The code that determines the process exit code ignores all stop_play
values other than PT_QUIT. Generally, PT_ERROR is meaningless outside of
play_current_file(), and is mostly equivalent to PT_NEXT_ENTRY.
Do something that makes it report a non-0 exit code, and indicates in
the terminal exit message that something went wrong.
Untested.
Really minor detail that doesn't really matter. If frame stepping pauses
playback on end (why does this special case even exist), it should
probably be done after on_unload, because all works is supposed to be
finished at that point.
When the current file changes (or rather, when starting/finishing
playback of a playlist entry), clients tend to have the problem that
it's hard to tell whether a property change notification (via
mpv_observe_property() and mechanisms layered on top of it) is from the
previous or new playlist entry. The previous commit probably helps, but
all the asynchronity is still a bit unhelpful.
Try to make this better by adding new hooks, that are run before/after
playback init/deinit. This is similar to the existing hooks, except
they're outside of "initialized" playback, which excludes that you might
accidentally get an overlap between the current and the previous/next
playlist entry.
That still doesn't seem quite enough, since normally, property change
notifications come after the hook event. So basically a client would
have to explicitly "drain" the event queue within the hook, and make the
hook continue only after that is done. Knowing when property
notifications are done is another asynchronous nightmare (how exactly it
works keeps changing within client.c, and an API user probably can't
tell anymore when all pending properties are truly done). So introduce
another guarantee: properties that were changed before the hook happens
will be returned before the hook event is returned. That means the
client will have received all pending property notifications from the
previous playlist entry (or whatever) before the hook is entered.
As another minor complication, we shouldn't just keep the hook pending
until _all_ property notifications are done, since the client's hook
could produce new ones. (Or just consider things like the demuxer thread
hammering the client with cache update events, while the "on_preloaded"
hook is run.) So there is some extra untested, fragile logic in client.c
to handle this (the waiting_for_hook flag).
This probably works, but was barely tested. Not sure if this helps
anyone, but I think it's fine for my own purposes. (I really hated this
aspect of the API whenever I used it myself.)
Some filters may block the playloop for a longer time. For example, if a
decoder fails to decode anything and somehow just discards packets, the
filter graph would run (in a blocking manner) until all packets are
read, which could take a longer time if the demuxer thread is fast
enough.
Make it exit every 100ms. That should at least give the user a chance to
stop playback.
Filtering could run on a different thread, but I don't see much value in
doing that in the general case. It would just waste a thread. Although
being able to use mp_filter_graph_interrupt() would be slightly nicer
than such a timeout mechanism. Decoding in particular can actually use a
separate thread (--vd-queue-enable), but again, this is not enabled by
default, because it just wastes a thread.
Like the previous f_decoder_wrapper commit, this is probably a sin.
This was a hack that attempted to line up external audio tracks with
video. The problem is that if you do a keyframe seek backwards, video
will usually seek much farther back than audio (due to much higher
keyframe aka seek point distances). The hack somehow made seeking a 2
step process.
This existed in 4 different forms in the history of this code base, and
it was always very cumbersome. We mostly needed this for ytdl_hook (I
think?), which uses the 4th form, which is nicely confined to
demux_timeline and is unrelated to the "external" audio tracks in the
high level player.
Since this is (probably) not really widely needed anymore, get rid of
it. Better do this now, than when somehow rewriting all the seeking code
(which might happen in this decade or the next or so) and when it
wouldn't be easily revertable anymore in case we find we "really" need
it unlike expected.
There is no issue if hr-seeks are used. Also, you can still use edl
files to "bundle" multiple streams as if it was a single stream (this is
what ytdl_hook does now).
Hr-seek past the last frame instantly enters EOF, which means
handle_playback_time() will not set playback_pts to the video PTS (as
all video frames are skipped), which leads to the playback time being
taken from the last seek target. This results in confusing behavior,
especially since the seek time will be clipped to the file duration for
display, but not for further relative seeks.
Obviously, the time should be set to the last video frame, so use the
last video frame as fallback if both audio and video have ended. Also,
since the same problem exists with audio-only playback, add a fallback
for audio PTS too. We don't know which was the "last" fragment of media
played (to decide whether to use the audio or video PTS as the
fallback), but it doesn't matter since the maximum works.
This could lead to some undesired effects. In particular the audio PTS
is basically a bad guess, and is for example not clipped against --end.
(But the ridiculous way audio syncing and clamping currently works, I'm
not going to touch that shit unless I rewrite it completely.) The cover
art case is slightly broken: using --keep-open with keyframe seeks will
result in 0 as playback PTS (the video PTS). OK, who cares, it got late.
Also casually get rid of last_vo_pts, since that barely made any sense
at all.
Fixes: #7487
Although a linked list was ideal at first, there are cases where it
sucks, and became increasingly awkward (with the mpv command API
preferring integer indexes to access the list). In future, we probably
want to add more playlist-related functionality, so better change it to
an array now.
An array isn't always ideal either. Since playlist entries are still
separate objects (because in some cases you need a stable "iterator" to
it), but you still need to efficiently get the next/previous playlist
entry, there's a pl_index field, that needs to be maintained. E.g.
adding an entry at the start of the playlist => update the pl_index
field for all other entries. Well, it's not really worth to do something
more complicated to avoid these things.
This commit is probably buggy as shit. It's not like I bothered to test
everything. That's _your_ role.
mpv has a very weak and very annoying policy that determines whether a
playlist should be used or not. For example, if you play a remote
playlist, you usually don't want it to be able to read local filesystem
entries. (Although for a media player the impact is small I guess.)
It's weak and annoying as in that it does not prevent certain cases
which could be interpreted as bad in some cases, such as allowing
playlists on the local filesystem to reference remote URLs. It probably
barely makes sense, but we just want to exclude some other "definitely
not a good idea" things, all while playlists generally just work, so
whatever.
The policy is:
- from the command line anything is played
- local playlists can reference anything except "unsafe" streams
("unsafe" means special stream inputs like libavfilter graphs)
- remote playlists can reference only remote URLs
- things like "memory://" and archives are "transparent" to this
This commit does... something. It replaces the weird stream flags with a
slightly clearer "origin" value, which is now consequently passed down
and used everywhere. It fixes some deviations from the described policy.
I wanted to force archives to reference only content within them, but
this would probably have been more complicated (or required different
abstractions), and I'm too lazy to figure it out, so archives are now
"transparent" (playlists within archives behave the same outside).
There may be a lot of bugs in this.
This is unfortunately a very noisy commit because:
- every stream open call now needs to pass the origin
- so does every demuxer open call (=> params param. gets mandatory)
- most stream were changed to provide the "origin" value
- the origin value needed to be passed along in a lot of places
- I was too lazy to split the commit
Fixes: #7274
Both the "stop" and "loadfile" commands are asynchronous. "stop" starts
terminating playback, which used to be done in the same playloop
iteration, but now it may take longer, so a "loadfile" command can be
received while this is going on. (Possible that it used to work if the
second command was issued at least in the next playloop iteration.)
Make the "loadfile" override the "stop" mode, so the next file will be
played, instead of quitting or going into idle mode.
Unlike the referenced bug report claims, this has nothing to do with
IPC.
Fixes: #7225
Instead of making m_config a special-case, it more or less uses the
underlying m_config_cache/m_config_shadow APIs properly. This makes the
player core a (relatively) equivalent user of the core option API. In
particular, this means that other threads can change core options with
m_config_cache_write_opt() calls (before this commit, this merely led to
diverging option values).
An important change is that before this commit, mpctx->opts contained
the "master copy" of all option data. Now it's just another copy of the
option data, and the shadow copy is considered the master. This is why
whenever mpctx->opts is written, the change needs to be copied to the
master (thus why this commits add a bunch of m_config_notify... calls).
If another thread (e.g. a VO) changes an option, async_change_cb is now
invoked, which funnels the change notification through the player's
layers.
The new self_notification parameter on mp_option_change_callback is so
that m_config_notify... doesn't trigger recursion, and it's used in
cases where the change was already "processed". It's still needed to
trigger libmpv property updates. (I considered using an extra
m_config_cache for that, but it'd only cause problems with no
advantages.)
I think the recent changes actually forgot to send libmpv property
updates in some cases. This should fix this anyway. In some cases,
property updates are reworked, and the potential for bugs should be
lower (probably).
The primary point of this change is to allow external updates, for
example by a VO writing the fullscreen option if the window state is
changed by the window manager (rather than mpv changing it). This is not
used yet, but the following commits will.
As preparation for making repl.lua part of the core (maybe), add some
mechanisms which are supposed to improve its behavior.
Add a silent mode. Calling mpv_request_log_messages() with the log level
name prefixed with "silent:" will disable logging from the API user's
perspective. But it will keep the log buffer, and record new messages,
without returning them to the user. If logging is enabled again by
requesting the same log level without "silent:" prefix, the buffered log
messages are returned to the user at once. This is not documented,
because it's far too messy and special as that I'd want anyone to rely
on this behavior, but it will be perfectly fine for an internal script.
Another thing is that we record early startup messages. The goal is to
make the repl.lua script show option and config parsing file errors.
This works only with the special "terminal-default" log level.
In addition, reduce the "terminal-default" capacity to only 100 log
messages. If this is going to be enabled by default, it shouldn't use
too much resources.
Used to contain flags for "save" setting of options at runtime. Now
there is nothing special needed anymore and it's 0. So drop it
completely, and remove anything that distinguishes between runtime and
initialization time.
Since a track may not be selected twice, it makes sense e.g. for
secondary subtitles to select the next best match and avoid the
duplicate selection.
This allows for example `--slang=en,ja --secondary-sid=auto` to select
'en' as primary and 'ja' as secondary without needing to know the actual
sid for 'ja'.
Looks like this didn't actually work. Prefetching will do nothing if
there isn't a thread to "drive" it, and the demuxer thread needs to be
explicitly enabled. (I guess I did the worst possible job in verifying
whether this actually worked when I implemented it. On the other hand,
the user didn't confirm back whether it worked, so who cares.)
Like in the previous commit, bad factoring makes everything worse. It
duplicates logic and implementation of enable_demux_thread(), since the
opener thread cannot access the mpctx->opts field freely. But it's deep
night, so fuck it.
Fixes: c1f1a0845eFixes: #6753
demux_start_prefetch() was called unconditionally in two cases. This is
completely wrong. I'm not sure what part of my brain died off that
something this obviously wrong went in.
The prefetch case is a bit more complicated. It's a different thread, so
you can't access just access mpctx->opts there. So add an explicit field
for this, which is the simplest way to get this done. (Even if it's bad
factoring.)
Fixes: c1f1a0845e
Fixes: 556e204a11
Although this was sort of elegant, it just seems to complicate things
slightly. Originally, the API meant that you cache mp_recorder_sink
yourself (which would avoid the mess of passing an index around), but
that too seems slightly roundabout.
In a later change, I want to change the set of streams passed to
mp_recorder_create(), and then I'd have to keep track of the index for
each stream, which would suck. With this commit, I can just pass the
unambiguous sh_stream to it, and it will be guaranteed to match the
correct stream.
The disadvantages are barely worth discussing. It's a new linear search
per packet, but usually only 2 to 4 streams are active at a time. Also,
in theory a user could want to write 2 streams using the same sh_stream
(same metadata, just writing different packets or so), but in practice
this is never done.
With the stream cache gone, this function had almost no use anymore
(other than opening the stream). Improve this by triggering demuxer
cache readahead.
This enables all streams. At this point we can't know yet what streams
the user's options would select (at least not without great additional
effort). Generally this is what you want, and the stream cache would
have read the same amount of data.
In addition, this will work much better for files that e.g. need to seek
to the end when opening (typically mp4, and mkv files produced by newer
mkvmerge versions).
Remove the deselection call from add_stream_track(). This should be
fine, as streams normally start out as deselected anyway. In the
prefetch case, some code in play_current_file() actually deselects it.
Streams that appear during demuxing are disabled by default, so this
doesn't break this logic either.
Fixes: #6753
That's right, and it's probably not the end of it. I'll just claim that
I have no idea how to create a proper user interface for this, so I'm
creating multiple partially-orthogonal, of which some may work better in
each of its special use cases.
Until now, there was --record-file. You get relatively good control
about what is muxed, and it can use the cache. But it sucks that it's
bound to playback. If you pause while it's set, muxing stops. If you
seek while it's set, the output will be sort-of trashed, and that's by
design.
Then --stream-record was added. This is a bit better (especially for
live streams), but you can't really control well when muxing stops or
ends. In particular, it can't use the cache (it just dumps whatever the
underlying demuxer returns).
Today, the idea is that the user should just be able to select a time
range to dump to a file, and it should not affected by the user seeking
around in the cache. In addition, the stream may still be running, so
there's some need to continue dumping, even if it's redundant to
--stream-record.
One notable thing is that it uses the async command shit. Not sure
whether this is a good idea. Maybe not, but whatever. Also, a user can
always use the "async" prefix to pretend it doesn't.
Much of this was barely tested (especially the reinterleaving crap),
let's just hope it mostly works. I'm sure you can tolerate the one or
other crash?
The old implementation didn't work for the OGG case. Discard the old
shit code (instead of fixing it), and write new shit code. The old code
was already over a year old, so it's about time to rewrite it for no
reason anyway.
While it's true that the old code appears to be broken, the main reason
to rewrite this is to make it simpler. While the amount of code seems to
be about the same, both the concept and the actual tag handling are
simpler. The result is probably a bit more correct.
The packet struct shrinks by 8 byte. That fact that it wasted 8 bytes
per packet for a rather obscure use case was the reason I started this
at all (and when I found that OGG updates didn't work). While these 8
bytes aren't going to hurt, the packet struct was getting too bloated.
If you buffer a lot of data, these extra fields will add up. Still quite
some effort for 8 bytes. Fortunately, it's not like there are any
managers that need to be convinced whether it's worth doing. The freedom
to waste time on dumb shit.
The old implementation attached the current metadata to each packet.
When the decoder read the packet, the packet's metadata was made
current. The new implementation stores metadata as separate list, and
requires that the player frontend tells it the current playback time,
which will be used to find the currently valid metadata. In both cases,
the objective was to correctly update metadata even if a lot of data is
buffered ahead (and to update them correctly when seeking within the
demuxer cache).
The new implementation is actually slightly more correct, because it
uses the playback time for the metadata lookup. Consider if you have an
audio filter which buffers 15 seconds (unfortunately such a filter
exists), then the old code would update the current title 15 seconds too
early, while the new one does it correctly.
The new code also simplifies mixing the 3 metadata sources (global, per
stream, ICY). We assume these aren't mixed in a meaningful way. The old
code tried to be a bit more "exact". I didn't bother to look how the old
code did this, but the new code simply always "merges" with the previous
metadata, so if a newer tag removes a field, it's going to stick around
anyway.
I tried to keep it simple. Other approaches include making metadata a
special sh_stream with metadata packets. This would have been
conceptually clean, but the implementation would probably have been
unnatural (and doesn't match well with libavformat's API anyway). It
would have been nice to make the metadata updates chapter points (makes
a lot of sense for the intended use case, web radio current song
information), but I don't think it would have been a good idea to make
chapters suddenly so dynamic. (Still an idea to keep in mind; the new
code actually makes it easier to work towards this.)
You could mention how subtitles are timed metadata, and actually are
implemented as sparse packet streams in some formats. mp4 implements
chapters as special subtitle stream, AFAIK. (Ironically, this is very
not-ideal for files. It would be useful for streaming like web radio,
but mp4 is extremely bad for streaming by design for other reasons.)
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
It seems the so called demuxer cache wasn't really disabled for
sub-demuxers (timeline stuff). This was relatively harmless, since the
actual packet data was shared anyway via refcounting. But with the
addition of a mmap cache backend, this may change a lot.
So strictly disable any caching for sub-demuxers. This assumes that
users of sub-demuxers (only demux_timeline.c by now?) strictly use
demux_read_any_packet(), since demux_read_packet_async() will require
some minor read-ahead if a low level packet read returned a packet for a
different stream.
This requires some awkward messing with this fucking heap of trash. The
thing that is really wrong here is that the demuxer API mixes different
concepts, and sub-demuxers get the same API as decoders, and use the
cache code.
Obviously should seek back to the end of the file when it loops.
Also remove some minor code duplication around start times. This isn't
the correct solution by the way. Rather than hoping we know a reasonable
start/end time, this stuff should instruct the demuxer to seek to the
exact location. It'll work with 99% of all normal files, but add an
appropriate comment (that basically says the function is bullshit) to
get_start_time() anyway.
This changes the behavior of the --ab-loop-a/b options. In addition, it
makes it work with backward playback mode.
The most obvious change is that the both the A and B point need to be
set now before any looping happens. Unlike before, unset points don't
implicitly use the start or end of the file. I think the old behavior
was a feature that was explicitly added/wanted. Well, it's gone now.
This is because of 2 reasons:
1. I never liked this feature, and it always got in my way (as user).
2. It's inherently annoying with backward playback mode.
In backward playback mode, the user wants to set A/B in the wrong order.
The ab-loop command will first set A, then B, so if you use this command
during backward playback, A will be set to a higher timestamps than B.
If you switch back to forward playback mode, the loop would stop
working. I want the loop to just continue to work, and the chosen
solution conflicts with the removed feature.
The order issue above _could_ be fixed by also switching the AB-loop
user option values around on direction switch. But there are no other
instances of option changes magically affecting other options, and doing
this would probably lead to unexpected misery (dying from corner cases
and such).
Another solution is sorting the A/B points by timestamps after copying
them from the user options. Then A/B options set in backward mode will
work in forward mode. This is the chosen solution. If you sort the
points, you don't know anymore whether the unset point is supposed to
signify the end or the start of the file.
The AB-loop code is slightly better abstracted now, so it should be easy
to restore the removed feature. It would still require coming up with a
solution for backwards playback, though.
A minor change is that if one point is set and the other is unset, I'm
rendering both the chapter markers and the marker for the set point.
Why? I don't know. My test file had chapters, and I guess I decided this
looked better.
This commit also fixes some subtle and obvious issues that I already
forgot about when I wrote this commit message. It cleans up some minor
code duplication and nonsense too.
Regarding backward playback, the code uses an unsanitary mix of internal
("transformed") and user timestamps. So the play_dir variable appears
more than usual.
To mention one unfixed issue: if you set an AB-loop that is completely
past the end of the file, it will get stuck in an infinite seeking loop
once playback reaches the end of the file. Fixing this reliably seemed
annoying, so the fix is "just don't do this". It's not a hard freeze
anyway.
The get_play_start_pts() function was supposed to return "rebased"
(relative to 0) timestamps. This was roundabout, because one of 2
callers just added the offset back, and the other caller actually
expected an absolute timestamp.
Change rel_time_to_abs() (whose return value get_play_start_pts()
returns without further changes) to return absolute times.
This should fix that absolute and relative times passed to --start and
--end were treated the same, which can't be right. It probably also
fixes --end if --rebase-start-time=no is used (which can't have been
correct either).
All in all I'm not sure why --rebase-start-time=no or absolute vs.
relative times in --start/--end even exist, when they were incorrectly
implemented for years.
Untested, because no sample file and I don't care. However, if anyone
cares, and I got it wrong, I hope it's simple to fix.
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.
This is just redundant and slightly annoying, at least for normal
command line usage. If there are multiple entries, still print it
(because you want to know where you are). Also still print it if the
player was redirected (because you want to know where you got redirected
to).
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.
Apparently this was so that when playing a video file from a .rar file,
it would load external subtitles with the same name (instead of looking
for mpv's rar:// mangled URL). This was requested on github almost 5
years ago. Seems like a weird feature, and I don't care. Drop it,
because it complicates some in progress change.
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.
As stated in the original commit message, if the demuxer set the start
time to the first subtitle packet, the subtitles would be shifted
incorrectly. It appears that it is the case for external PGS subtitles.
This reverts commit 520fc74036.
Fixes#5485
--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.
mpctx->current_track[0][STREAM_VIDEO] (and STREAM_AUDIO) are empty when
using --lavfi-complex. Moving the muxer stream hinting after audio/video chain
initialization and checking if the chains exist fixes encoding with --lavfi-complex.
Previously, the output audio/video streams did not get prepared and the encode
would fail due to unexpected stream addition.
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.
The player fully restarts playback when the edition or disk title is
changed. Before this, the player tried to reinitialized playback
partially. For example, it did not print a new "Playing: <file>"
message, and did not send playback end to libmpv users (scripts or
applications).
This playback restart code was a bit messy and could have unforeseen
interactions with various state. There have been bugs before. Since it's
a mostly cosmetic thing for an obscure feature, just change it to a full
restart. This works well, though since it may have consequences for
scripts or client API users, mention it in interface-changes.rst.
This was always a legacy thing. Remove it by applying an orgy of
mp_get_config_group() calls, and sometimes m_config_cache_alloc() or
mp_read_option_raw().
win32 changes untested.
Until now, stopping playback aborted the demuxer and I/O layer violently
by signaling mp_cancel (bound to libavformat's AVIOInterruptCB
mechanism). Change it to try closing them gracefully.
The main purpose is to silence those libavformat errors that happen when
you request termination. Most of libavformat barely cares about the
termination mechanism (AVIOInterruptCB), and essentially it's like the
network connection is abruptly severed, or file I/O suddenly returns I/O
errors. There were issues with dumb TLS warnings, parsers complaining
about incomplete data, and some special protocols that require server
communication to gracefully disconnect.
We still want to abort it forcefully if it refuses to terminate on its
own, so a timeout is required. Users can set the timeout to 0, which
should give them the old behavior.
This also removes the old mechanism that treats certain commands (like
"quit") specially, and tries to terminate the demuxers even if the core
is currently frozen. This is for situations where the core synchronized
to the demuxer or stream layer while network is unresponsive. This in
turn can only happen due to the "program" or "cache-size" properties in
the current code (see one of the previous commits). Also, the old
mechanism doesn't fit particularly well with the new one. We wouldn't
want to abort playback immediately on a "quit" command - the new code is
all about giving it a chance to end it gracefully. We'd need some sort
of watchdog thread or something equally complicated to handle this. So
just remove it.
The change in osd.c is to prevent that it clears the status line while
waiting for termination. The normal status line code doesn't output
anything useful at this point, and the code path taken clears it, both
of which is an annoying behavior change, so just let it show the old
one.
Before this, mpctx->playing was often used to determine whether certain
new state could be added to the playback state. In particular this
affected external files (which added tracks and demuxers). The variable
was checked to prevent that they were added before the corresponding
uninit code. We want to make a small part of uninit asynchronous, but
mpctx->playing needs to stay in the place where it is. It can't be used
for this purpose anymore.
Use mpctx->stop_play instead. Make it never have the value 0 outside of
loading/playback. On unloading, it obviously has to be non-0.
Change some other code in playloop.c to use this, because it seems
slightly more correct. But mostly this is preparation for the following
commit.
Alway give each demuxer its own mp_cancel instance. This makes
management of the mp_cancel things much easier. Also, instead of having
add/remove functions for mp_cancel slaves, replace them with a simpler
to use set_parent function. Remove cancel_and_free_demuxer(), which had
mpctx as parameter only to check an assumption. With this commit,
demuxers have their own mp_cancel, so add demux_cancel_and_free() which
makes use of it.
Them being separate is just dumb. Replace them with a single
demux_free() function, and free its stream by default. Not freeing the
stream is only needed in 1 special case (demux_disc.c), use a special
flag to not free the stream in this case.
The player fully restarts playback when the edition or disk title is
changed. Before this, the player tried to reinitialized playback
partially. For example, it did not print a new "Playing: <file>"
message, and did not send playback end to libmpv users (scripts or
applications).
This playback restart code was a bit messy and could have unforeseen
interactions with various state. There have been bugs before. Since it's
a mostly cosmetic thing for an obscure feature, just change it to a full
restart. This works well, though since it may have consequences for
scripts or client API users, mention it in interface-changes.rst.
