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.
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.
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.
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 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.
--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.
I don't ever use them, so kill them.
Linux TV is excessively complex, and whenever I attempted to use it, it
didn't work well or would have required some major work to update it.
(For example, when I tried to use a webcam-type device with tv://, it
worked badly; even the libavdevice garbage worked better.)
The "program" property was rather complex and rather obscure. I didn't
ever use it. Should there ever be a proper use for it (maybe HLS stream
selection?), it should be rewritten anyway.
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.
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 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.
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.
