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.
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.
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.
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.)
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.
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
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.
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 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.
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.
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.
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.
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.
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.
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.
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
--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.
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.
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.
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.
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.
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.
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.
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 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.
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.
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.
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".)
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.
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.
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.
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.
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.
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.)
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
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.
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.
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.
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.
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.
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 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.
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.)
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
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.
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.
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.
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.
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.
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.
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.
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.
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 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
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.
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.
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
Usable for uniquely identifying mpv instances from
subprocesses, controlling mpv with AppleScript, ...
Adds a new mp_getpid() wrapper for cross-platform reasons.
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 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.
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
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.
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 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.
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.
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.
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.
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.
Restores behaviour prior to aef2ed5dc1.
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 9812e276aa.
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
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.)
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
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.
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.)
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.
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 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
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.
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.
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>
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 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.
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.
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 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.
I don't want to add another field to display stream and demuxer cache
separately, so just add them up. This strangely makes sense, since the
forward buffered stream cache amount consists of data not read by the
demuxer yet. (If the demuxer cache has buffered the full stream, the
forward buffered stream cache amount is 0.)
Reduce it from 75MB in both directions (forward/backwards) to 10MB each.
The stream cache is kind of becoming useless in favor of the demuxer
cache. Using both doesn't make much sense, because they will contain
duplicated data for no reason.
Still leave it at 10MB, which may help with mp4 a bit. libavformat's mp4
demuxer tends to seek too much, so we try to avoid triggering network
level seeks by having some caching in the stream layer.
Some old crap which nobody needs and which probably nobody uses.
This relies on a GCC extension: using "## __VA_ARGS__" to remove the
comma from the argument list if the va args are empty. It's supported
by clang, and there's some chance newer standards will introduce a
proper way to do this. (Even if it breaks somewhere, it will be a
problem only for 1 release, since I want to drop the deprecated
properties immediately.)
This was off for mpv CLI, but on for libmpv. The motivation behind this
was that it would be confusing for applications if libmpv continued
playback in a severely "degraded" way (without either audio or video),
and that it would be better to fail early.
In reality the behavior was just a confusing difference to mpv CLI, and
has confused actual users as well. Get rid of it.
Not bothering with a version bump, since this is so minor, and it's easy
to ensure compatibility in affected applications by just setting the
option explicitly.
(Also adding the missing next-release-marker in client-api-changes.rst.)
Annoying exception that makes no sense to keep. Normally, users or
client applications will either use --hwdec=auto, or not set the option
at all, which both leads to the expected result.
This commit introduces mp.utils.file_info() for querying information
on file paths, implemented for both Lua and Javascript.
The function takes a file path as an argument and returns a Lua table /
JS object upon success. The table/object will contain the values:
mode, size, atime, mtime, ctime and the convenience booleans is_file, is_dir.
On error, the Lua side will return `nil, error` and the Javascript side
will return `undefined` (and mark the last error).
This feature utilizes the already existing cross-platform `mp_stat()`
function.
This was a bit confused, and I bet nobody understood whether to use
--sub-file or --sub-files, and what the difference is. Explicitly
mention that both variants exist, and how they are related.
Previously when using a libmpv instance to play multiple videos,
once --start was set there was no clear way to unset it. You could
use --start=0, but 0 does not always mean the beginning of the file
(especially when using --rebase-start-time=no). Looking up the start
timestamp and passing that in also does not always work, particularly
when the first timestamp is negative (since negative values to --start
have a special meaning).
This commit adds a new "none" value which maps to the internal
REL_TIME_NONE, matching the default value of the play_start option.
