the full size can avoid stutter upon (re)start of playback.
Here are the various buffer sizes on different track/sink rate when on Bluetooth audio on Android O:
aptX @ 48kHz:
Sink rate: 48000 Hz
44100 Hz: 10632 frames (241.09 ms)
48000 Hz: 11544 frames (240.50 ms)
88200 Hz: 21216 frames (240.54 ms)
96000 Hz: 23088 frames (240.50 ms)
176400 Hz: 42384 frames (240.27 ms)
192000 Hz: 46128 frames (240.25 ms)
SBC/AAC/aptX @ 44.1kHz:
Sink rate: 44100 Hz
44100 Hz: 10776 frames (244.35 ms)
48000 Hz: 11748 frames (244.75 ms)
88200 Hz: 21552 frames (244.35 ms)
96000 Hz: 23448 frames (244.25 ms)
176400 Hz: 43056 frames (244.08 ms)
192000 Hz: 46848 frames (244.00 ms)
The above results were produced with the following code:
import android.media.AudioAttributes;
import android.media.AudioFormat;
import android.media.AudioTrack;
class AudioInfo {
public static void main(String[] args) {
int nosr = AudioTrack.getNativeOutputSampleRate(3);
System.out.printf("Sink rate: %d Hz\n", nosr);
int[] rates = {44100,48000,88200,96000,176400,192000};
for (int rate: rates) {
AudioAttributes aa = new AudioAttributes.Builder().setFlags(256).build();
AudioFormat af = new AudioFormat.Builder().setSampleRate(rate).build();
AudioTrack at = new AudioTrack(aa, af, 4, 1, 0);
int sr = at.getSampleRate();
int bs = at.getBufferSizeInFrames();
float ms = bs * (float) 1000 / sr;
at.release();
System.out.printf("%d Hz: %d frames (%.2f ms)\n", sr, bs, ms);
}
}
}
Therefore bumping the device buffer size to 250ms.
On machines with multiple GPUs, /dev/dri/renderD128 isn't guaranteed
to point to a valid vaapi device. This just adds the option to specify
what path to use.
The old fallback /dev/dri/card0 is gone but that's not a loss as its
a legacy interface no longer accepted as valid by libva.
Fixes#4320
libavcodec normally drops subtitle lines that fail a check for invalid
UTF-8 (their check is slightly broken too, by the way). This was always
annoying and inconvenient, but now there is a mechanism to prevent
it from doing this. Requires newst libavcodec.
There was a "generic" function to run a hook and to wait for its
completion, yet there were two duplicated functions doing the same
anyway. Replace them with a single function.
They differed in how stop_play was handled, but it was broken anyway.
stop_play is set when playback is stopped due to quitting or changing
the playlist entry - but we still can't stop hook processing, because
that would mean asynchronously doing something else while the user hook
code is still busy and might still have the expectation that running the
hook stops everything else. So not waiting until the hook ends properly
is against the whole hook idea. That this was done inconsistently is
even worse. (Though it could be argued that when quitting the player,
everything should just be stopped violently. But I still think that's
up to the hook handler.)
process_hooks() does not return anything, since hook processing doesn't
really have a result (it's all about blocking and letting some other
code synchronously do something). Just let the caller check whether
loading was aborted in the meantime.
Also change the potentially misleading name of mp_hook_run().
As it turns out, there are multiple libmpv users who saw a need to
use the hook API. The API is kind of shitty and was never meant to be
actually public (it was mostly a hack for the ytdl script).
Introduce a proper API and deprecate the old one. The old one will
probably continue to work for a few releases, but will be removed
eventually.
There are some slight changes to the old API, but if a user followed
the manual properly, it won't break.
Mostly untested. Appears to work with ytdl_hook.
Move all of this stuff to a common function. This makes the error
messages less specific, but I don't think anyone will miss it.
The OSD flag handling is annoying, but it's nothing that should be
changed with this commit.
I think this will help with reducing code duplication (see following
commit). The error messages loses the multiplication factor, but the
error message will be replaced by a generic one in the following commit
anyway.
Hardware decoding things often need access to additional handles from
the windowing system, such as the X11 or Wayland display when using
vaapi. The opengl-cb had nothing dedicated for this, and used the weird
GL_MP_MPGetNativeDisplay GL extension (which was mpv specific and not
officially registered with OpenGL).
This was awkward, and a pain due to having to emulate GL context
behavior (like needing a TLS variable to store context for the pseudo GL
extension function). In addition (and not inherently due to this), we
could pass only one resource from mpv builtin context backends to
hwdecs. It was also all GL specific.
Replace this with a newer mechanism. It works for all RA backends, not
just GL. the API user can explicitly pass the objects at init time via
mpv_render_context_create(). Multiple resources are naturally possible.
The API uses MPV_RENDER_PARAM_* defines, but internally we use strings.
This is done for 2 reasons: 1. trying to leave libmpv and internal
mechanisms decoupled, 2. not having to add public API for some of the
internal resource types (especially D3D/GL interop stuff).
To remain sane, drop support for obscure half-working opengl-cb things,
like the DRM interop (was missing necessary things), the RPI window
thing (nobody used it), and obscure D3D interop things (not needed with
ANGLE, others were undocumented). In order not to break ABI and the C
API, we don't remove the associated structs from opengl_cb.h.
The parts which are still needed (in particular DRM interop) needs to be
ported to the render API.
At least the libavcodec MediaCodec wrapper sometimes seems to break the
libavcodec API, and does the following sequence:
send_packet() -> EAGAIN
receive_frame() -> EAGAIN
send_packet() -> OK
The libavcodec API never allows returning EAGAIN from both functions, so
we discarded packets in this case. Change it to retrying decoding, for
the sake of MediaCodec. Note that it could also happen due to internal
bugs in the vd_lavc.c hw fallback code, but if there are any remaining,
they should be fixed properly instead.
Requested.
When this happens, network calls are forcibly aborted (more or less),
but demuxers might keep going, as most of them do not check for forced
exits properly. This can possibly lead to broken packets being added.
Also do not attempt to read more packets in this situation.
Also do not print a stream open failed message if opening was aborted
anyway.
we rendered on the displaylink thread which wasn't the best idea. if
rendering took too long or was blocking it also blocked the displaylink
callback. when that happened new vsyncs were reported delayed or not at
all. consequently the mpv_render_context_report_swap function wasn't
called consistently and that could cause bad video playback. so the
rendering is moved to a dedicated dispatch queue. furthermore the update
callback starts a layer update directly instead of the displaylink
callback, making the rendering a bit more consistent.
Previously, section titles (File/Video/Audio) were printed as suffix of
a property that was assumed to always exist. However, with e.g.
lavi-complex this is not the case, therfore, print them without being
dependent on a property.
Switch from audio|video to audio|video-out-params properties as per
recommendation in #5670. These properties are tables and include
information explicitly queried later, so switch to using these tables
and reduce the amount of queried properties.
Fixes#5670
Right now the atomic request is alive during the renderloop.
We want it to be alive until the drm egl context is destroyed because some properties
might still be set upon interop close
This patch make the request to be kept created even outside the renderloop.
The context uninit will commit the last request.
commit 2edf00f changed the MPV_EVENT_SHUTDOWN behaviour slightly, such
that it will only be sent once. cocoa-cb relied on it being sent
continuously till all mpv_handles are destroyed. now it manually shuts
down and destroys the mpv_handle after the animation instead of relying
on this removed behaviour.
This passed the display size as source size to the renderer, which is of
course nonsense. I don't know what I was doing in 569383bc54.
Yet another fix for those damn anamorphic videos.
As a somewhat redundant/cosmetic change, use image_params instead of
real_image_params in the code above. They should have the same, dimensions
(but possibly different formats when doing hw decdoing), and mixing them
is confusing. p->image_params wins because it's shorter.
Actually fixes#5619.
There is some sort-of awkwardness here, because option access needs to
happen in a synchronized manner, and the framedrop flag is not in the VO
option struct. Remove the mp_read_option_raw() call and the awkward
change notification via VO_EVENT_WIN_STATE from command.c, and pass it
through as new vo_frame flag.
Removes the awkward notification through VO_EVENT_WIN_STATE.
Unfortunately, some awkwardness remains in mp_property_display_fps(),
because the property has conflicting semantics with the option.
It's a WTF that we have something as specific in the API. It could be
argued that we should provide helpers for other language and GUI toolkit
combinations. Obviously that's not going to scale, and it's somewhat
likely that it will bitrot. The rest is said in the API changelog.
We took the storage size instead of the display size for "unscaled"
screenshots. Even if it's called "unscaled", it's still supposed to
scale to compensate for aspect ratio.
(How many commits fixing anamorphic screenshots in various situations
are there?)
Fixes#5619.
vf_vdpaupp crashed on certain files (with --hwdec=vdpau --deinterlace).
This happened for example with mpeg2 files, which for some reason
typically contain some AVFrame side data. It turns out the last change
in 55c88fdb8f was not quite clean, and forgot the special cases in
mp_image_new_dummy_ref(). This function is supposed to copy all metadata
from the argument passed, except buffer refs. But there were new buffer
refs, that were not cleared properly. Also, the ff_side_data pointer
must be cleared, or the new mp_image would try to free it on
destruction.
The bottom line is that mp_image_new_dummy_ref() is a pretty bad idea,
and I suppose all callers with non-NULL arguments should be changed to
create a blank mp_image, and copy frame properties as needed (this
includes callers of mp_image_new_custom_ref()).
Fixes#5630.
Before this change, mpv_wait_event() could inconsistently return
multiple MPV_EVENT_SHUTDOWN events to a single mpv_handle, up to the
point of spamming the event queue under certain circumstances. Change
this and just send it exactly once to each mpv_handle.
Some client API users might have weird requirements about destroying
their state asynchronously (and not reacting immediately to the SHUTDOWN
event). This change will help a bit to make this less weird and
surprising.
This changes how mpv_terminate_destroy() and mpv_detach_destroy()
behave. The doxygen in client.h tries to point out the differences. The
goal is to make this more useful to the API user (making it behave like
refcounting).
This will be refined in follow up commits.
Initialization is unfortunately closely tied to termination, so that
changes as well. This also removes earlier hacks that make sure that
some parts of FFmpeg initialization are run in the playback thread
(instead of the user's thread). This does not matter with standard
FFmpeg, and I have no reason to care about this anymore.
The intention is to reduce annoying differences between mpv CLI and
libmpv, and there's no reason to have the locale check only in libmpv
(although it doesn't help with any real issues either).
the first mouse events, that try to hide the title bar, could happen
before the title bar was actually initialised. that caused our hiding
code to access a nil value. check for an available title bar before
trying to hide it.
there were actually a few small problems. the fatalError() function
wasn't supposed to be called there and caused an "Illegal instruction".
this was replaced by a print and exit() call. the second problem was
that cocoa returns a kCGLBadPixelFormat instead of a kCGLBadAttribute
error, which broke our check, immediately exited our loop and no working
pixel format was ever created. the third problem was that macOS 10.12
didn't return any errors but also didn't return a pixel format, that
also broke our check. now the code checks for both cases.
Fixes#5631
when we transitioned to the new libmpv API with commit ae29725 i
reintroduced an old bug that was fixed with 7f714c6 because of a dumb
rebasing error on my part. the branch i based the libmpv changed on was
originally without the fbo fix and on rebasing i forgot to change the
variable to the proper one, basically deactivating the fix.
mouse events and the tracking area are needed for (un)hiding the new
title bar, which was broken when input-cursor=no was set. no tracking
area was ever created and set which completely deactivated any mouse
events. the specific mouse event functions were already deactivated
proactively and have the needed check. no events are being propagated to
the mpv core when input-cursor=no is set, even with an active tracking
area.
Overall, just shuffled code around and added a few debugging messages
for future issues.
The issue could be reproduced easily by quickly navigating through the
playlist inside a network mount.
Closes#5618
The s_size() function, whatever it was supposed to do, caused the
surface size to increase indefinitely. Fix by making it always use the
maximum size that was last used, which is less optimal (many surface
recreations when making the window slowly larger), but at least it
works.
The rotation code didn't mark the old surface as invalid when it was
freed, so it could destroy random other surfaces (let's call it dangling
ID).
Also, the required rotation surface size depends on the rotation mode,
so recreate the surfaces on rotation as well.
If you set desired.samples to 0, SDL will return a default buffer size
on obtained.samples. This was broken, because ceil_power_of_two(0)
returns 1. Since 0 is usually not considered a power of two, this is
probably correct, but we still want to set desired.samples to 0 in this
case.
You can use --audio-buffer=0 to minimize the audio buffer size. But if
the AO reports no device buffer size (like e.g. ao_jack does), then the
buffer size is actually 0, and playback can never work properly.
Make it fallback to a size of 1, which is unlikely to work properly, but
you get what you asked for, instead of a freeze.
Since the demuxer cache addition, ds->queue->head can actually be set to
non-NULL, but the decoder can still be at EOF (with no packets to come).
This made it report an unknown buffered size, instead of 0. Fix this by
checking the decoder part of the packet queue instead.
Probably doesn't matter much, but fixes an annoying "???" on the CLI
status line in some situations.
there were several problems that had to be fixed because of differences
between python2 to python3:
- subprocess.check_output returned an unicode instead of a string
- filter() returns an iterator instead of a list
- recursion limit was reached
first two were fixed by explicitly converting to the needed type or
using the proper function invocation. third was fixed by changing the
recursive process_libraries function to an iterative one.
Fixes#5600, #3316
While the soft buffer size is already by default 200ms, it is not enough to guarantee dropout-free playback on Bluetooth audio. Bumping the device buffer size to the same value seems to suffice.
We use triple buffering for this interop and we were only unreffing the
data structures, which doesn't destroy the drm buffers.
This patch allows to make sure that we release the drm buffers on
playback end.
we activated the rendering loop a bit too early and it was possible that
the first draw function was called before it was actually ready. this
was a remnant from the old init routine and should have been changed.
start the queue on reconfigure instead of preinit.
on a file change and when the aspect ratio of the window changed, the
first live resize state had a wrong aspect ratio because the new aspect
ratio was only set after the first resize. just set the new content
frame before the resize.
i tried being smart and handle aspect ratio differences manually via
atomic drawing and resizing to aspect fitted frames. there were a few
issues with that. like unexpected visibility of certain System GUI
elements on entering fullscreen or visually dropped frames due to the
atomic drawing. now we rely on system mechanics to keep the proper
aspect ratio of our layer, the recommended way. as a side effect it also
fixes a segfault.
Fixes#5581
It turns out that Mali drivers are likely broken, and do not return
GBM_FORMAT_ARGB8888 (they return GBM_FORMAT_XRGB8888) when getting
EGL_NATIVE_VISUAL_ID for any EGLConfig, even though the resulting
EGLConfig appears to be capable of alpha.
It could also be potentially useful to allow an ARGB EGLConfig used
with an XRGB framebuffer on some platforms, so we do that. (cf. weston)
Unrelated indentation fix in gbm_format_to_string.
Naturally, there's more than one fourcc that indicates an mjpeg
stream.
I have a particular ancient webcam here (Logitech QuickCam Messanger)
that only supports the single 'JPEG' format, but there are other
devices out there which support both 'JPEG' and 'MJPG' with no visible
differences, and others where the streams are slightly different.
Regardless of those details, it remains correct to treat 'JPEG'
the same as 'MJPG' from a stream consumption perspective.
This adds key bindings for some semi-popular features. It also tries to
cleanup some old bindings. For example w/e for panscan is now changed to
w/W. In all cases, the old bindings are still kept and work, though.
Part of an ongoing attempt to cleanup the default key bindings.
See #973 for some context.
this API was introduced in c5e4538 solely for the use with cocoa-cb and
was not public. with the port to the new API in 760d471 it is now unused
and can safely be removed.
it's possible to get a function pointer through a closure after all in
swift. remove the GL dummy function from the c header and do it in the
swift code instead.
After switching, the playback state was not reset, which could leave it
in a strange, pause like state, that could be fixed by e.g. seeking.
This seems to be an older regression - it's even in 0.27.
Sometimes, playback needs to be fully uninitialized and reinitialized
without "officially" closing and reopening the playlist entry. This
happens with PT_RELOAD_FILE, which is triggered by edition switching and
also DVD/BD title switching. (Not really sure why it goes through so
much pain for such obscure cases. All it gains is not resetting "local"
options, and not signaling a reload to the client API. Whatever.)
The recent filter change freed filter_root too early without recreating
it, so it crashed on edition switching.
Fixes#5587.
If you used --aufio-file=file.mkv, and file.mkv included a video track
marked as default, then the logic in select_default_track() would pick
the video track from file.mkv. This is 100% broken, so fix it.
The playback start logic explicitly waits until the first frame has been
displayed. Usually this will introduce a wait of 1 vsync. For normal
playback this doesn't matter, but with respect to low latency needs,
this only leads to additional data getting queued up in the demuxer or
network buffers.
Another thing is that the timing logic decodes 1 frame ahead (= 1 frame
extra latency) to determine the exact duration of a frame.
To be fair, there doesn't really seem to be a hard reason why this is
needed. With the current code, enabling the option does lead to A/V
desync sometimes (if the demuxer FPS is too inaccurate), and also frame
drops at playback start in some situations. But this all seems to be
avoidable, if the timing logic were to be rewritten completely, which
should probably happen in the future. Thus the new option comes with the
warning that it can be removed any time. This is also why the option has
"hack" in the name.
Well I guess it doesn't help that much.
Also add some stuff that might help to the manpage.
The fundamental problem with some "live" sources (e.g. x11grab) is
actually that the player gets behind initially, and never thinks it has
to catch up. This is also why --untimed can help.
It's a mess: mp3 files have user tags as global metadata (because the
id3v2 tag is global and there is only 1 stream), while OGG files have it
per-track (because it's per-stream on the lowest level). mpv needs to
try to make something nice out of the mess.
It did so by trying to detect audio-only OGG files, and then copying the
per-stream metadata to the global metadata. Make the heuristic for
detecting this slightly more clever, so it works for files with extra,
unrelated streams, like the awful libavformat cover art hack.
Fixes#5577.
Useful for libavfilter. Somewhat risky, because we can't ensure the
consistency of the unknown side data (but this is a general problem with
side data, and libavfilter filters will usually get it wrong too _if_
there are conflict cases).
Fixes#5569.
We must not create new references herem because mp_image_new_ref() is
called later, and actually creates new references (including doing
actual error checking). Blame C, not me.
It appears some (or all) mkv files with EAC3 are muxed in a way that
breaks FFmpeg's spdifenc. I suspect it's because either dependent
substream packets are localted in their own packets, or the reverse. Or
possibly this is case where the muxer did not respect packet boundaries
at all. Enabling the EAC3 parser seems to fix this anyway, because why
waste your precious time on retarded Dolby bullshit technology? (Which
idiot came up with this shitty substream garbage?)
Observed with dolby_digital_plus_channel_check_lossless-DWEU.mkv.
Fixes#5578.
I think DASH playback tends to work much better with FFmpeg's DASH
demuxer, which requires libxml2.
For nvdec, FFmpeg git master now requires the external nvidia headers,
since the builtin ones were removed.
the NSWindowButton enum was moved to be a member of NSWindow and renamed
to ButtonType in SDK 10.13. apparently that wasn't documented anywhere.
not even in the SDK changes Document and the official Documentations
makes it look like it was always like this. the old NSWindowButton enum
though is still around on SDK 10.13 or at least got a typealias. so we
will just use that.
The purpose of the new API is to make it useable with other APIs than
OpenGL, especially D3D11 and vulkan. In theory it's now possible to
support other vo_gpu backends, as well as backends that don't use the
vo_gpu code at all.
This also aims to get rid of the dumb mpv_get_sub_api() function. The
life cycle of the new mpv_render_context is a bit different from
mpv_opengl_cb_context, and you explicitly create/destroy the new
context, instead of calling init/uninit on an object returned by
mpv_get_sub_api().
In other to make the render API generic, it's annoyingly EGL style, and
requires you to pass in API-specific objects to generic functions. This
is to avoid explicit objects like the internal ra API has, because that
sounds more complicated and annoying for an API that's supposed to never
change.
The opengl_cb API will continue to exist for a bit longer, but
internally there are already a few tradeoffs, like reduced
thread-safety.
Mostly untested. Seems to work fine with mpc-qt.
The stdatomic emulation adds "_" to each variable used inside the
macros, to avoid that compilers print -Wshadow warnings for identifiers
that are also used in surrounding code. Do this more consistently,
because new warnings have been showing up.
when resizing async it's possible that the layer, and the underlying gl
surface, is stretched on an aspect ratio change. to prevent that we do
an atomic resize (resize and draw at the same time). usually max one
unique frame should be dropped but it's possible, depending on the
performance, that more are dropped.
the title bar is now within the window bounds instead of outside. same
as QuickTime Player. it supports several standard styles, two dark and
two light ones. additionally we have properly rounded corners now and
the borderless window also has the proper window shadow.
Also make the earliest supported macOS version 10.10.
Fixes#4789, #3944
By blocking the VT switcher signal in the VO thread we get less races
and other oddities.
This gets rid of tearing (at least for me) when VT switching with
--gpu-context=drm.
crtc_setup gets called on VT reacquire as well as during normal setup. When
called during VT reacquire p->front_buf might not be 0, so the maths was wrong,
and could cause array OOB errors. Use mathematically correct (for negative
numbers) modulo to always pick the farthest away buffer (should work
even for larger values of BUF_COUNT).
The VT switcher was being set up, but it was being neither polled nor
interrupted.
Insert wait_events and wakeup functions based on those from vo_drm,
and add return early in drm_egl_swap_buffers if p->active isn't set.
This should get basic VT switching working, however there will likely
still be some random glitches. Switching between mpv and X11/weston is
unlikely to work satisfactorily until we can solve the problems with
drmSetMaster and drmDropMaster.
This introduces the option --drm-format (currently used only by
context_drm_egl, vo_drm implementation is pending) which allows you to
pick between a xrgb8888 or a xrgb2101010 visual for --gpu-context=drm.
Requires a recent mesa (18.0.0_rc4 or later) to work.
This also fixes a bug when using --gpu-context=drm on a 30bpp-enabled
mesa (allow_rgb10_configs set to true). Previously it would've set up
an XRGB8888 format at the DRM/GBM level, while a 30bpp EGLConfig would
be picked, resulting in a garbled image.
even though the fullscreen animation has a shorter duration than the
system wide animation (space sliding effect) there are still cases where
it takes longer, eg performance issues (especially on init). furthermore
the final size of the animation is usually different than the actual
fullscreen size because of spect ratio differences. the actual resize to
fullscreen is done automatically by cocoa itself when the actual
transition to fullscreen happens (system event). so it could happen that
the last animation resize happened after the actual resize to fullscreen
leading to a wrongly sized frame after entering fullscreen. to prevent
this we cancel the animation when entering fullscreen, we always set the
proper frame size when in fullscreen and discard any other frame sizes,
and to prevent some performance problems on init we push entering
fullscreen to the end of the main queue to execute it when most of the
init routines are done.
Fixes#5525
on live resize, eg async resize, the layer's bounds size is not in sync
with the actual surface size. this led to a wrongly sized frame and a
perceived flicker. get and use the actual surface size instead.
Mobius isn't well-defined for sig_peak <= 1.0. We can solve this by just
soft-clamping sig_peak to 1.0. Although, in this case, we can just skip
tone mapping altogether since the limit of mobius as sig_peak -> 1.0 is
just a linear function.
Based on testing with real-world non-HDR BT.2020 clips, clipping the
color space looks better than attempting to gamut map using a tone
mapping shader that's (by now) optimized for HDR content.
If anything, we'd have to develop a separate gamut mapping shader that
works in LCh space.
Do this because retrying reading on higher levels (like the demuxer)
usually causes tons of problems. A hack like this is simpler and could
allow to remove some of the higher level retry behavior.
This works by trying to detect whether the file is appended. If we reach
EOF, check if the file size changed compared to the initial value. If it
did, it means the file was appended at least once, and we set the
p->appending flag. If that flag is set, we simply retry reading more
data every time we encounter EOF. The only way to do this is polling,
and we poll for at most 10 times, after waiting for 200ms every time.
Just bail out immediately (and disable audio) if format probing has no
result, instead of doing nothing and then apparently freezing.
This can happen with bogus filters, cases where the first audio frame is
essentially dropped by filters (can happen with large resampling
factors), and if the audio track contains no packets at all, or all
packets fail to decode.
in certain circumstances the returned fbo for drawing is 0, but that
fbo is solely used internally by the CAOpenGLLayer for its drawing and
should never be used. in that case we fallback to 1 or the last used fbo
instead if it was not 0.
Fixes#5546
When pixels are non-square, the appropriate value of vo->monitor_par is
necessary to determine the destination rectangle, which in turn tells
how to scale the video along the x and y axis. Before this commit, the
drm driver only used --monitorpixelaspect. For example, to play a video
with the right aspect on a 4:3 screen and 640:400 pixels,
--monitorpixelaspect=5:6 had to be given.
With this commit, vo->monitor_par is determined from the size of the
screen in pixels and the --monitoraspect parameter. The latter is
usually easier to determine than --monitorpixelaspect, since it is
simply the proportion between the width and the height of the screen,
in most cases 16:9 or 4:3. If --monitoraspect is not given,
--monitorpixelaspect is used if given, otherwise pixel aspect is
assumed 1:1.
This solves a number of problems simultaneously:
1. When outputting HLG, this allows tuning the OOTF based on the display
characteristics.
2. When outputting PQ or other HDR curves, this allows soft-limiting the
output brightness using the tone mapping algorithm.
3. When outputting SDR, this allows HDR-in-SDR style output, by
controlling the output brightness directly.
Closes#5521
The HLG OOTF is defined as a one-parameter family of OOTFs depending on
the display's peak luminance. With the preceding change to OOTF scale
and handling, we no longer have any issues with outputting values in
whatever signal range we need.
So as a result, it's easy for us to support a tunable OOTF which may
(drastically) alter the display brightness. In fact, this is also the
only correct way to do it, because the HLG appearance depends strongly
on the OOTF configuration. For the OOTF, we consult the mastering
display's tagging (via src.sig_peak). For the inverse OOTF, we consult
the output display's target peak.
The primary need for this change is the fact that the OOTF was
incorrectly scaled, due to the fact that the application of the OOTF can
itself change the required normalization peak. (Plus, an oversight in
pass_inverse_ootf meant we forgot to normalize at the end of it)
The linearize/delinearize functions still normalize the scale since it's
used in a number of places throughout gpu/video.c, but the color
management function now converts to absolute scale right away, instead
of in an awkward way inside the tone mapping branch. The OOTF functions
now work in absolute scale only.
In addition, minor changes have been made to the way normalization is
handled for tone mapping - we now divide out the dst_peak *after* peak
detection, in order to make the scale of the peak detection buffer
consistent even if the dst_peak were to (hypothetically) change
mid-stream. In theory, we could also do this for desaturation, but doing
the desaturation before tone mapping has the advantage of preserving
much more brightness than the other way around - and even mid-stream
changes are not that drastic here.
Finally, some preparation work has been done for allowing the user to
customize the `dst.sig_peak` in the future.
drawing off-screen failed because we didn't have a valid context. the
problem is we force off-screen drawing because the CAOpenGLLayer refuses
to draw anything while being off-screen. set the current context before
starting to draw anything off-screen.
Fixes#5530
This fixes playback stalls on some mediacodec hardware decoders,
which expect that frame buffers will be rendered and returned back
to the decoder as soon as possible.
Specifically, the issue was observed on an NVidia SHIELD Android TV,
only when playing an H264 sample which switched between interlaced
and non-interlaced frames. On an interlacing change, the decoder
expects all outstanding frames would be returned to it before it
would emit any new frames. Since a single extra frame always remained
buffered by mpv, playback would stall. After this commit, no extra
frames are buffered by mpv when using vo_mediacodec_embed.
Coverity complained about the redundant init of hratio etc. - just
remove that and merge declaration/init of these variables. Also the
first double cast in each expression is unnecessary.
The rest of the function should be executed only if both are set. It
seems like in practice this didn't happen yet with only one of them
unset, but in theory it's possible. Found by Coverity.
the CVDisplayLinkSetOutputHandler function introduced with 10.11 is
broken on the very same version of the OS, which caused our render loop
never to start. fallback to the old display link callback on 10.11.
for reference the radar http://www.openradar.me/26640780Fixes#5527
the swift object file wasn't linked when libmpv was linked, which
resulted in a missing symbol error. add the swift object to the linking
list for libmpv too.
Fixes#5522
for convenience reasons i used strings for subprocess commands instead
of command lists, which made it mandatory to use a shell. for security
reasons, among others, we removed the shell usage and converted the
commands to actual command lists.
c82fed8 fixed the detection with python3 but broke it on python2. the
decode function on python2 converts the str to unicode which causes
problems when concatenating to str when building. instead of decoding
the output we change the subprocess to operate in text mode.
also use check_output instead of Popen for simplicity.
Usable for uniquely identifying mpv instances from
subprocesses, controlling mpv with AppleScript, ...
Adds a new mp_getpid() wrapper for cross-platform reasons.
There is now a better way. Reading the font framebuffer was always a
hack. The new code via VOCTRL_SCREENSHOT renders it into a FBO, which
does not come with the disadvantages of reading the front buffer (like
not being supported by GLES, possibly black regions due to overlapping
windows on some systems).
For now keep VOCTRL_SCREENSHOT_WIN on the VO level, because there are
still some lesser VOs and backends that use it.
Also print type and help string. Also print AV_OPT_TYPE_CONST, which are
like the mpv choice option type, except different. Print them as
separate lines because FFmpeg usually has help strings for them too.
Most options starting with --no-<name> are automatically translated to
--<name>=no. Make the code slightly nicer by using a flag instead of
explicitly comparing option types. Also fix an issue that made the
option parser print nonsense error messages for if --no-... was used for
options which don't support it.
This should be helpful for the new OSX Cocoa backend, which uses
opengl-cb internally. Since it comes with a behavior change that could
possibly interfere with libmpv/opengl_cb users, we mark it as explicit
API change.
This includes codec/muxer/demuxer iteration (different iteration
function, registration functions deprecated), and the renaming of
AVFormatContext.filename to url (plus making it a malloced string).
Libav doesn't have the new API yet, so it will break. I hope they will
add the new APIs too.
There was the following problem: if a filter graph had asynchronous
filters, and the filter graph user did not call mp_filter_run() (and
only accessed the mp_pins), then filtering could stall, because using
mp_pin_out_request_data() only recursively invoked filtering if the
data_requested flag wasn't already set. The latter can happen if a
request was tried earlier but failed, and then an asynchronous filter
actually produced output that would satisfy the request. Obviously, it
has to invoke filtering again to get the requested frame.
Fix this by organizing the code differently, and making sure to invoke
mp_filter_run() on every request (if there's nothing to do, it doesn't
do anything anyway). Simplify it a bit by removing things which are not
needed, like connecting filter graphs with different root filters.
Reduce backward/forward from 400MB/400MB to 50MB/150MB. Too many
complaints about high memory usage.
Note that external tracks (like ytdl DASH with external audio tracks)
will double the amounts, because an external track uses its own demuxer
and cache.
Avoids that the audio decoder shows up with a "[root/ad]" log prefix.
This is an annoying consequence of mp_log_new(): if a log parent doesn't
have a prefix with "!", it'll add the prefix to all mp_logs created from
it. This should probably be fixed in the mp_log code itself, but doing
so would be a big deal as we'd have to make sure all the other log
prefixes are what we want. So work it around for now.
This switches the default away from "bob" to the best algorithm reported
as supported by the driver. This is convenient for users, and there is
no reason to use something worse by default.
Untested.
Before this, we made deinterlacing dependent on the video codec metadata
(AVFrame.interlaced_frame for libavcodec). So even if --deinterlace=yes
was set, we skipped deinterlacing if the flag wasn't set. This is very
unreliable and there are many streams with flags incorrectly set.
The potential problem is that this might upset people who alwase enabled
deinterlace and hoped it worked. But it's likely these people were
screwed by this setting anyway. The new behavior is less tricky and
easier to understand, and this preferable. Maybe one day we could
introduce a --deinterlace=auto, which does the right thing, but of
course this would be hard to implement (esecially with hwdec).
Fixes#5219.
The recent changes to player/audio.c moved PTS jump detection to after
audio filtering. This was mostly done for convenience, because dataflow
between decoder and filters was made "automatic", and jump detection
would have to be done as filter. Now move it back to after decoders,
again out of convenience. The future direction is to make the dataflow
between filters and AO automatic, so this is a bit in the way. Another
reason is that speed changes tend to cause jumps - these are legitimate,
but get annoying quickly.
This allows the new GPU screenshot functionality introduced in
9f595f3a80 to work with the D3D11 backend. It replaces the old window
screenshot functionality, which was shared between D3D11 and ANGLE. The
old code can be removed, since it's not needed by ANGLE anymore either.
Similar spirit to edb4970ca8. check_gl_features() has a confusing
early-return. This also adds compute_hdr_peak to the list of options
that is copied to the dumb-mode options struct, since it seems to make a
difference. Otherwise it would be impossible to disable HDR peak
detection in dumb mode.
this is meant to replace the old and not properly working vo_gpu/opengl
cocoa backend in the future. the problems are various shortcomings of
Apple's opengl implementation and buggy behaviour in certain
circumstances that couldn't be properly worked around. there are also
certain regressions on newer macOS versions from 10.11 onwards.
- awful opengl performance with a none layer backed context
- huge amount of dropped frames with an early context flush
- flickering of system elements like the dock or volume indicator
- double buffering not properly working with a none layer backed context
- bad performance in fullscreen because of system optimisations
all the problems were caused by using a normal opengl context, that
seems somewhat abandoned by apple, and are fixed by using a layer backed
opengl context instead. problems that couldn't be fixed could be
properly worked around.
this has all features our old backend has sans the wid embedding,
the possibility to disable the automatic GPU switching and taking
screenshots of the window content. the first was deemed unnecessary by
me for now, since i just use the libmpv API that others can use anyway.
second is technically not possible atm because we have to pre-allocate
our opengl context at a time the config isn't read yet, so we can't get
the needed property. third one is a bit tricky because of deadlocking
and it needed to be in sync, hopefully i can work around that in the
future.
this also has at least one additional feature or eye-candy. a properly
working fullscreen animation with the native fs. also since this is a
direct port of the old backend of the parts that could be used, though
with adaptions and improvements, this looks a lot cleaner and easier to
understand.
some credit goes to @pigoz for the initial swift build support which
i could improve upon.
Fixes: #5478, #5393, #5152, #5151, #4615, #4476, #3978, #3746, #3739,
#2392, #2217
on macOS mpv was linked to the system SDK which didn't cause any
problems as long as the system SDK was the same as the dev SDK. though
it started to cause linking warnings when a new xcode version with the
SDK for the next macOS was installed. in the worst case it could also
cause linking errors. to fix this we explicitly set the SDK path to the
SDK that is used for building instead.
early flushing only caused problems on macOS, which includes:
- performance problems and huge amount of dropped frames
- problems with playing back video files with fps close to the display
refresh rate
- rendering at twice the rate of the video fps
- not properly detected display refresh rate
we always deactivate any early flush for macOS to fix these problems.
Disable by default.
This feature was added in 7eb342757, which allowed stream selection
in runtime. Problem with this atm is that FFmpeg will try to demux
every first packet of every track leading to noticeable delay opening
the URL.
This option can be changed to enabled by default or removed when
HLS/DASH demuxers are improved upstream.
Remove obsolete comment about FFmpeg ignoring non-http proxies
which was repeated in ytdl_hook before the feature was added.
Remove unnecessary conditions for not nil. Lua tables will always
return nil for non-existent keys.
This commit allows for video to be shown with the right aspect even when
pixels are not square in the selected drm mode. For example, if drm mode
5 is "640x400", the right aspect on a 4:3 monitor is obtained by mpv
--vo=drm --drm-mode=5 --monitorpixelaspect=5:6 ...
Other vo's seem to make this parameter change the size of the window,
but in the drm vo this is fixed, being as large as the screen.
The last image is stored in vo->priv->last_input to be used when
redrawing a frame is necessary (control: VOCTRL_REDRAW_FRAME). At the
beginning it is NULL, so a redraw request has no effect since
draw_image ignores calls with image=NULL.
When using --force-window the size of the image may change without the
vo structure being re-created. Before this commit, the size of
vo->priv->last_input could become inconsistent with the cropping
rectangle vo->priv->src_rc, which could trigger an assert in
mp_image_crop_rc(). Even if it did not, the last image of a video
remained on the screen when the next file in the playlist had no video
(e.g., it was an mp3 without an embedded cover).
This commit deallocates and resets to NULL the image
vo->priv->last_input when reconfiguring video.
Before this commit, the drm vo drew the osd over the scaled image, and
then copied the result onto the framebuffer, shifted. This made the
frame centered, but forced the osd to be only as large as the image.
This was inconsistent with other vo's, covered the image with the
progress indicator even when a black band was at the top of the screen,
made the progress indicator wrap on narrow videos, etc.
The change is to always use an image as large as the screen. The frame
is copied scaled and shifted to it, and the osd drawn over it. The
result is finally copied to the framebuffer without any shift, since it
is already as large as it.
Technically, cur_frame is an image as large as the screen and
cur_frame_cropped is a dummy reference to it, cropped to the size of
the scaled video. This way, copying the scaled image to
cur_frame_cropped positions the image in the right place in cur_frame,
which can then have the osd added to it and copied to the framebuffer.
Using the GL renderer for color conversion will make sure screenshots
will use the same conversion as normal video rendering. It can do this
for all types of screenshots.
The logic when to write 16 bit PNGs changes. To approximate the old
behavior, we decide by looking whether the source video format has more
than 8 bits per component. We apply this logic even for window
screenshots. Also, 16 bit PNGs now always include an unused alpha
channel. The reason is that FFmpeg has RGB48 and RGBA64 formats, but no
RGB064. RGB48 is 3 bytes and usually not supported by GPUs for
rendering, so we have to use RGBA64, which forces an alpha channel.
Will break for users who use --target-trc and similar options.
I considered creating a new gl_video context, but it could double GPU
memory use, so I didn't.
This uses FBOs instead of glGetTexImage(), because that increases the
chance it could work on GLES (e.g. ANGLE). Untested. No support for the
Vulkan and D3D11 backends yet.
Fixes#5498. Also fixes#5240, because the code for reading back is not
used with the new code path.
The re-ordering of commits e3d93fd and 0870859 ended up swallowing the
change which made the HDR tone mapping algorithm actually check for
RA_CAP_NUM_GROUPS support.
The major changes are as follows:
1. Use `uint32_t` instead of `unsigned int` for the SSBO size
calculation. This doesn't really matter, since a too-big buffer will
still work just fine, but since `uint` is a 32-bit integer by
definition this is the correct way to do it.
2. Pre-divide the frame_sum by the num_wg immediately at the end of a
frame. This change was made to prevent overflow. At 4K screen size,
this code is currently already very at risk of overflow, especially
once I started playing with longer averaging sizes. Pre-dividing this
out makes it just about fit into 32-bit even for worst-case PQ
content. (It's technically also faster and easier this way, so I
should have done it to begin with). Rename `frame_sum` to `frame_avg`
to clearly signal the change in semantics.
3. Implement a scene transition detection algorithm. This basically
compares the current frame's average brightness against the
(averaged) value of the past frames. If it exceeds a threshold, which
I experimentally configured, we reset the peak detection SSBO's state
immediately - so that it just contains the current frame. This
prevents annoying "eye adaptation"-like effects on scene transitions.
4. As a result of the previous change, we can now use a much larger
buffer size by default, which results in a more stable and less
flickery result. I experimented with values between 20 and 256 and
settled on the new value of 64. (I also switched to a power-of-2
array size, because I like powers of two)
Before this commit, auto_loaded and lang were only set for the first
track in auto-loaded external files. Likewise, for the title and
lang arguments to the sub-add and audio-add commands.
Fixes#5432
Currently using the drmprime interop with external mpv intgration can lead
to rendering issues because the current frame is being released too early.
Typically using this with Qt results in one frame shift because Qt
will do waitforvsync and swap, rather than swap and waitforvsync.
This leads to tearing as the frambuffer is released while being
displayed on screen.
In order to avoid releasing the framebuffer that is displayed, We keep
the framebuffer alive for one more frame with triple buffering to make
sure that whatever rendering process is used, the framebuffer will not
be released when it's still on screen.
This was tested on RockChip Rock64
Also a regression of the filter change. The new code is more picky about
EOF states, and it turns out the weird delay queue (used with some hwdec
copy back modes only) accidentally dropped an EOF event. It reset the
avctx before the delay queue was drained, which meant it never returned
the expected AVERROR_EOF status code.
Also don't signal EOF when copy back fails. It should just try to
continue until fallback is performed.
The current peak detection algorithm was very bugged (which contributed
to the excessive cross-frame flicker without long normalization) and
also didn't take into account the frame average brightness level.
The new algorithm both takes into account frame average brightness (in
addition to peak brightness), and also computes the values in a more
stable/correct way. (The old path was basically undefined behavior)
In addition to improving the algorithm, we also switch to hable tone
mapping by default, and try to enable peak computation automatically
whever possible (compute shaders + SSBOs supported). We also make the
desaturation milder, after extensive testing during libplacebo
development.
I also had to compensate a bit for the representational differences
between mpv and libplacebo (libplacebo treats 1.0 as the reference peak,
but mpv treats it as the nominal peak), but it shouldn't have caused any
problems.
This is still not quite the same as libplacebo, since libplacebo also
allows tagging the desired scene average brightness on the output, and
it also supports reading the scene average brightness from static
metadata (MaxFALL) where available. But those changes are a bit more
involved. It's possible we could also read this from metadata in the
future, but we have problems communicating with AVFrames as it is and I
don't want to touch the mpv colorimetry structs for the time being.
The vulkan validation layers warn you if you try requesting a query
result from a timer that hasn't even been started yet, so we have to do
some extra bit of work to keep track of which indices we've seen so far,
and avoid the queries on them.
Instead of enabling every feature under the sun, make an effort to just
whitelist the ones we actually might use. Turns out the extended storage
format support is needed for some of the storage formats we use, in
particular rgba16.
Another "what was I thinking" thing - destroying filters explicitly
skipped async wakeups for no reason. These were notifications for
filters that are not going to be destroyed too, and so their wakeup will
be lost, leading to stalled playback. This is completely unnecessary and
the special code can be removed.
Fixes#5488. (This case destroyed all audio filters due to AO init
failure, which could make clear out the f_demux_in.c wakeup for video,
and "freeze" playback.)
This is a dataflow issue caused by the filters change. When the fallback
happens, vd_lavc does not return a frame, but also does not accept a new
packet, which confuses lavc_process(). Fix this by immediately retrying
to feed the buffered packet and decode a frame on fallback.
Fixes#5489.
In theory (and practice), this is not needed, because the VS filter get
frame callback will cause the process function to be called again if
there's not enough data. But it's still a bit weird to just add one more
frame on each iteration, so make it cleaner and make it request frames
until the input array is full.
This was obviously nonsense, and a previous "fix" to this code was
nonsense too. What is really needed here is temporarily dropping the
lock while calling destroy_vs()/reinit_vs().
Fixes#5470.
We called mp_pin_out_request_data() if there was input _and_ output.
This is not how it should be: we should request new input only if output
was requested, but we could not produce any output.
On the other hand, the upper half of the process() function will request
new input if output is required, but all input was consumed. But this
requires calling mp_filter_internal_mark_progress(), as otherwise the
general filter logic would not know that we can continue.
If the speed is changed by a large amount, we need to effectively change
the output rate by a large amount, and swr_set_compensation() is
apparently not designed to handle such large changes well. So it's
better to reinitialize the resampler on all large changes.
Also, strictly reinitialize the resampler if the rate changes, otherwise
it could happen that libavresample (which does not automatically
initialize resampling if avresample_set_compensation() is used) would
never apply speed changes properly.
Also document some conditions better that handle corner cases (remove
the inline condition from the if gating the compensation code).
It also appears that we crashed with very large compensation ratios
(when raising audio speed quickly by keeping the "[" key down), and this
commit accidentally mitigates it by not allowing large compensation.
Setting lavfi-complex at runtime will now forcefully reselect the tracks
as needed, even if it was a "proper" track selection via --aid or --vid.
Before this commit, it just failed and complained that the VO/AO was
already "used".
Requested.
This makes it actually somewhat simpler, and doesn't have any
disadvantages. It should also make some new features easier.
Mostly just moves code around.
The somewhat confusing thing is that many filters (including track->dec)
have a public struct, but to free them, you need to free the mp_filter
pointer itself (track->dec->f). The assignment wrote to a dangling
pointer, instead of removing the dangling pointer.
(Other than that, this idiom is actually nice.)
Unknown frames were not freed properly. Although this doesn't really
happen anyway, because we're never going to feed audio frames to a video
filter chain. Since it's theoretically possible, and all other filters
handle this consistently, fix it anyway.
Properly initialize the output frame parameters other than image format
and size. This includes colorspace hints. (We're still not reading them
back from VapourSynth if it sets them, though. Usually it doesn't
anyway.)
VapourSynth can't pass through timestamps, only frame durations. So we
need to remember the timestamp of the very first frame passed to it.
This was accidentally set to 0 instead of NOPTS on init, so inserting
the filter during playback could show strange behavior.
Might be part of #5470.
Similar to the previous commit, and for the same reasons. Unlike with
af_scaletempo, resampling does not have a natural frame size, so we set
an arbitrary size limit on output frames. We add a new option to control
this size, although I'm not sure whether anyone will use it, so mark it
for testing only.
Note that we go through some effort to avoid buffering data in
libswresample itself. One reason is that we might have to reinitialize
the resampler completely when changing speed, which drops the buffered
data. Another is that I'm not sure whether the resampler will do the
right thing when applying dynamic speed changes.
This helps the filter to adapt much faster to speed changes. Before this
commit, the filter just converted and output the full input frame, which
could cause problems with large input frames. This was made worse by
certain filters like dynaudnorm or loudnorm outputting pretty large
frames.
This commit changes the filter from trying to convert all input at once
to only outputting a single internally filtered frame. Internally, this
filter already output data in units of 60ms by default (controlled by
the "stride" sub-option), and concatenated as many output frames as
necessary to consume all input.
Behavior is still kind of bad when inserting the filter. This is because
the large frames can be buffered up after the insertion point, so the
speed change will be performed with a larger latency. The scaletempo
filter can't do anything against this, although it can be fixed by
inserting scaletempo as user filter as part of --af.
--vf=help will now list libavfilter filters, and e.g. --vf=yadif=help
will list libavfilter filter options.
The latter is rather bare, because the AVOption API is really awful
(holy shit how is it so bad), and would require us to handle _every_
option type manually.
Alternatively we could call av_opt_show2(), which ffmpeg uses for help
output in its CLI tools and which is much more detailed. But it's rather
foreign and forces output through av_log(), so I don't really want to
use it.
MPlayer used this to distinguish multiple decoder wrappers (such as
libavcodec vs. binary codec loader vs. builtin decoders). It lost
meaning in mpv as non-libavcodec things were dropped. Now it doesn't
serve any purpose anymore.
Parsing was removed quite a while ago, and the recent filter change
removed any use of the internal family field. Get rid of it.
This reverts commit b7f90be567.
The author agreed to the relicensing now (if that code is affected by
the original copyright at all - that was the only line possibly left of
it).
If tags like TITLE have the whole parameter in " quotes, strip them.
Also remove the leading whitespace, since even with a single space it
was always included.
Fixes#5462.
FFmpeg only suppports http proxies and ignores it if
the resulting url is https. Also, no SOCKS.
Use it like `--ytdl-raw-options=proxy=[http://127.0.0.1:3128]` so
it doesn't confuse mpv because of the colons.
You need to pass it as an option because youtube-dl doesn't give
us the proxy.
Or just set `http_proxy` environment variable as recommended before.
Added example using -append, which doesn't need escaping.
This is supposed to undo the ] binding. This uses a value closer to the
inverse. (Although it's not fully exact since the values are still
stored as floating point instead as fractions.)
Use the decoder wrapper that was introduced for video. This removes all
code duplication the old audio decoder wrapper had with the video code.
(The audio wrapper was copy pasted from the video one over a decade ago,
and has been kept in sync ever since by the power of copy&paste. Since
the original copy&paste was possibly done by someone who did not answer
to the LGPL relicensing, this should also remove all doubts about
whether any of this code is left, since we now completely remove any
code that could possibly have been based on it.)
There is some complication with spdif handling, and a minor behavior
change (it will restrict the list of codecs to spdif if spdif is to be
used), but there should not be any difference in practice.
Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes
a source filter. vd.h mostly disappears, because mp_filter takes care of
the dataflow, but its remains are in struct mp_decoder_fns.
One goal is to simplify dataflow by letting the filter framework handle
it (or more accurately, using its conventions). One result is that the
decode calls disappear from video.c, because we simply connect the
decoder wrapper and the filter chain with mp_pin_connect().
Another goal is to eventually remove the code duplication between the
audio and video paths for this. This commit prepares for this by trying
to make f_decoder_wrapper.c extensible, so it can be used for audio as
well later.
Decoder framedropping changes a bit. It doesn't seem to be worse than
before, and it's an obscure feature, so I'm content with its new state.
Some special code that was apparently meant to avoid dropping too many
frames in a row is removed, though.
I'm not sure how the source code tree should be organized. For one,
video/decode/vd_lavc.c is the only file in its directory, which is a bit
annoying.
This is supposed to help making data flow easier and wakeup handling
more efficient. Once that change is done, reading a packet on any
stream won't have to wakeup and poll all decoders (which helps reducing
the mess even if all decoders are on the same thread).
This also improves the accuracy of wakeups by tracking better whether
a wakeup is needed.
This is preparation for a change in vd_lavc.c: it should not have to
access the demuxer (to pass along closed captions), so the idea is to
make them part of mp_image, and to let the layer above vd_lavc propagate
the buffer.
Don't bother with preserving them for mp_image->AVFrame, because we
don't need this.
Reduce the trivial but still annoying code duplication in
mp_image_new_ref(), which has to create new buffer references and deal
with possible failure of creating them. The tricky part is that if
creating a reference fails, we must set the target to NULL, so that
unreferencing the failed new mp_image reference does not release the
buffer references of the original mp_image. For the same reason, the
code can't jump to error handling when it can't create a new reference,
and has to set a flag instead.
Get rid of the old vf.c code. Replace it with a generic filtering
framework, which can potentially handle more than just --vf. At least
reimplementing --af with this code is planned.
This changes some --vf semantics (including runtime behavior and the
"vf" command). The most important ones are listed in interface-changes.
vf_convert.c is renamed to f_swscale.c. It is now an internal filter
that can not be inserted by the user manually.
f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed
once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is
conceptually easy, but a big mess due to the data flow changes).
The existing filters are all changed heavily. The data flow of the new
filter framework is different. Especially EOF handling changes - EOF is
now a "frame" rather than a state, and must be passed through exactly
once.
Another major thing is that all filters must support dynamic format
changes. The filter reconfig() function goes away. (This sounds complex,
but since all filters need to handle EOF draining anyway, they can use
the same code, and it removes the mess with reconfig() having to predict
the output format, which completely breaks with libavfilter anyway.)
In addition, there is no automatic format negotiation or conversion.
libavfilter's primitive and insufficient API simply doesn't allow us to
do this in a reasonable way. Instead, filters can use f_autoconvert as
sub-filter, and tell it which formats they support. This filter will in
turn add actual conversion filters, such as f_swscale, to perform
necessary format changes.
vf_vapoursynth.c uses the same basic principle of operation as before,
but with worryingly different details in data flow. Still appears to
work.
The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are
heavily changed. Fortunately, they all used refqueue.c, which is for
sharing the data flow logic (especially for managing future/past
surfaces and such). It turns out it can be used to factor out most of
the data flow. Some of these filters accepted software input. Instead of
having ad-hoc upload code in each filter, surface upload is now
delegated to f_autoconvert, which can use f_hwupload to perform this.
Exporting VO capabilities is still a big mess (mp_stream_info stuff).
The D3D11 code drops the redundant image formats, and all code uses the
hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a
big mess for now.
f_async_queue is unused.
The RA_CAP_FRAGCOORD checks apply to dumb mode as well, but they were
after the check for dumb mode, which returns early, so they never ran.
Fixes#5436
Using vdpau will allocate additional textures for the reinterleaving
step, which uninit_rendering() will free. This is a problem because the
hwdec image remains mapped when reinitializing, so the reinterleaving
textures are turned into dangling pointers. Fix this by freeing the
reinterleave textures on full uninit instead.
Fixes#5447.
AV_DISPOSITION_ATTACHED_PIC usually means the video track isn't real,
and merely reflects the presence of an embedded image in tag data (such
as ID3v2 tags), with some inconsistent hack to make libavformat return
it as video packet once.
Except it doesn't mean that. It can be randomly set on other streams
that do sort of behave like video streams, such as chapter thumbnail
tracks in mp4 files. AV_DISPOSITION_TIMED_THUMBNAILS is set in these
cases. In theory, there can supposedly be more such cases, but only the
chapter thumbnail one currently exists. So add it as exception.
This restores displaying these thumbnails as video frames, for better or
worse. (Before, only the first thumbnail was displayed.)
Requires newest FFmpeg git, which has a change that makes the HLS
demuxer set an AVFMTCTX_UNSEEKABLE flag if seeking is not available,
which is the case for HLS live streams. This should make the player
frontend behave pretty well, instead of crapping up irrecoverably.
Restores behaviour prior to aef2ed5dc13e37dec0670c451b4369b151d5c65f.
That change was apparently unpopular. However, given the amount of
complaining over how hard it is to change the defaults by rebinding every
key, I think the extra option introduced by this commit is justified.
Technically not all behaviour is restored, because now --no-osd-bar will
not instead display the msg text on seek. I think that feature was a
little weird and is now easy enough to remedy with the --osd-on-seek
option.
This reverts commit 9812e276aa1bb0bddeb73677aa9e9f87e73cd930.
This was apparently unpopular. I still think the pause OSD should be the
same as seek even if it's not visible by default, but it seems that
whether to display a given property change is currently conflated with
what to display.
The reverted behaviour can be restored by adding something like the
following to input.conf:
SPACE cycle pause; show_progress
This makes all the video/audio variants available for selection.
Might break with non-hls/dash, or even with dash if FFmpeg wasn't
compiled with the demuxer.
The use of the FFmpeg hls protocol (as opposed to demuxer) is
"discouraged", and probably only causes additional potential security
problems at best, so drop it.
Instead of duplicating the append code, reimplement it using the
existing code. The difference between -add and -append is that -append
does not take multiple items (thus removing the need for escaping), but
-append can reuse all code for -add by pretending the separator is never
found.
Requested. See manpage additions.
The main reason why this goes through the trouble to keep the
action/operation parameter separate is so that we don't expose some
option parser implementation details to the command (although that is a
relatively weak reason), and also to make it more different from the
"set" command, which can't support this type of option as it goes
through the property layer.
Fixes#5435.
And use it for 2 demuxer options. It could be used for more options
later. (Though the --cache options can not use this, because they use KB
as base unit.)
Add the print callback to all option types (except pseudo option types
which don't represent values). This makes it less confusing for client
API users (no strange properties that can't be read), and also lists the
default properly with --list-options.
Fix the option type for audio formats - they use int, not uint32_t.
Fix some identation cosmetic issues.
Always make the hw params dump function use MSGL_DEBUG, and remove the
MSGL_V use. That means you need -v -v to see them. The detailed
information is usually not very interesting, so this reduces the log
noise.
The af_get_best_sample_formats() function had an argument of
int[AF_FORMAT_COUNT], which is slightly incorrect, because it's 0
terminated and should in theory have AF_FORMAT_COUNT+1 entries. It won't
actually write this many formats (since some formats are fundamentally
incompatible), but it still feels annoying and incorrect. So fix it, and
require that callers pass an AF_FORMAT_COUNT+1 array.
Note that the array size has no meaning in C function arguments (just
another issue with C static arrays being weird and stupid), so get rid
of it completely.
Not changing the af_lavcac3enc use, since that is rewritten in another
branch anyway.
When overlay-add etc. is run, make sure the playlop is rerun so that it
considers actually redrawing the screen.
(I considered making the OSD code generally wakeup the player, but that
will probably lead to redundant wakeups, so I didn't bother.)
Fixes#5431.
Not very clean since there's a lot of potential unsafe urls that youtube-dl
can give us, depending on whether it's a single url, split tracks,
playlists, segmented dash, etc.
I found that at least for mjpeg streams, FFmpeg will set packet pts/dts
anyway. The mjpeg raw video demuxer (along with some other raw formats)
has a "framerate" demuxer option which defaults to 25, so all mjpeg
streams will be played at 25 FPS by default.
mpv doesn't like this much. If AVFMT_NOTIMESTAMPS is set, it prints a
warning, that might print a bogus FPS value for the assumed framerate.
The code was originally written with the assumption that FFmpeg would
not set pts/dts for such formats, but since it does, the printed
estimated framerate will never be used. --fps will also not be used by
default in this situation.
To make this hopefully less confusing, explicitly state the situation
when the AVFMT_NOTIMESTAMPS flag is set, and give instructions how to
work it around.
Also, remove the warning in dec_video.c. We don't know what FPS it's
going to assume anyway. If there are really no timestamps in the stream,
it will trigger our normal missing pts workaround. Add the assumed FPS
there.
In theory, we could just clear packet timestamps if AVFMT_NOTIMESTAMPS
is set, and make up our own timestamps. That is non-trivial for advanced
video codecs like h264, so I'm not going there. For seeking and
buffering estimation the situation thus remains half-broken.
This is a mitigation for #5419.
this adds the standard menu bar items Services, Hide Others, Show All
and Close, as suggested by Apple's HIG.
https://developer.apple.com/macos/human-interface-guidelines/menus/menu-bar-menus/#app-menu
- Services are useful to add custom actions and shortcuts via the
System Preferences to mpv
- Close is important since the menu bar can open secondary windows, like
About or the Open dialogue. those couldn't be closed with the standard
system shortcut before. this is now possible.
NSFileHandlingPanelOKButton is deprecated with macOS 10.13, but the
replacement NSModalResponseOK is not available on 10.8 and earlier.
added a declaration for 10.8 and earlier since i only officially
dropped support for 10.7 and earlier. this is untested.
It was actually already implemented as ta_dup_ptrtype(), but that seems
like a clunky name. Also we still use the talloc_ names throughout the
source, and I'd rather use an old name instead of a mixing inconsistent
naming conventions.
If you play a video with an external audio track, and do backwards
keyframe seeks, then audio can be missing. This is because a backwards
seek can end up way before the seek target (this is just how this seek
mode works). The audio file will be seeked at the correct seek target
(since audio usually has a much higher seek granularity), which results
in silence being played until the video reaches the originally intended
seek target.
There was a hack in audio.c to deal with this. Replace it with a
different hack. The new hack probably works about as well as the old
hack, except it doesn't add weird crap to the audio resync path (which
is some of the worst code here, so this is some nice preparation for
rewriting it). As a more practical advantage, it doesn't discard the
audio demuxer packet cache. The old code did, which probably ruined
seeking in youtube DASH streams.
A non-hacky solution would be handling external files in the demuxer
layer. Then chaining the seeks would be pretty easy. But we're pretty
far from that, because it would either require intrusive changes to the
demuxer layer, or wouldn't be flexible enough to load/unload external
files at runtime. Maybe later.
mp_sws_set_from_cmdline() has the only purpose to respect the --sws-
command line options. Instead of forcing callers to get the option
struct containing these, let callers pass mpv_global, and get it from
the option core code directly. This avoids minor annoyances later on.
FFmpeg has its own rather "special" image pools (AVHWFramesContext)
specifically for hardware decoding. So it's not really practical to use
our own pool implementation. Add these helpers, which make it easier to
use FFmpeg's code in mpv.
This fixes that AVFrames passing through libavfilter (such as with
--lavfi-complex) implicitly stripped some fields. I'm not actually sure
what to do with the mp_image_params.color.light field here (what happens
if the colorspace changed?) - there is no equivalent in AVFrame or
FFmpeg at all.
It did not affect the old --vf code, because it doesn't allow
libavfilter to change the metadata.
Also log the .light field in verbose mode.
There is some craziness here: the function allocates m_config_cache,
which in turn allocates the actual option struct, which is what the
function returns. The user would expect to be able to use talloc_free()
to deallocate everything. Of course this didn't work, because the
returned pointer is not the root parent in the talloc tree.
But with some additional talloc craziness, this can be fixed. We
rearrange the parent pointers such that freeing the option struct will
free m_config_cache first, which uninits the contents in the option
struct, but fortunately not the option struct itself.
This change should simplify API use on the caller side, and reduce
surprises.
mp_read_option_raw() should not print the deprecation warning if the
option is deprecated. This change also means you can't pass an alias
to the function, but all existing uses should be fine.
This commit eliminates the following clang warning:
warning: macro expansion producing 'defined' has undefined behavior [-Wexpansion-to-defined]
Going by the clang commit message, this seems to be explicitly specified
as UB by the standard, and they added this warning because MSVC
apparently results in different behavior. Whatever, we can just avoid
the warning with some small changes.
ipc-unix.c does this out of convenience. Since signals are global
process state, this deserves a mention, since applications could in
theory rely on SIGPIPE being set to something else.
DR (direct rendering) works by having the decoder decode into the GPU
staging buffers, instead of copying the video data on texture upload. We
did this even for formats unsupported by the GPU or the renderer. This
"worked" because the staging memory is untyped, and the video frame was
converted by libswscale to a supported format, and then uploaded with a
copy using the normal non-DR texture upload path.
Even though it "works", we don't gain anything from using the staging
buffers for decoding, since we can't use them for upload anyway. Also,
staging memory might be potentially limited (what really happens is up
to the driver). It's easy to avoid, so just skip it in these cases.
The check_gl_features(p) call here checks whether dumb mode can be used.
It uses the field use_integer_conversion, which is set _after_ the call
in the same function. Move check_gl_features() to the end of the
function, when use_integer_conversion is finally set.
Fixes that it tried to use bilinear filtering with integer textures. The
bug disabled the code that is supposed to convert it to non-integer
textures.
This segfaults otherwise. The conditional is needed to break a circular
dependency (gl_init depends on mpgl_load_functions which depends on
recreate_dispmanx which calls gl_ctx_resize).
Fixes#5398
This commit introduces the multiply-pitch af-command. Users may bind
keys to this command in order to incrementally adjust the pitch of a
track. This will probably mostly be useful for musicians trying to
transpose up and down by semi tones without having to calculate
the correct ratio beforehand.
As an example, here is an input.conf to test this feature:
{ af-command all multiply-pitch 0.9438743126816935
} af-command all multiply-pitch 1.059463094352953
This is achieved by adding the new function get_term_status_msg that
returns the status message specified by the user, or the mpv default.
Previously, term_osd_print_status_lazy would exit early as soon as a
user status message had been set, which potentially skipped adding the
term_osd_bar if the user also requested that.
fixes#3280
The future direction might be not having such a user-visible filter at
all, similar to how vf_scale went away (or actually, redirects to
libavfilter's vf_scale).
This is part of trying to get rid of --af-defaults, and the af
resample filter.
It requires a complicated mechanism to set the defaults on the resample
filter for backwards compatibility.
Remove the max_count creation parameter, because it's pointless and
rarely ever did anything. Add a talloc parent parameter instead (which
is something completely different, but convenient, and all callers needs
to be changed anyway).
Instead of clearing the pool when the now removed maximum is reached,
clear it on image parameter changes instead.
With the recent changes to the script it does not incur a startup delay
by default due to starting youtube-dl and waiting for it. This was the
main reason for making libmpv have a different default.
Starting sub processes from a library can still be a bit fishy, but I
think it's ok. Still mention it in the libmpv header. There were already
other cases where libmpv would start its own processes, such as the X11
backend calling xdg-screensaver. (The reason why this is fishy is
because UNIX process management sucks: SIGCHLD and the wait() syscall
make sub processes non-transparent and could potentially introduce
conflicts with code trying to use them.)
package.config is available in 5.1, 5.2, 5.3 and luajit, so should be fine.
The first character is the path separator, so it's '\' on windows and '/'
on *nix.
This should also prevent cases where users download the wrong binary.
Similar to 1eec7d2315, but for the beginning of the stream (named BOF in
this commit).
We can know this only if demuxing actually started from the beginning.
If there is a seek to the beginning (even if you use --start=-1000), we
don't know in general whether the demuxer truly returns the start of the
file. We could probably make a heuristic with assuming that this is what
happens if the seek target is before the start time or so, but this is
not included in this commit.
libavformat's cover art hack (aka attached pictures) breaks the ability
of the demuxer cache to keep multiple seek ranges. This happens because
the cover art packet has neither position nor timestamp, and libavformat
gives us the packet even though we intended to drop it.
The cover art hack works by adding the cover art packet to the read
packet stream once when demuxing starts (or after seeks). mpv treats
cover art in a similar way internally, but we have to compensate for
libavformat's shortcomings, and add the cover art packet ourselves when
we need it. So we don't want libavformat to return the packet.
We normally prevent this in demux_lavc.c/select_tracks() and explicitly
disable cover art streams. (We add it in dequeue_packet() instead.) But
libavformat will actually add the cover art packet even if we disable
the cover art stream, because it adds it at initialization time, and
does not bother to check again in av_read_frame() (apparently). The
packet is actually read, and upsets the demuxer cache logic. In
addition, this also means we probably decoded the cover art picture
twice in some situations.
Fix this by explicitly checking/discarding this in yet another place.
(Screw this hack...)
The impact was that you couldn't exactly seek to the join point with a
keyframe seek, even though there was a keyframe. This commit fixes it by
preserving the necessary metadata that got lost on cached range joining.
This is so absurdly obscure that it gets a longer code comment.
Replace the static array with dynamic memory allocation. This also
requires some code to honor mp_cmd.nargs more strictly. Generally
allocates more stuff.
Fixes#5375 (although we could also just raise the static limit).
If feed_packet() ended with DATA_WAIT, the player should have gone to
sleep, until the demuxer wakes it up again when there is new data. But
the call to read_frame() unconditionally overwrote this status code, so
it never waited. The consequence was that the core burned CPU by
effectively polling the demuxer status, which was noticeable especially
when seeking in network streams (since seeking is async, decoders will
start out with having to wait for network).
Regression since commit 33e5755c.
Previously, toggling pause would generate no osd response, and changing
that wasn't even configurable. This was surprising to users who
generally expect to see *where* pause / unpause is taking place (#3028).
The previous default was osd-bar (unless the user specified
--no-osd-bar, in which case case it was osd-msg). Aside from requiring
some twisted logic to implement, this surprised users since osd-msg3
wasn't displayed when seeking with the keyboard (#3028), so the time
seeked to was never displayed.
No idea why this wasn't done earlier. This makes playback start in audio
only tracks closer to video-only or video/audio restart. It has the
consequence that --cache-pause-initial now works for audio-only streams
too.
This warning was printed when the demuxer cache tried to join two
adjacent seek ranges, but failed if the last keyframe in the second
range was within the (overlapping) first range. This is a weird corner
case which to support probably would not be worth it.
So this code just printed a warning and discarded the second range. As
it turns out, this can happen relatively often if you seek a lot, and
the seek ranges are very tiny (such as consisting of only 1 keyframe).
Dropping the second range in these cases is OK and probably cheaper than
trying to actually join them. Change the warning to verbose level.
(It seems this could actually be "supported", because if keyframe_latest
is not set, there will be no other keyframes, so it could just be unset,
with the exception that q1->keyframe_latest in the code below must not
be overwritten. But still, too much trouble for a special case that
likely does not matter, and it would have to be tested too.)
For quite some time, msg.c hasn't output partial log messages anymore,
and instead buffered them in memory. This means the MP_INFO() statement
here just kept appending the message to memory, instead of outputting
it.
Easy enough to fix by abusing the status line (which means the frontend
and this code will "fight" for the status line, but this code seems to
win usually, as the frontend doesn't update it so often).
Users should probably really switch to --cache-pause-initial.
Fixes#5360.
The underlying logic is still the same (basically pausing if the demuxer
cache underruns), but clean up the higher level logic a bit. It goes
from 3 levels of nested if statements to 1.
Also remove the code duplication for the --cache-pause-initial logic.
In addition, make sure an earlier buffering state has no influence on
the new state after a seek (this is also why some of the state resetting
can be removed from loadfile.c).
Initialize cache_buffer always to 100. It basically means we start out
assuming all buffers are filled enough. This actually matters for
verbose messages only, but removes some weird special casing.
Before this commit, some autoselection of tracks coming from files
loaded with --external-files was still done. This commit removes all of
it, and the only way to select a track is via the explicit stream
selection options like --vid/--sid/--aid.
I think this was always the original intention. The change could in
theory still unintentionally surprise some users, so add a changelog
entry.
This does not affect --audio-file/--sub-file, even if these contain
mismatching track types. E.g. if audio files passed to --audio-file
contain subtitles, these should still be selected. Past feature requests
indicate that users want this.
This means if the user tries to seek past EOF, and we know EOF was seen
already, then use a cached seek, instead of triggering a low level seek.
This requires some annoying tracking, but seems pretty simple otherwise.
One advantage of doing this is that if the user tries to do this kind of
seek, there's no unnecessary waiting for a reaction by network (and in
most cases, redundant downloading of data just to discard it again).
Another is that this avoids creating overlapping seek ranges: previously, the
low level seek would naturally create a new range. Then it would read and add
data from the end of the stream due to the low level demuxer not being able to
seek to the target and selecting the last seek point before the end of the
stream. Consequently, this new range would overlap with the previous cached
range. But since the cache joining code is written such that you join the
current range with the _next_ range (instead of the previous as it would be
needed in this case), the overlapping ranges were left alone, until seeking back
to the previous range. That was ugly, sort of harmless, and could happen in
other cases, but this avoidable case was pretty easy to trigger.
Export them as explicitly undocumented debugging fields for the
"demuxer-cache-state" property.
Should be somewhat helpful to debug "wtf is the demuxer" doing
situations better, especially when seeking. It also becomes visible how
long the demuxer is blocked on an "old" seek when you keep seeking while
the first seek hasn't finished.
update_seek_ranges() has some special code that attempts to correctly
adjust seek ranges for subtitle tracks. (Subtitle are a nightmare for
seek ranges, because they are sparse, so using the packet list is not
enough to reliably determine the valid cached range.)
This had code like this inside the modified if statement:
range->seek_start = MP_PTS_MAX(range->seek_start, <something>);
If seek_start is NOPTS, then seek_start will be set to <something>,
breaking some other code that checks seek_start for NOPTS to see if it's
empty. Fix this by explicitly checking whether seek_start is NOPTS
before adjusting it.
The crash happened in prune_old_packets() because the range was marked
as non-empty, yet there was no packet in it to prune. This was with
files with muxed subtitles, when seeking back to the start. This should
not happen anymore with the change. Also add an assert() to
check_queue_consistency() that checks for this specific case.
There's still some mess. In theory, subtitle tracks could be completely
empty, yet their seek range would span the entire file. Seek range
tracking of subtitle files is slightly broken (even before this change).
Some of this should probably be revisited later, including not just
using seek_start to determine whether a seek range should be pruned due
to being empty.
This enables DXVA2 hardware decoding with ra_d3d11. It should be useful
for Windows 7, where D3D11VA is not available. Images are transfered
from D3D9 to D3D11 using D3D9Ex surface sharing[1].
Following Microsoft's recommendations, it uses a queue of shared
surfaces, similar to Microsoft's ISurfaceQueue. This will hopefully
prevent surface sharing from impacting parallelism and allow multiple
D3D11 frames to be in-flight at once.
[1]: https://msdn.microsoft.com/en-us/library/windows/desktop/ee913554.aspx
Remove our own hacky reconnection code, and use libavformat's feature for
that. It's disabled by default, and until recently it did not work too
well. This has been fixed in recent ffmpeg git master[1], so there's no reason
to keep our own code.
[1] FFmpeg/FFmpeg@8a108bdea0
We set "reconnect_delay_max" to 7, which limits the maximum time it
waits. Since libavformat doubles the wait time on each reconnect attempt
(starting with 1), and stops trying to reconnect once the wait time is
over the reconnect_delay_max value, this allows for 4 reconnection
attempts which should add to 11 seconds maximum wait time. The default
is 120, which seems too high for normal playback use.
(The user can still override these parameters with --stream-lavf-o.)
In a lost device scenario, resize() will fail and p->backbuffer will be
NULL. We can't recover from lost devices yet, but we should still check
for a NULL backbuffer in start_frame() rather than crashing.
Also remove a NULL check for p->swapchain. This was a red herring, since
p->swapchain never becomes NULL in an error condition, but p->backbuffer
actually does.
This should fix the crash in #5320, but it doesn't fix the underlying
reason for the lost device (which is probably a driver bug.)
Previously, mpv would attempt to use a BGRA swapchain in the hope that
it would give better performance, since the Windows desktop is also
composited in BGRA. In practice, it seems like there is no noticable
performance difference between RGBA and BGRA swapchains and BGRA
swapchains cause trouble with a42b8b1142, which attempts to use the
swapchain format for intermediate FBOs, even though D3D11 does not
guarantee BGRA surfaces will work with UAV typed stores.
Reasons why you'd want this see manpage additions. Disabled by default,
because it would increase latency of live streams by default. (Or well,
at least it would be another problem when trying getting lower latency.)
This tried to be clever by waiting for a longer time each time the
buffer was underrunning, or shorter if it was getting better. I think
this was pretty weird behavior and makes no sense. If the user really
wants the stream to buffer longer, he/she/it can just pause the player
(the network caches will continue to be filled until they're full).
Every time I actually noticed this code triggering in my own use, I
didn't find it helpful. Apart from that it was pretty hard to test.
Some waiting is needed to avoid that the player just plays the available
data as fast as possible (to compensate for late frames and underrunning
audio). Just use a fixed wait time, which can now be controlled by the
new --cache-pause-wait option.
Don't drop the stream buffers, because the read call (that must have
been failing) might try to extend an existing read buffer in the first
place. Just move the messy seek logic to stream_lavf.c. (In theory,
stream_lavf should probably make libavformat connect at the correct
offset instead of using a seek to reconnect it again. This patch doesn't
fix it, but at least it's a good argument to have the messing with the
position not in the generic code.)
Also update the comment about avio not supporting reconnecting. It has
that feature now. Maybe we should use it, but only after it gets fixed.
Remove them from the big MPOpts struct and move them to their sub
structs. In the places where their fields are used, create a private
copy of the structs, instead of accessing the semi-deprecated global
option struct instance (mpv_global.opts) directly.
This actually makes accessing these options finally thread-safe. They
weren't even if they should have for years. (Including some potential
for undefined behavior when e.g. the OSD font was changed at runtime.)
This is mostly transparent. All options get moved around, but most users
of the options just need to access a different struct (changing sd.opts
to a different type changes a lot of uses, for example).
One thing which has to be considered and could cause potential
regressions is that the new option copies must be explicitly updated.
sub_update_opts() takes care of this for example.
Another thing is that writing to the option structs manually won't work,
because the changes won't be propagated to other copies. Apparently the
only affected case is the implementation of the sub-step command, which
tries to change sub_delay. Handle this one explicitly (osd_changed()
doesn't need to be called anymore, because changing the option triggers
UPDATE_OSD, and updates the OSD as a consequence). The way the option
value is propagated is rather hacky, but for now this will do.
It was split at least across osd.c and sd_ass.c/sd_lavc.c. sd_lavc.c
actually ignored most of the more obscure subtitle timing things.
There's no reason for this - just move it all to dec_sub.c (mostly from
sd_ass.c, because it has some of the most complex stuff).
Now timestamps are transformed as they enter or leave dec_sub.c.
There appear to have been some subtle mismatches about how subtitle
timestamps were transformed, e.g. sd_functions.accepts_packet didn't
apply the subtitle speed to the timestamp. This patch should fix them,
although it's not clear if they caused actual misbehavior.
The semantics of SD_CTRL_SUB_STEP are slightly changed, which is the
reason for the changes in command.c and sd_lavc.c.
Was only available with --demuxer-lavf-format=help and the demuxer
needed to be used for it to actually print the list.
This can be used in the future to check if 'dash' support was compiled
with FFmpeg so ytdl_hook can use it instead. For now, dashdec is too
rudimentary to be used right away.
The old code tried to make sure at all times to try to read a new
packet. Only once that was read, it tried to retrieve new video or audio
frames the decoder might already have decoded.
Change this to strictly read frames from the decoder until it signals
that it wants a new packet, and only then read and feed a new packet.
This is in theory nicer, follows the libavcodec recommended data flow,
and and reduces the minimum latency by 1 frame.
This merely requires switching the order in which those calls are done.
Normally, the decoder will return only 1 frame until a new packet is
required. If we would just feed it 1 packet, return DATA_AGAIN, and wait
until the next frame is decoded, we would run the playloop 1 time too
often for no reason (which is fine but might have some overhead). To
avoid this, try to read a frame again after possibly feeding a packet.
For this reason, move the feed/read code to its own functions each,
instead of merely moving the code.
The audio and video code for this particular thing is basically
duplicated. The idea is to unify them one day, so make the change to
both. (Doing this for video is the real motivation for this change, see
below.)
The video code change is slightly more complicated, because we have to
care about the framedrop counting (which is just a heuristic, but for
now considered better than nothing, and possibly considered required to
warn the user of framedrops happening - maybe).
Apparently this change helps with stalling streams on Android with the
mediacodec wrapper and mpeg2 decoder implementations which deinterlace on
decoding (and return 2 frames per packet).
Based on an idea and observations by tmm1.
Uses the EGL width/height by default when the user fails to set
the android-surface-width/android-surface-height options.
This means the vo-resize command is optional, and does not need to
be implemented on android devices which do not support rotation.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Now macosx_menubar.m and mpv.rc (win32) use the same copyright string.
(This is a bit roundabout, because mpv.rc can't use C constants. Also
the C code wants to avoid rebuilding real source files if only version.h
changed, so only version.c includes version.h.)
This seems to fix issues when building on windows where compiling mpv.rc
after a `waf clean` resulted in a failure because version.h was not
always present
Apparently some Intel drivers have a bug where copying from staging
buffers to constant buffers does not work. We used to keep a copy of the
buffer data in a staging buffer to enable partial constant buffer
updates. To work around this bug, keep the copy in talloc-allocated
system memory instead.
There doesn't seem to be any noticable performance difference from
keeping the copy in system memory. Our cbuffers are probably too small
for it to matter anyway.
See also: https://crbug.com/593024Fixes#5293
This reverts commit 9513165c99
and commit 4efe330efb.
I had changed --loop-file to interact with --start to work
the same way that --loop-playlist does. (That is, --loop-file
seeks to the --start time upon looping, not the beginning of
the file.) However, the consensus is that the old behavior is
preferred and the interaction with --loop-playlist is the one
that is incorrect.
In addition, this change introduced a bug in the interaction
between Quit-Watch-Later and --loop-file, where upon reaching
playback end it would seek to the resume timestamp, not the
start of the file.
As a result, this commit reverts that change.
The x264 hack requires reading the first video packet, which in turn we
handle with a hack in demux_mkv.c to get the packet without having to
add special crap to demux.c. Another useless MKV feature (which they
enabled by default at one point and which caused many demuxers to break
completely, only to disable it again when it was too late) conflicts
with this, because we actually pass a block as packet contents, instead
of after "decompression".
Fix this by calling demux_mkv_decode().
This fixes when resuming certain broken h264 files encoded by x264. See
FFmpeg commit 840b41b2a643fc8f0617c0370125a19c02c6b586 about the x264
bug itself.
Normally, the unregistered user data SEI (that contains the x264 version
string) is informational only. But libavcodec uses it to workaround a
x264 bug, which was recently fixed in both libavcodec and x264. The fact
that both encoder and decoder were buggy is the reason that it was not
found earlier, and there are apparently a lot of files around created by
the broken decoder. If libavcodec sees the SEI, this bug can be worked
around by using the old behavior.
If you resume a file with mpv (i.e. seeking when the file loads),
libavcodec never sees the first video packet. Consequently it has to
assume the file is not broken, and never applies the workaround,
resulting in garbage being played.
Fix this by always feeding the first video packet to the decoder on
init, and then flushing the codec (to avoid that an unwanted image is
output). Flushing the codec does not remove info such as the x264
version. We also abuse the fact that the first avcodec_send_packet()
always pushes the frame into the decoder (so we don't have to trigger
the decoder by requsting an output frame).
Technically, the user could just use --vd-lavc-o with the same result.
But I find it better to make this an explicit option, so we can document
the ups and downs, and also avoid setting it for non-h264.
This means that we now explicitly set an interval of 1. Although that
should be the EGL default, some drivers could possibly ignore this
(unconfirmed). In any case, this commit also allows disabling vsync, for
users who want it.
Crashed when no vdpau device was loaded. Also there was a mistake of not
setting p->ctx, which broke software surface input mode. This was not
found before, because p->ctx is not needed for anything else.
Fixes#5294.
Show total cache as well as demuxer cache separately.
This adjusts the presented values to be consistent with status line
and OSC modifications made in https://github.com/mpv-player/mpv/pull/5250
This commit introduces a new --oset-metadata key-value-list option,
allowing the user to specify output metadata when encoding
(eg. --oset-metadata=title="Hello",comment="World").
A second option --oremove-metadata is added to exclude existing metadata
from the output file (assuming --ocopy-metadata is enabled).
Not all output formats support all tags, but luckily libavcodec
simply discards unsupported keys.
--copy-metadata describes the result of the option better, (copying metadata
from the source file to the output file). Marks the old --no-ometadata
OPT_REMOVED with a suggestion for the new --no-ocopy-metadata.
A release has been made, so drop options deprecated for that release.
Also drop some options which have been deprecated a much longer time
before.
Also fix a typo in client-api-changes.rst.
The queue family index and the queue info index are not necessarily the
same, so we're forced to do a check based on the queue family index
itself.
Fixes#5049
A vulkan validation layer update pointed out that this was wrong; we
still need to use the access type corresponding to the stage mask, even
if it means our code won't be able to skip the pipeline barrier (which
would be wrong anyway).
In additiona to this, we're also not allowed to specify any source
access mask when transitioning from top_of_pipe, which doesn't make any
sense anyway.
Async compute in particular seems to cause problems on some drivers, and
even when supprted the benefits are not that massive from the tests I
have seen, so it's probably safe to keep off by default.
Async transfer on the other hand seems to work better and offers a more
substantial improvement, so it's kept on.
This gets confused by e.g. SPARSE_BIT on the TRANSFER_BIT, leading to
situations where "more specialized" is ambiguous and the logic breaks
down. So to fix it, only compare the subset we care about.
blit() implies scaling, copy() is the equivalent command to use when the
formats are compatible (same pixel size) and the rects have the same
dimensions.
This allows RAs with support for non-opaque FBO formats to use a more
appropriate FBO format for the output tex, possibly enabling a more
efficient blit operation.
This requires distinguishing between real formats (which can be used to
create textures) and fake formats (e.g. ra_gl's FBO hack).
On AMD devices, we only get one graphics pipe but several compute pipes
which can (in theory) run independently. As such, we should prefer
compute shaders over fragment shaders in scenarios where we expect them
to be better for parallelism.
This is amusingly trivial to do, and actually improves performance even
in a single-queue scenario.
Instead of using a single primary queue, we generate multiple
vk_cmdpools and pick the right one dynamically based on the intent.
This has a number of immediate benefits:
1. We can use async texture uploads
2. We can use the DMA engine for buffer updates
3. We can benefit from async compute on AMD GPUs
Unfortunately, the major downside is that due to the lack of QF
ownership tracking, we need to use CONCURRENT sharing for all resources
(buffers *and* images!). In theory, we could try figuring out a way to
get rid of the concurrent sharing for buffers (which is only needed for
compute shader UBOs), but even so, the concurrent sharing mode doesn't
really seem to have a significant impact over here (nvidia). It's
possible that other platforms may disagree.
Our deadlock-avoidance strategy is stupidly simple: Just flush the
command every time we need to switch queues, and make sure all
submission and callbacks happen in FIFO order. This required lifting the
cmds_pending and cmds_queued out from vk_cmdpool to mpvk_ctx, and some
functions died/got moved as a result, but that's a relatively minor
change.
On my hardware this is a fairly significant performance boost, mainly
due to async transfers. (Nvidia doesn't expose separate compute queues
anyway). On AMD, this should be a performance boost as well due to async
compute.
This is especially interesting for vulkan since it allows completely
skipping the layout transition as part of the renderpass. Unfortunately,
that also means it needs to be put into renderpass_params, as opposed to
renderpass_run_params (unlike #4777).
Closes#4777.
This uses the new vk_signal mechanism to order all access to textures.
This has several advantageS:
1. It allows real synchronization of image access across multiple frames
when using multiple queues for parallelism.
2. It allows using events instead of pipeline barriers, which is a
finer-grained synchronization primitive that allows for more
efficient layout transitions over longer durations.
This commit also restructures some of the implicit transition code for
renderpasses to be more flexible and correct. (Note: this technically
drops the ability to transition the image out of undefined layout when
not blending, but that was a bug anyway and needs to be done properly)
vo_gpu: vulkan: remove no-longer-true optimization
The change to the output_tex format makes this no longer true, and it
actually seems to hurt performance now as well. So just don't do it
anymore. I also realized it hurts performance when drawing an OSD, so
it's probably not a good idea anyway.
This combines VkSemaphores and VkEvents into a common umbrella
abstraction which can resolve to either.
We aggressively try to prefer VkEvents over VkSemaphores whenever the
conditions are met (1. we can unsignal the semaphore, i.e. it comes from
the same frame; and 2. it comes from the same queue).
Instead of being submitted immediately, commands are appended into an
internal submission queue, and the actual submission is done once per
frame (at the same time as queue cycling). Again, the benefits are not
immediately obvious because nothing benefits from this yet, but it will
make more sense for an upcoming vk_signal mechanism.
This also cleans up the way the ra_vk submission interacts with the
synchronization/callbacks from the ra_vk_ctx. Although currently, the
way the dependency is signalled is a bit hacky: normally it would be
associated with the ra_tex itself and waited on in the appropriate stage
implicitly. But that code is just temporary, so I'm keeping it in there
for a better commit order.
Instead of associating a single VkSemaphore with every command buffer
and allowing the user to ad-hoc wait on it during submission, make the
raw semaphores-to-signal array work like the raw semaphores-to-wait-on
array. Doesn't really provide a clear benefit yet, but it's required for
upcoming modifications.
1. No more static arrays (deps / callbacks / queues / cmds)
2. Allows safely recording multiple commands at the same time
3. Uses resources optimally by never over-allocating commands
- if [ "$TRAVIS_COMPILER" = "clang" ]; then export CXX="clang++"; fi
- if [ "$TRAVIS_COMPILER" = "gcc" ]; then export CXX="g++"; fi
- if [ "$TRAVIS_OS_NAME" = "linux" ]; then docker pull $CONTAINER; fi
- if [ "$TRAVIS_OS_NAME" = "osx" ]; then "$TRAVIS_BUILD_DIR/ci/get_ffmpeg.sh"; fi
script:
- ./bootstrap.py
- ./waf configure
- ./waf build
- if [ "$TRAVIS_OS_NAME" = "linux" ]; then docker run --env CC --env TARGET -v $TRAVIS_BUILD_DIR:/build $CONTAINER /bin/sh -c "cd /build && $CI_SCRIPT"; fi
- if [ "$TRAVIS_OS_NAME" = "osx" ]; then ./ci/build-macos.sh; fi
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