Until now, they could be aborted only by ending playback, and calling
mpv_abort_async_command didn't do anything.
This requires furthering the mess how playback abort is done. The main
reason why mp_cancel exists at all is to avoid that a "frozen" demuxer
(blocked on network I/O or whatever) cannot freeze the core. The core
should always get its way. Previously, there was a single mp_cancel
handle, that could be signaled, and all demuxers would unfreeze. With
external files, we might want to abort loading of a certain external
file, which automatically means they need a separate mp_cancel. So give
every demuxer its own mp_cancel, and "slave" it to whatever parent
mp_cancel handles aborting.
Since the mpv demuxer API conflates creating the demuxer and reading the
file headers, mp_cancel strictly need to be created before the demuxer
is created (or we couldn't abort loading). Although we give every
demuxer its own mp_cancel (as "enforced" by cancel_and_free_demuxer),
it's still rather messy to create/destroy it along with the demuxer.
This is nonsense. Didn't matter in most situations, because seeking
itself set this after it was cleared. But some callers don't do this,
see e.g. commit ed73ba8964. There is no need to clear it at all, and
it causes issues with the next commit. It only needs to be reset on
loading.
Also move the initialization on loading up, which doesn't change
behavior, but makes the intention clearer.
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.
Many asynchronous commands are potentially long running operations, such
as loading something from network or running a foreign process.
Obviously it shouldn't just be possible for them to freeze the player if
they don't terminate as expected. Also, there will be situations where
you want to explicitly stop some of those operations explicitly. So add
an infrastructure for this.
Commands have to support this explicitly. The next commit uses this to
actually add support to a command.
If a struct as large as MPContext contains a field named "lock", it
creates the impression that it is the primary lock for MPContext. This
is wrong, the lock just protects a single field.
Basically, the ytdl_hook script will not terminate the script, even if
you change to a new playlist entry. This happens because ytdl_hook keeps
the player core in an early loading stage, and the forceful playback
abort is done only in the ermination code.
This does not handle the "stop" and "quit" commands, which can still
take longer than expected, but on the other hand have some weird special
handling (see below). I'm not doing this out of laziness. Playback
stopping will have to be somewhat redone anyway. Basically we want to
give everything a chance to terminate, and if it doesn't work, we want
to stop loading or playback forcefully after a small timeout. We also
want to remove the mess with input.c's special handling of "quit" and
some other commands (see abort_playback_cb stuff).
It seems the ytdl script like to continue loading external tracks even
if loading was aborted. Trying to do so will still quickly fail, but not
without a load of log noise. So check and error out early.
Pretty trivial, since commands can be async now, and the common code
even provides convenience like running commands on a worker thread.
The only ugly thing is that mp_add_external_file() needs an extra flag
for locking. This is because there's still some code which calls this
synchronously from the main thread, and unlocking the core makes no
sense there.
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.
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).
There was a "generic" function to run a hook and to wait for its
completion, yet there were two duplicated functions doing the same
anyway. Replace them with a single function.
They differed in how stop_play was handled, but it was broken anyway.
stop_play is set when playback is stopped due to quitting or changing
the playlist entry - but we still can't stop hook processing, because
that would mean asynchronously doing something else while the user hook
code is still busy and might still have the expectation that running the
hook stops everything else. So not waiting until the hook ends properly
is against the whole hook idea. That this was done inconsistently is
even worse. (Though it could be argued that when quitting the player,
everything should just be stopped violently. But I still think that's
up to the hook handler.)
process_hooks() does not return anything, since hook processing doesn't
really have a result (it's all about blocking and letting some other
code synchronously do something). Just let the caller check whether
loading was aborted in the meantime.
Also change the potentially misleading name of mp_hook_run().
As it turns out, there are multiple libmpv users who saw a need to
use the hook API. The API is kind of shitty and was never meant to be
actually public (it was mostly a hack for the ytdl script).
Introduce a proper API and deprecate the old one. The old one will
probably continue to work for a few releases, but will be removed
eventually.
There are some slight changes to the old API, but if a user followed
the manual properly, it won't break.
Mostly untested. Appears to work with ytdl_hook.
After switching, the playback state was not reset, which could leave it
in a strange, pause like state, that could be fixed by e.g. seeking.
This seems to be an older regression - it's even in 0.27.
Sometimes, playback needs to be fully uninitialized and reinitialized
without "officially" closing and reopening the playlist entry. This
happens with PT_RELOAD_FILE, which is triggered by edition switching and
also DVD/BD title switching. (Not really sure why it goes through so
much pain for such obscure cases. All it gains is not resetting "local"
options, and not signaling a reload to the client API. Whatever.)
The recent filter change freed filter_root too early without recreating
it, so it crashed on edition switching.
Fixes#5587.
If you used --aufio-file=file.mkv, and file.mkv included a video track
marked as default, then the logic in select_default_track() would pick
the video track from file.mkv. This is 100% broken, so fix it.
Before this commit, auto_loaded and lang were only set for the first
track in auto-loaded external files. Likewise, for the title and
lang arguments to the sub-add and audio-add commands.
Fixes#5432
Setting lavfi-complex at runtime will now forcefully reselect the tracks
as needed, even if it was a "proper" track selection via --aid or --vid.
Before this commit, it just failed and complained that the VO/AO was
already "used".
Requested.
This makes it actually somewhat simpler, and doesn't have any
disadvantages. It should also make some new features easier.
Mostly just moves code around.
The somewhat confusing thing is that many filters (including track->dec)
have a public struct, but to free them, you need to free the mp_filter
pointer itself (track->dec->f). The assignment wrote to a dangling
pointer, instead of removing the dangling pointer.
(Other than that, this idiom is actually nice.)
Use the decoder wrapper that was introduced for video. This removes all
code duplication the old audio decoder wrapper had with the video code.
(The audio wrapper was copy pasted from the video one over a decade ago,
and has been kept in sync ever since by the power of copy&paste. Since
the original copy&paste was possibly done by someone who did not answer
to the LGPL relicensing, this should also remove all doubts about
whether any of this code is left, since we now completely remove any
code that could possibly have been based on it.)
There is some complication with spdif handling, and a minor behavior
change (it will restrict the list of codecs to spdif if spdif is to be
used), but there should not be any difference in practice.
Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes
a source filter. vd.h mostly disappears, because mp_filter takes care of
the dataflow, but its remains are in struct mp_decoder_fns.
One goal is to simplify dataflow by letting the filter framework handle
it (or more accurately, using its conventions). One result is that the
decode calls disappear from video.c, because we simply connect the
decoder wrapper and the filter chain with mp_pin_connect().
Another goal is to eventually remove the code duplication between the
audio and video paths for this. This commit prepares for this by trying
to make f_decoder_wrapper.c extensible, so it can be used for audio as
well later.
Decoder framedropping changes a bit. It doesn't seem to be worse than
before, and it's an obscure feature, so I'm content with its new state.
Some special code that was apparently meant to avoid dropping too many
frames in a row is removed, though.
I'm not sure how the source code tree should be organized. For one,
video/decode/vd_lavc.c is the only file in its directory, which is a bit
annoying.
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.
If you play a video with an external audio track, and do backwards
keyframe seeks, then audio can be missing. This is because a backwards
seek can end up way before the seek target (this is just how this seek
mode works). The audio file will be seeked at the correct seek target
(since audio usually has a much higher seek granularity), which results
in silence being played until the video reaches the originally intended
seek target.
There was a hack in audio.c to deal with this. Replace it with a
different hack. The new hack probably works about as well as the old
hack, except it doesn't add weird crap to the audio resync path (which
is some of the worst code here, so this is some nice preparation for
rewriting it). As a more practical advantage, it doesn't discard the
audio demuxer packet cache. The old code did, which probably ruined
seeking in youtube DASH streams.
A non-hacky solution would be handling external files in the demuxer
layer. Then chaining the seeks would be pretty easy. But we're pretty
far from that, because it would either require intrusive changes to the
demuxer layer, or wouldn't be flexible enough to load/unload external
files at runtime. Maybe later.
The underlying logic is still the same (basically pausing if the demuxer
cache underruns), but clean up the higher level logic a bit. It goes
from 3 levels of nested if statements to 1.
Also remove the code duplication for the --cache-pause-initial logic.
In addition, make sure an earlier buffering state has no influence on
the new state after a seek (this is also why some of the state resetting
can be removed from loadfile.c).
Initialize cache_buffer always to 100. It basically means we start out
assuming all buffers are filled enough. This actually matters for
verbose messages only, but removes some weird special casing.
Before this commit, some autoselection of tracks coming from files
loaded with --external-files was still done. This commit removes all of
it, and the only way to select a track is via the explicit stream
selection options like --vid/--sid/--aid.
I think this was always the original intention. The change could in
theory still unintentionally surprise some users, so add a changelog
entry.
This does not affect --audio-file/--sub-file, even if these contain
mismatching track types. E.g. if audio files passed to --audio-file
contain subtitles, these should still be selected. Past feature requests
indicate that users want this.
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.
This will help with things like livestreams.
As a minor detail, subtitles are excluded, because they sometimes have
"unused" events after video and audio ends. To avoid this annoying
corner case, just ignore them.
Until now, using --sub-file would add only subtitle tracks from the
given file. (E.g. if you passed a video file, only the subtitle tracks
from it were added, not the video or audio tracks.)
This is slightly messy (because streams are hidden), and users don't
even want it, as shown by #5132. Change it to always add all streams.
But if there's no stream of the wanted type, we still report an error
and do not add any streams. It's also made sure none of the other track
types are autoselected.
Also adjust the error messages on load failure slightly.
Fixes#5132.
It appears libavformat never sets the file start time for subtitles, so
this special check is not needed. The original idea was probably that
_if_ the demuxer set the start time to the first subtitle packet, the
subtitles would be shifted incorrectly.
Added a get_play_start_pts function to coincide with the
already-existing get_play_end_pts. This prevents code duplication
and also serves to make it so code that probes the start time
(such as get_current_pos_ratio) will work correctly with chapters.
Included is a bug fix for misc.c/rel_time_to_abs that makes it work
correctly with chapters when --rebase-start-time=no is set.
Always display the duration as "unknown" if the duration is known. Also
fix that at least demux_lavf reported unknown duration as 0 (fix by
setting the default to unknown in demux.c).
Remove the dumb _u formatter function, and use a different approach to
avoiding displaying "unknown" as playback time on playback start (set
last_seek_pts for that).
This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).
The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.
Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.
For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.
Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.
Refresh seeks are automatically issued when changing filters, which
improves user experience if these filters change buffering or such.
The refresh seek could actually overwrite a previously ongoing seek:
set pause yes
set time-pos 10
set vf ""
Here, the video code issued a refresh seek to the previous video
position, which could be different from the previously triggered (and
still ongoing) seek, this overwriting the seek.
Factor all refresh seek handling into a new function, and make it handle
ongoing seeks correctly.
Remove the weird new canonical_pts field, which actually had no use.
Fixes#4757.
Tends to be somewhat glitchy if subtitles are enabled, and you enable
and disable tracks.
On error, this will disable --lavfi-complex, which will result in
whatever behavior.
Remove the various redundant m_config_set_option* calls, rename the
remaining one to m_config_set_option_cli(), and merge the
m_config_parse_option() function.
These files have all in common that they were fully or mostly taken from
mplayer.c. (mplayer.c was a huge file that contains almost all of the
playback core, until it was split into multiple parts.) This was
probably the hardest part to relicense, because so much code was moved
around all the time.
player/audio.c still does not compile. We'll have to redo audio
filtering. Once that is done, we can probably actually provide an
actual LGPL configure switch.
Here is a relatively detailed list of potential issues:
8d190244: author did not reply, parts were made GPL-only in a previous
commit.
7882ea9b: author could not be reached, but the code is gone. wscript
still has --datadir switch, but I don't think this is relevant to
copyright.
f197efd5: unclear origin, but I consider the code gone anyway (replaced
with generic OSD mechanisms).
8337d9c2: author did not reply, but only the option still exists (under
a different name), other code was removed.
d8fd7131: did not reply. Disabled in a previous commit.
05258251: same author as above. Both fields actually seem to have
vanished (even when tracking renames), so no action taken.
d459e644, 268b2c1a: author did not reply, but we reuse only the options
(with different names and slightly or fully different semantics, and
completely different implementations), so I don't think this is relevant
for copyright.
09e742fe, 17c39c4e: same as above.
e8a173de, bff4b3ee: author could not be reached. The commands were
reworked to properties, and the code outside of the TV code were moved
back to the TV code. So I don't think copyright applies to the current
command.c parts (mp_property_tv_color, mp_property_tv_freq,
mp_property_tv_scan). The TV parts remain GPL.
0810e427: could not be reached. Disabled in a previous commit.
43744a2d: unknown author, but this was replaced by dynamic alloc (if the
change is even copyrightable).
116ca0c7: unknown author; reasoning see input.c relicensing commit.
e7e4d1d8: these semantics still exist, but as generic code, and this
code was fully removed.
f1175cd9: the author of the cited patch is unknown, and upon inspection
it turns out that I was only using the idea to pause the player on EOF,
so I claim it's not copyright relevant.
25affdcc: author could not be reached (yet) - but it's only a function
rename, not copyrightable.
5728504c was committed by Arpi (who agreed), but hints that it might be
by a different author. In fact it seems to be mostly this patch:
http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2001-November/002041.html
The author did not respond, but it all seems to have been removed later.
It's a terrible mess though. Arpi reverted the A-V sync code at first,
but left the RTC code for a while. The following commits remove these
changes 100%: 14b35442, 7181a091, 31482783, 614f8475, df58e822.
cehoyos did explicitly not agree to LGPL, but was involved in the
following changes:
c99d8fc8: applied a patch and didn't modify it, the original author
agreed.
40ac0d31: author could not be reached, but all code is gone anyway. The
"af" command has a similar function, but works completely different and
actually reuses a mechanism older than this patch.
54350436: applied a patch, but didn't modify it, except for adding a
German translation, which was removed later.
a2dda036: same situation as above
240b743e: this was made GPL-only in a previous commit
7b25afd7: same as above (for now)
kirijua could not be reached, but was a regular patch contributor:
c2c997fd: video equalizer code move; probably not copyrightable. Is GPL
due to Nick anyway.
be54f481: technically, this became the audio track property later. But
all what is left is the fact that you pass a track ID to it, so consider
the original coypright non-relevant.
2f376d1b: this was rewritten in b7052b43, but for now we can afford to
be careful, so this was marked as GPL only in a previous commit.
43844d09: remaining parts in main.c were reverted in a previous commit.
anders has mostly disagreed with the LGPL relicensing. Does not want
libaf to become LGPL, but made some concessions. In particular, he
granted us permission to relicense 4943e9c52c and 242aa6ebd4. We also
consider some of his changes remaining in mpv not relevant for copyright
(such as 735de602 - we won't remove the this option completely). We will
completely remove his other contributions, including the entire audio
filter chain. For now, this stuff is marked as GPL only. The remaining
question is how much code in player/audio.c (based on the former
mplayer.c and dec_audio.c) is under his copyright. I made claims about
this in a previous commit.
Nick(ols) Kurshev, svn username "nick" and "nickols_k", could not be
reached. He had a lot of changes in early MPlayer. It seems all of that
was removed, at least in mpv. His main work, like VIDIX or libswscale
work, does not exist in mpv anymore, but the changes to mplayer.c and
other core parts still deserve attention:
a4119f6b, fb927549, ad3529b8, e11b23dc, 5f2178be, 93c371d5: removed in
b43d67e0, d1628d12, 24ed01fe, df58e822.
0a83c6ec, 104c125e, 4e067f62, aec5dcc8, b587a3d6, f3de6e6b: DR, VAA, and
"tune" stuff was fully removed later on or replaced with other
mechanisms.
340183b0: screenshots were redone later (the VOCTRL was even removed,
with an independent implementation using the same VOCTRL a few years
later), so not relevant anymore. Basically only the 's' shortcut remains
(but not its implementation).
92c5c274, bffd4007, 555c6766: for now marked as GPL only in a previous
commit.
Might contain some trace amounts of "michael"'s copyright, who agreed to
LGPL only once the core is relicensed. This will still be respected, but
I don't think it matters at this in this case. (Some code touched by him
was merged into mplayer.c, and then disappeared after heavy
refactoring.)
I tried to be as careful and as complete as possible. It can't be
excluded that amends to this will be made later.
This does not make the player LGPL yet.
Commit d8fd7131 changes this. "tibcu" did not reply. While I'm not sure
whether copyrightable code remains, I'd tend towards saying yes (the
basic idea is still intact after years of refactoring), so make it
GPL-only for now.
The previous commit set "mpctx->playback_active = false;" before unload
hooks were processed. This was intentional, but could in theory cause
playback_active to be set to true again, and actually it's plain wrong
if playback was exited in the middle it. There needs to be something
else that forces playback_active to be set to false while in this
unloading state.
Make mpv_observe_property() work correctly on them even with
--keep-open-pause=no.
This also changes the situations in which the screensaver is
enabled/disabled subtly.
Merge the pause_player() and unpause_player() functions. Make sure the
pause events are emitted properly. We can now set the internal pause
state based on a predicate, instead of e.g. handle_pause_on_low_cache()
making a mess to trigger the internal pause state as wanted.
Preparation for some more changes.
This was excessively useless, and I want my time back that was needed to
explain users why they don't want to use it.
It captured the byte stream only, and even for types of streams it was
designed for (like transport streams), it was rather questionable.
As part of the removal, un-inline demux_run_on_thread() (which has only
1 call-site now), and sort of reimplement --stream-dump to write the
data directly instead of using the removed capture code.
(--stream-dump is also very useless, and I struggled coming up with an
explanation for it in the manpage.)
Disabling cache readahead by default until at least 1 track is selected
is mainly for external files and such, where you don't want them to use
up resources until they're actually used.
It doesn't make sense to disable the cache for the demuxer opened for
prefetch. Also, it's fine to let it do that for the main file too (doing
or not doing it is of little consequence). That saves us from having to
distinguish them.
Since for mpv CLI, the player state is a singleton, full prefetching is
a bit tricky. We do it only on the demuxer layer.
The implementation reuses the old "open thread". This means there is
significant potential for regressions even if the new option is not
used. This is made worse by the fact that I barely tested this code.
The generic mpctx_run_reentrant() wrapper is also removed - this was its
only user, and its remains become part of the new implementation.
As preparation for file prefetching, we basically have to get rid of
using mpctx->playback_abort for the main demuxer (i.e. the thing that
can be prefetched). It can't be changed on a running demuxer, and always
using the same cancel handle would either mean aborting playback would
also abort prefetching, or that playback can't be aborted anymore.
Make this more flexible with some refactoring.
Thi is a quite shitty solution if you ask me, but YOLO.
Was intended to show a "nice" message on edition switching. In practice,
the message was never visible. The OSD code checks whether a demuxer is
loaded, and if not, discards the message - meaning if the OSD code
happened to run before the demuxer was fully loaded, no message was
shown. This is apparently a regression due to extensions to the OSD and
the situations in which it can be used.
Remove the broken code since it's too annoying to fix. Instead, a
default property message will be shown, which is a bit uglier, but
actually not too unuseful.
The way playback/loading is stopped on the demuxer layer makes it report
an error to the higher levels of the player. But if playback/loading was
explicitly aborted, printing such an error is confusing and misleading.
This was probably just an oversight anyway. Fix it by using the libmpv
API reported error field instead, which handles this better.
Move the screensaver enable/disable determination to a central place,
and call it if the stop-screensaver property is changed.
Also, do not stop the screensaver when in idle mode (i.e. no file is
loaded).
Fixes#3615.
Add this flag where needed. You shouldn't be able to set e.g. config-dir
in these situations.
Remove the mpctx->initialized check from the property/option bridge,
since it's in use strictly only after initialization. Likewise, the
apply-profile command doesn't need to check this.
Move the MPV_LEAK_REPORT env query to mp_create(), where it will also be
used by the client API (it might be helpful, so why not). The same
applies to MPV_VERBOSE.
The prepare_playlist() call doesn't need to be in mp_initialize() and
can just be in mp_play_files() to reduce the size of mp_initialize().
Also, remove wakeup_playloop(), which is 100% redundant with
mp_wakeup_core_cb().
This does 3 kinds of changes:
- change sleeptime=x to mp_set_timeout()
- change sleeptime=0 to mp_wakeup_core() calls (to be more explicit)
- change commands etc. to call mp_wakeup_core() if they do changes that
require the playloop to be rerun
This is preparation for the following changes. The goal is to process
client API requests without having to rerun the playloop every time. As
of this commit, the changes should not change behavior. In particular,
the playloop is still implicitly woken up on every command.
Currently, calling mp_input_wakeup() will wake up the core thread (also
called the playloop). This seems odd, but currently the core indeed
calls mp_input_wait() when it has nothing more to do. It's done this way
because MPlayer used input_ctx as central "mainloop".
This is probably going to change. Remove direct calls to this function,
and replace it with mp_wakeup_core() calls. ao and vo are changed to use
opaque callbacks and not use input_ctx for this purpose. Other code
already uses opaque callbacks, or has legitimate reasons to use
input_ctx directly (such as sending actual user input).
Remove the per-part force_redraw flags, and instead make the difference
between flagging dirty state and returning it to the player frontend
more explicit. The big issue is that 1. the OSD needs to know the dirty
state, and it should be cleared strictly when it is re-rendered
(force_redraw flag), and 2. the player core needs to be notified once,
and the notification must be reset (want_redraw flag).
The call in loadfile.c is replaced by making osd_set_sub() set the
change flag. Increasing the change flag on dirty state (the force_redraw
check in render_object()) should not be needed, because OSD part
renderers set it correctly (at least now).
Doing this just because someone pointed this out.
This has all been made unnecessary recently. The change not to copy the
global option struct in particular can be made because now nothing
accesses the global options anymore in the demux and stream layers.
Some code that was accidentally added/changed in commit 5e30e7a0 is also
removed, because it was simply committed accidentally, and was never
used.
Cleaner and makes it easier to change the underlying stream.
mp_property_stream_capture() still directly accesses it directly via
demux_run_on_thread(). This is evil, but still somewhat sane and is not
getting into the way here.
Not sure if I got all field accesses.
Change the last parameter from a bool to an int, which is supposed to
take bit-flags. The at this point only flag is MPSEEK_FLAG_DELAY, which
replaces the previous bool parameter. The old false parameter becomes 0,
the old true parameter becomes MPSEEK_FLAG_DELAY.
Since the old "immediate" parameter is now essentially inverted, two
coalesced immediate and delayed seeks end up as delayed instead of
immediate. This change doesn't matter, since there are no relative
immediate seeks anyway.
When switching tracks, we normally have the problem that data gets lost
due to readahead buffering. (Which in turn is because we're stubborn and
instruct the demuxers to discard data on unselected streams.) The
demuxer layer has a hack that re-reads discarded buffered data if a
stream is enabled mid-stream, so track switching will seem instant.
A somewhat similar problem is when all tracks of an external files were
disabled - when enabling the first track, we have to seek to the target
position.
Handle these with the same mechanism. Pass the "current time" to the
demuxer's stream switch function, and let the demuxer figure out what to
do. The demuxer will issue a refresh seek (if possible) to update the
new stream, or will issue a "normal" seek if there was no active stream
yet.
One case that changes is when a video/audio stream is enabled on an
external file with only a subtitle stream active, and the demuxer does
not support rrefresh seeks. This is a fuzzy case, because subtitles are
sparse, and the demuxer might have skipped large amounts of data. We
used to seek (and send the subtitle decoder some subtitle packets
twice). This case is sort of obscure and insane, and the fix would be
questionable, so we simply don't care.
Should mostly fix#3392.
Assume you use a large value like --audio-delay=20. Then until now the
player would just have seeked normally to a "too late" position, and
played silence for about 20 seconds until audio in the correct time
range is coming again.
Change this by offsetting seeks by the right amount. This works for both
external and muxed files. If a seek isn't precise, then it works only
for external files.
This might cause issues with very large delay options. Hr-seek skipping
could take a lot of time (especially because it affects video too), the
demuxer queue could overflow, and other weird corner cases could appear.
But we just try this on best-effort basis, and if the user uses extreme
values we don't guarantee good behavior.
mixer.c didn't really deserve to be separate anymore, as half of its
contents were unnecessary glue code after recent changes. It also
created a weird split between audio.c and af.c due to the fact that
mixer.c could insert audio filters. With the code being in audio.c
directly, together with other code that unserts filters during runtime,
it will be possible to cleanup this code a bit and make it work like the
video filter code.
As part of this change, make the balance code work like the volume code,
and add an option to back the current balance value. Also, since the
balance semantics are unexpected for most users (panning between the
audio channels, instead of just changing the relative volume), and there
are some other volumes, formally deprecate both the old property and the
new option.
Calculate the buffering percentage in the same code which determines
whether the player is or should be buffering. In particular it can't
happen that percentage and buffering state are slightly out of sync due
to calling DEMUXER_CTRL_GET_READER_STATE and reusing it with the
previously determined buffering state.
Now it's also easier to guarantee that the buffering state is updated
properly.
Add some more verbose output as well.
(Damn I hate this code, why did I write it?)
Ever since a change in mplayer2 or so, relative seeks were translated to
absolute seeks before sending them to the demuxer in most cases. The
only exception in current mpv is DVD seeking.
Remove the SEEK_ABSOLUTE flag; it's not the implied default. SEEK_FACTOR
is kept, because it's sometimes slightly useful for seeking in things
like transport streams. (And maybe mkv files without duration set?)
DVD seeking is terrible because DVD and libdvdnav are terrible, but
mostly because libdvdnav is terrible. libdvdnav does not expose seeking
with seek tables. (Although I know xbmc/kodi use an undocumented API
that is not declared in the headers by dladdr()ing it - I think the
function is dvdnav_jump_to_sector_by_time().) With the current mpv
policy if not giving a shit about DVD, just revert our half-working seek
hacks and always use dvdnav_time_search(). Relative seeking might get
stuck sometimes; in this case --hr-seek=always is recommended.
Some oddity that is not needed anymore. The only thing which still
referenced them was avoiding loading external files more than once,
which is now prevented by checking the list of tracks instead.
When playback of a video ends, and the next file has no video at all (no
cover art or anything), then the window must be cleared.
This also resizes the window forcibly, which is by design.
Fixes#2825.
See --lavfi-complex option.
This is still quite rough. There's no support for dynamic configuration
of any kind. There are probably corner cases where playback might freeze
or burn 100% CPU (due to dataflow problems when interaction with
libavfilter).
Future possible plans might include:
- freely switch tracks by providing some sort of default track graph
label
- automatically enabling audio visualization
- automatically mix audio or stack video when multiple tracks are
selected at once (similar to how multiple sub tracks can be selected)
Slightly helps with timeline stuff, like EDL. There is no need to keep
network (or even just disk I/O) busy for all segments at the same time,
because 1. the data won't be needed any time soon, and 2. will probably
be discarded anyway if the stream is seeked when segment is resumed.
Partially fixes#2692.
Eventually we want the VO be driven by a A->V filter, so a decoder
doesn't even have to exist. Some features definitely require a decoder
though (like reporting the decoder in use, hardware decoding, etc.), so
for each thing which accessed d_video, it has to be redecided if and how
it can access decoder state.
At least the "framedrop" property slightly changes semantics: you can
now always set this property, even if no video is active.
Some untested changes in this commit, but our bio-based distributed
test suite has to take care of this.
Basically reimplement it. The old implementation was quite stupid, and
was probably done this way because video filtering and output used to be
way less decoupled. Now we can reimplement it in a very simple way: when
backstepping, seek to current time, but keep the last frame that was
supposed to be discarded when reaching the target time. When the seek
finishes, prepend the saved frame to the video frame queue.
A disadvantage is that the new implementation fails to skip over
timeline boundaries (ordered chapters etc.), but this never worked
properly anyway. It's possible that this will be fixed some time in the
future.
This is mainly a refactor. I'm hoping it will make some things easier
in the future due to cleanly separating codec metadata and stream
metadata.
Also, declare that the "codec" field can not be NULL anymore. demux.c
will set it to "" if it's NULL when added. This gets rid of a corner
case everything had to handle, but which rarely happened.
