This uses the normal autoprobing rules like "auto", but rejects anything
that isn't flagged as copying data back to system memory.
The chunk in command.c was dead code, so remove it instead of updating
it.
Commit 786f37ae accidentally changed seeking behavior such that
continuous seeking (holding the seek button down) would use the previous
seek target timestamp, instead of the new video timestamp. (This is for
the default mode, seeking to keyframes.)
The result is that the movement on the seekbar is smooth, but the way
the video updates is awkward. Some might actually prefer the new
behavior (and some players effectively show similar bahavior), but I
don't. So restore the old behavior.
This is done in two steps:
First: strictly wait for the entire seek process to finish, which will
effectively make the seeking code pick up the new video timestamp
correctly.
This would play audio immediately, which would result in noise during
continuous seeking, which leads to second: explicitly abort the playback
restarting process if this case is detected, and never play audio.
The main change is with video/hwdec.h. mp_hwdec_info is made opaque (and
renamed to mp_hwdec_devices). Its accessors are mainly thread-safe (or
documented where not), which makes the whole thing saner and cleaner. In
particular, thread-safety rules become less subtle and more obvious.
The new internal API makes it easier to support multiple OpenGL interop
backends. (Although this is not done yet, and it's not clear whether it
ever will.)
This also removes all the API-specific fields from mp_hwdec_ctx and
replaces them with a "ctx" field. For d3d in particular, we drop the
mp_d3d_ctx struct completely, and pass the interfaces directly.
Remove the emulation checks from vaapi.c and vdpau.c; they are
pointless, and the checks that matter are done on the VO layer.
The d3d hardware decoders might slightly change behavior: dxva2-copy
will not use the VO device anymore if the VO supports proper interop.
This pretty much assumes that any in such cases the VO will not use any
form of exclusive mode, which makes using the VO device in copy mode
unnecessary.
This is a big refactor. Some things may be untested and could be broken.
Add --taskbar-progress command line option and property which controls taskbar
progress indication rendering in Windows 7+. This option is on by default and
can be toggled during playback.
This option does not affect the creation process of ITaskbarList3. When the
option is turned off the progress bar is just hidden with TBPF_NOPROGRESS.
Closes#2535
Introduce hwdec-current and hwdec-interop properties.
Deprecate hwdec-detected, which never made a lot of sense, and which is
replaced by the new properties. hwdec-active also becomes useless, as
hwdec-current is a superset, so it's deprecated too (for now).
For "current" markers on OSD properties like chapter-list. The marker is
now an actual arrow instead of "> ", and non-current entries will have
the same indentation as the current entry.
While I'm not entirely sure about the new look of those lists, it's a
bit better than the visual mess that was before.
Because it's annoying and feels unnatural.
If the B point is set while paused, don't seek. If not paused, it should
properly loop immediately.
In theory there's a chance that it will show at least 1 frame after the
loop point when setting the B point. But let's not care about that.
This fixes backstepping getting "stuck" when e.g. holding down a key
bound to the backstep command. The reason is that even if the backstep
itself is finished, the next backstep might not take the new video PTS
as reference if the hr-seek itself isn't finished yet.
The intention of not waiting for the hr-seek to finish was faster
backstepping by possibly skipping audio decoding. But it probably
doesn't matter enough to make the rest of the code more complex.
As a positive side-effect, this also errors out gracefully for the
extremely unlikely but possible case certain builtin filters are not
available. (This could happen only with crippled libavfilter builds that
can't be used by anything using its public API.)
Another crappy fix for timestamp reset issues. This time, we try to fix
files which have very weird but legitimate frame durations, such as
cdgraphics. It can have many short frames, but once in a while there are
potentially very long frames.
Fixes#3027.
Commit 382bafcb changed the behavior for ab-loop-a. This commit changes
ab-loop-b so that the behavior is symmetric.
Adjust the OSD rendering accordingly to the two changes.
Also fix mentions of the "ab_loop" command to the now preferred
"ab-loop".
The check whether video is ready yet was done only in STATUS_FILLING.
But it also switched to STATUS_READY, which means the next time
fill_audio_out_buffers() was called, audio would actually be started
before video.
In most situations, this bug didn't show up, because it was only
triggered if the demuxer didn't provide video packets quickly enough,
but did for audio packets.
Also log when audio is started.
(I hate fill_audio_out_buffers(), why did I write it?)
Strictly schedule an update in regular intervals as long as either
stream cache or demuxer are prefetching. Don't update just always
because the stream cache is enabled ("idle != -1") or cache-related
properties are observed (mp_client_event_is_registered()).
Also, the "idle" variable was awkard; get rid of it with equivalent
code.
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?)
And remove the same thing from the client API code.
The command.c code has to deal with many specialized M_PROPERTY_SET_*
actions, and we bother with a subset only.
If a mpv_node wrapped a string, the behavior was different from calling
mpv_set_property() with MPV_FORMAT_STRING directly. Change this.
The original intention was to be strict about types if MPV_FORMAT_NODE
is used. But I think the result was less than ideal, and the same change
towards less strict behavior was made to mpv_set_option() ages ago.
Commit 57506b27 accidentally broke this. The status (including the
usually always active demuxer cache) should be shown only if the stream
cache is actually enabled.
Instead of having a separate for each, which also requires separate
additional caching in the demuxer. (The demuxer adds an indirection,
since STREAM_CTRLs are not thread-safe.)
Since this includes the cache speed, this should fix#3003.
This would get stuck in reconfiguring the filter chain forever, because
params was mutated ("params.rotate = 0;"). This was used as input for
vf_reconfig(), but the filter chain input must always be equivalent to
the decoder output, or filter chain reconfiguration will be triggered.
The line of code to reset the rotation is from a time when this used to
work differently.
Also remove the unnecessary try_filter() parameter.
This implements the JSON IPC protocol with named pipes, which are
probably the closest Windows equivalent to Unix domain sockets in terms
of functionality. Like with Unix sockets, this will allow mpv to listen
for IPC connections and handle multiple IPC clients at once. A few cross
platform libraries and frameworks (Qt, node.js) use named pipes for IPC
on Windows and Unix sockets on Linux and Unix, so hopefully this will
ease the creation of portable JSON IPC clients.
Unlike the Unix implementation, this doesn't share code with
--input-file, meaning --input-file on Windows won't understand JSON
commands (yet.) Sharing code and removing the separate implementation in
pipe-win32.c is definitely a possible future improvement.
Should reflect I/O speed.
This could go into the terminal status line. But I'm not sure how to put
it there, since it already uses too much space, so it's not there yet.
This changes behavior somewhat. The old behavior can be restored by
running "mp.use_suspend=true". It was originally introduced for the OSC,
but I can't reproduce whatever misbehavior I was seeing.
(See mp.suspend()/resume() for explanations what the suspend mechanism
does.)
This pause stuff is bothersome and is needed only for a few corner-
cases. This commit removes it from the demuxer public API and replaces
it with a demux_run_on_thread() function and refactors the code which
needed demux_pause(). The next commit will change the implementation.
Changing the byte stream position without cooperation of the demuxer
seems a bit insane, and is certainly useless. A user should do factor
seeks instead. For formats like ts, this will actually translate to byte
seeks, while treating the rest of the playback chain a bit more
gracefully. With this argument, remove write access to this property.
If someone really complains, proper byte seeks could be added as seek
mode (although I'm going to need a convincing argument for this).
Read access changes too, but in a more subtle way.
No need to have them everywhere. The only exception/annoyance is
MAX_OSD_PARTS, which is now basically duplicated (and at runtime
initialization is checked with an assert()).
Until now, there was only 1 global ASS overlay that could be set by all
scripts. This was often perceived as bug when multiple scripts tried to
set their own ASS overlay.
This was kind of hard to solve because the script could set its own ASS
PlayResX/Y, which makes it impossible to share a single ASS_Renderer for
multiple scripts. The OSC unfortunately makes use of this feature (and
unfortunately can't be fixed because it's a POS), so we're stuck with
this complication.
Implement the worst-case solution and fix this by creating separate ASS
track and renderer objects for each script that wants to set an ASS
overlay.
The z-order is decided by the order the scripts set their text first.
This is essentially random, unless you do it at script init, and you
pass scripts in a specific order. Script initialization is currently
serialized (as a feature), so the first loaded script gets lowest
Z-order.
The Lua script API interestingly remains the same. (And also will remain
undocumented, unsupported, and potentially volatile.)
Do not scale OSD mouse input to the ASS OSD script resolution. The
original idea of this mechanism was that the user doesn't have to care
about the actual resolution of anything, and can just use the OSD
resolution consistently. But this made things worse.
Remove the implicit scaling, and always use the screen resolution.
(Except with --vo=xv, where additional scaling is forced upon
everything.)
Drop get_osd_resolution(). There is no replacement. Rename
get_screen_size() and get_screen_margins() to use "osd" instead of
"screen". For anything but --vo=xv these are equivalent, but with
--vo=xv the OSD resolution has additional implicit scaling.
Add code to osc.lua which emulates the old behavior.
Note that none of the changed functions were public API, so implicit
breakage of scripts which used it is just going to happen.
Subtitles can be preloaded, which means they're fully read and copied
into ASS_Track. This in turn is mainly for the sake of being able to do
subtitle seeking (when it comes down to it, subtitle seeking is the
cause for most trouble here).
Commit a714f8e92 broke preloaded subtitles which have events with
unknown duration, such as some MicroDVD samples. The event list gets
cleared on every seek, so the property of being preloaded obviously gets
lost.
Fix this by moving most of the preloading logic to dec_sub.c. If the
subtitle list gets cleared, they are not considered preloaded anymore,
and the logic for demuxed subtitles is used.
As another minor thing, preloadeding subtitles did neither disable the
demux stream, nor did it discard packets. Thus you could get queue
overflows in theory (harmless, but annoying). Fix this by explicitly
discarding packets in preloaded mode.
In summary, now the only difference between preloaded and normal
demuxing are:
1. a seek is issued, and all packets are read on start
2. during playback, discard the packets instead of feeding them to the
subtitle decoder
This is still petty annoying. It would be nice if maintaining the
subtitle index (and maybe a subtitle packet cache for instant subtitle
presentation when seeking back) could be maintained in the demuxer
instead. Half of all file formats with interleaved subtitles have
this anyway (mp4, mkv muxed with newer mkvmerge).
Commit 8d4a179c made subtitle decoders pick up fonts strictly from the
same source file (i.e. the same demuxer).
It breaks some fucked up use-case, and 2 people on this earth complained
about the change because of this. Add it back.
This copies all attached fonts on each subtitle init. I considered
converting attachments to use refcounting, but it'd probably be much
more complex.
Since it's slightly harder to get a list of active demuxers with
duplicate removed, the prev_demuxer variable serves as a hack to achieve
almost the same thing, except in weird corner cases. (In which fonts
could be added twice.)
Was only available via --vd=help and --ad=help (i.e. not at all via
client API). Not bothering with separating audio and video codecs, since
this list isn't all that useful anyway in general. If someone complains,
a type field could be added.
Export a number of container fields, which may or may not be useful in
some scenarios. They are explicitly marked as originating from the
demuxer, in order to make it explicit that they might be unreliable.
I'd actually like to remove all other cases where container information
is exported, but those numerous cases are going to be somewhat hard to
deprecate.
Also, not directly related, export the description of the currently
active decoder. (This has been requested before.)
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.
Adds always-on mode by internally utilizing hidetimeout as negative and
forbidding the user to set negative values.
This removes script-message to enable/disable the osc, and instead introduces a
combined 'visibility' control with the values never/auto/always.
It's available via script_opts and script_message as 'osc-visibility'.
As message, it also supports a 'cycle' value.
The del key is bound to cycling the visibility modes.
There were few issues:
- When it's disabled and then enabled, it was displaying the osc briefly and
then autohide right away. Don't do that.
- When it's enabled and then disabled, it was not removing the osc from screen
if called while the osc is visible (because tick() is responsible for the hide
but it doesn't render() the empty osc when the osc is disabled).
- Due to delayed/async unbinding of mouse events it was possible to show_osc()
after it got disabled e.g. from mouse_move. Prevent this.
_Of course_ the previous commit broke --force-window behavior (like it
does every single time I touch it).
vo_has_frame() gets cleared after a seek, so e.g. stopping playback of a
file and going to the next by keeping the seek key down will enter a
short moment without video at the end of the first file, which will set
the stalled_video variable to true. Prevent it by using the indication
whether the window was properly created (which is probably exactly what
we want here).
This function is also responsible for destroying the window when needed,
and obviously we should never do that while video is active. (This is
the actual bug, although the other change in this commit already hides
the common breakage it caused.)
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.
Especially useful to see what video formats are involved on the various
filter links.
I suspect this function is not available on Libav, so add necessary
ifdeffery preemptively.
It would make somewhat sense for libcs which don't implement locales at
all, such as Bionic.
Beyond that, setlocale() is specified that it can return NULL, and we
shouldn't crash if that happens.
Unfortunately I see no better solution.
The refresh seek is skipped if the amount of buffered audio is not
overly huge.
Unfortunately softvol af_volume insertion still can cause this issue,
because it's outside of the normal dynamic filter chain changing code.
Move the video refresh call to reinit_video_filters() to make it more
uniform along with the audio code.
This was dumb. Could make it burn 100% CPU and not exit at the end.
(Because it would retry as instructed, instead of terminating playback.)
It also needs to consider EOF as waiting for input. lavfi_process() will
decide if it's really EOF, or if further input might come in the future.
Without this, it'd would think that it does not need to wait for input,
i.e. that new input will be available immediately.
(Not so fond of the duplication of subtle logic.)
It doesn't provide this function. The code is not really designed to
work without it, so it will probably mess up big time, but at least
make it compile again.
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)
track can't be NLUL at this point, so the if is redundant. Remove it and
unindent the block. Also, make the function check whether the track is
selected at all, which makes it safer and idempotent.
Will be helpful for the coming filter support. I planned on merging
audio/video decoding, but this will have to wait a bit longer, so only
remove the duplicate status codes.
Let's fix broken samples with questionable heuristic without real
reasoning. Until this gets fixed properly, this is a good compromise,
though. A proper fix would properly resync audio and video without
brutally resetting the decoders, but on the other hand not doing the
brutal reset would cause issues in other obscure corner cases such
resyncing might cause.
This code is tricky because it has to wakeup the mainloop to make
progressing during syncing audio, but also has to avoid waking it up
when it's not needed. Failure to do so either burns CPU by not ever
going to sleep, or causes apparent "freezes" by going to sleep (and it
will continue if the mainloop is woken up e.g. due to user input).
In this case, simply starting A/V playback with --start=5 and removing
an unrelated wakeup in osd.c can trigger such a "freeze". The unrelated
wakeup did hide this bug, nonetheless it's a bug.
(Can't wait to rewrite this shitty audio resync code. And it's all my
fault.)
We just need to provide an entrypoint for it, and move the main init
code to a separate function. This gets rid of the messy video chain full
reinit in command.c, which completely destroyed and recreated the video
state for the purpose of mid-stream hw/sw switching.
These changes don't make too much sense without context, but are
preparation for later. Then the audio_src/video_src fields will be
actually be NULL under circumstances.
Before this commit, reinit_audio_chain() did 2 things: create all the
management data structures and initialize the decoder, and handling lazy
filter/output init (as well as dealing with format changes). For the
second purpose, it could be called multiple times (even though it wasn't
really idempotent). This was pretty weird, so make them separate
functions. The new function is actually idempotent too.
It also turns out the reinit functions don't have to call themselves
recursively for the spdif PCM fallback.
Regression caused by commit 3b95dd47. Also see commit 4c25b000. We can
either use video_next_pts and add "delay", or we just use video_pts. Any
other combination breaks. The reason why the assumption that delay==0 at
this point was wrong exactly because after displaying the first video
frame (usually done before audio resync) a new frame might be "added"
immediately, resulting in a new video_next_pts and "delay", which will
still amount to video_pts.
Fixes#2770. (The reason why display-sync was blamed in this issue is
because enabling display-sync in the options forces a prefetch by 2
instead of 1 frames for seeks/playback restart, which triggers the
issue, even if display-sync is not actually enabled. In this case,
display-sync is never enabled because the frames have a unusually high
frame duration. This is also what exposed the initial desync issue.)
This seems generally easier when using libmpv (and was already requested
and implemented before: see commit 327a779a; it was reverted some time
later).
With the weird internal logic we have to deal with, in particular the
--softvol=no case (using system volume), and using the audio API's mixer
(--softvol=auto on some systems), we still can't avoid all glitches and
corner cases that complicate this issue so much. The API user is either
recommended to use --softvol=yes or auto, or to watch the new
mixer-active property, and assume the volume/mute properties have
significant values if the mixer is active.
Remaining glitches:
- changing the volume/mute properties has no effect if no internal mixer
is used (--softvol=no) and the mixer is not active; the actual mixer
controls do not change, only the property values
- --volume/--mute do not have an effect on the volume/mute properties
before mixer initialization (the options strictly are only applied
during mixer init)
- volume-max is 100 while the mixer is not active
With the format left untouched, this would just try to reinit with a
spdif format again.
We're not clearing the format in reset_audio_state() so the audio chain
can be recreated any time without having to wait for a frame to be
decoded.
Even though the timing logic is correct, it tends to mess with looping
videos and such in unappreciated ways.
It also has to be admitted that most file formats seem not to properly
define the duration of the last video frame (or libavformat does not
export it in a useful way), so whether or not we should use the demuxer
reported framerate for the last frame is questionable. (Still, why would
you essentially just discard the last frame?)
The timing logic is kept, but disabled for video with "normal" FPS
values. In particular, we want to keep it for displaying images, which
implicitly set the frame duration to 1 second by reporting 1 FPS. It's
also good for slide shows with mf://.
Fixes#2745.
Most text subtitles are read completely on loading (libavformat works
this way, and there are good reasons to do it on the higher levels too).
This leads to some messy problems. For example, the subtitle path is the
only one which might read packets during decoder initialization. This is
not neccessary; get rid of it.
This fixes a potential problem of seeking to position 0 for image
subtitles on init, and if the start position is not at the beginning of
the timeline.
It doesn't need to be part of the big context, but is strictly part of
shuffling data from the audio filters to audio output, and thus belongs
into ao_chain.
It also turns out that clearing it in clear_audio_output_buffers() is
completely redundant.
(Of course ao_buffer is an abomination in the first place and shouldn't
exist at all.)
vo_chain_uninit() isn't supposed to care much about the decoder
(although decoders and outputs still go strictly together, so there is
not much of an actual difference now).
Also unset track.d_video correctly.
Remove a stale declaration from dec_video.h as well.
Similar to the video path. dec_audio.c now handles decoding only. It
also looks very similar to dec_video.c, and actually contains some of
the rewritten code from it. (A further goal might be unifying the
decoders, I guess.)
High potential for regressions.
Seems useless.
This only helped in one case: one audio stream in the sample
av_find_best_stream_fails.ts had a AC3 packets which couldn't be
decoded, and for which avcodec_decode_audio4() returned 0 forever. In
this specific case, playback will now not start, and you have to
deselect audio manually.
(If someone complains, the old behavior might be restored, but
differently.)
Also remove the stale "bitrate" field.
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.
There are probably more files to which this applies, but I'm being
conservative here.
A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).
common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.
codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.
From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).
misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.
screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
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.
This moves some code related to decoding from video.c to dec_video.c,
and also removes some accesses to dec_video.c from the filtering code.
dec_video.ch is starting to make sense, and simply returns video frames
from a demuxer stream. The API exposed is also somewhat intended to be
easily changeable to move decoding to a separate thread, if we ever want
this (due to libavcodec already being threaded, I don't see much of a
reason, but it might still be helpful).
Makes the next commit simpler. It's probably a bad idea to add more
fields to the global state, but on the other hand the client API state
is pretty much per-instance anyway. It also will help with things like
the proposed libmpv custom stream API.
The aspect ratio calculations are cached (mainly so that aspect ratio
related messages are not logged on every frame). The cache is not clared
anymore when video filters are reconfigured, but changing the
video-aspect-ratio property relied on it. Make it explicit.
Fixes#2714.
The binding is similar to the tv-binding, just with capital letters.
Switching the dvb-channel-name property compared to dvb-channel
means the channel-name is shown on-screen when switching instead of
"dvb-channel (error)" otherwise,
and switching anyways happens without changing the card.
Channel switching is treated inside the global DVB state
by now. Anyways the last switching direction is not really useful
and of no interest inside the player.
On read, it returns the name of the current DVB program,
on write, it triggers a channel-switch to the program
if it is found in the channel list of the currently active card.
Compared to the dvb-channel property which already exists
and is a pair of integers (card + channel number) this has the limitation
of not switching the card, but is probably of much more common use.
Lots of noise to remove the vfilter/vo fields from dec_video.
From now on, video filtering and output will still be done together,
summarized under struct vo_chain.
There is the question where exactly the vf_chain should go in such a
decoupled architecture. The end goal is being able to place a "complex"
filter between video decoders and output (which will culminate in
natural integration of A->V filters for natural integration of
libavfilter audio visualizations). The vf_chain is still useful for
"final" processing, such as format conversions and deinterlacing. Also,
there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems
ideal, since otherwise there would be no natural way to handle all these
existing options and mechanisms.
There is still some work required to truly decouple decoding.
Instead of handling this on filter chain reinit, do it directly after
the decoder. This makes the code less entangled. In particular, this
gets rid of the really weird "override params" concept in the video
filter code.
The last_format/fixed_formats have some redundance with decoder_output,
but unfortunately the latter has a slightly different use.
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.
This slightly changes behavior when seeking with external audio/subtitle
tracks if transport streams and mpeg files are played, as well as
behavior when seeking with such external tracks.
get_main_demux_pts() is evil because it always blocks on the demuxer (if
there isn't already a packet queued). Thus it could lock up the player,
which is a shame because all other possible causes have been removed.
The reduced "precision" when seeking in the ts/mpeg cases (where
SEEK_FACTOR is used, resulting in byte seeks instead of timestamp seeks)
might lead to issues. We should probably drop this heuristic. (It was
introduced because there is no other way to seek in files with PTS
resets with libavformat, but its value is still questionable.)
This is another attempt at making files with sparse video frames work
better.
The problem is that you generally can't know whether a jump in video
timestamps is just a (very) long video frame, or a timestamp reset. Due
to the existence of files with sparse video frames (new frame only every
few seconds or longer), every heuristic will be arbitrary (in general,
at least).
But we can use the fact that if video is continuous, audio should also
be continuous. Audio discontinuities can be easily detected, and if that
happens, reset some of the playback state.
The way the playback state is reset is rather radical (resets decoders
as well), but it's just better not to cause too much obscure stuff to
happen here. If the A/V sync code were to be rewritten, it should
probably strictly use PTS values (not this strange time_frame/delay
stuff), which would make it much easier to detect such situations and
to react to them.
PT_RELOAD_FILE is a somewhat obscure case when using DVB or when
switching Matroska editions. Both cases were broken, because the
asynchronous playback abort mechanism was still triggered. This
mechanism is used to force the demuxer and stream layers to exit
immediately (instead of blocking on I/O possibly forever), and
is normally disabled on playback start. The reopen path is a bit
strange, and needs to reset it manually.
Pointed out in #2568.
If you do "mpv /bla/", and then branch out into sub-directories using
playlist navigation, and then used quit and watch later, then playing
the same directory did not resume from the previous point. This was
because resuming is based on the path hash, so a path prefix can't be
detected when resuming the parent directory.
Solve this by writing each path prefix when playing directories is
involved. (This includes all parent paths, so interestingly, "mpv /"
would also resume in the above example.)
Something like this was requested multiple times, and I want it too.
When using --start with timeline/ordered chapters, then the
timeline_switch_to_time() function will look at playback_initialized
whether to rselect the currently selected streams on the demuxer level.
So we need to set this field to true at an earlier stage during
initialization, and in particular before the code for --start is called.
Slightly change how it is decided when a new packet should be read.
Switch to demux_read_packet_async(), and let the player "wait properly"
until required subtitle packets arrive, instead of blocking everything.
Move distinguishing the cases of passive and active reading into the
demuxer, where it belongs.
Just simplify by removing parts not needed anymore. This includes
merging dec_sub allocation and initialization (since things making
initialization complicated were removed), or format support queries (it
simply tries to create a decoder, and if that fails, tries the next
one).
So that the video FPs is not required at initialization, and can be set
later.
(As for whether this MicroDVD crap is worth the trouble to handle it
"correctly": MicroDVD files are unfortunately still around, and in at
least one case using the video FPS seemed to help indeed.)
Keeping ASS_Renderers around for a potentially large number of subtitle
tracks could lead to excessive memory usage, especially since the libass
cache is broken (caches even unneeded data), and might consume up to
~500MB of memory for no reason.
This includes the case of switching ordered chapter boundaries. It will
now be recreated on each timeline part switch. This shouldn't be much of
a problem with modern libass. (Older libass versions use fontconfig for
memory fonts, and will be very slow to reinitialize memory fonts.)
Since commit 6d9cb893, subtitle state doesn't survive timeline switches
(ordered chapters etc.). So there is no point in caching the state per
sh_stream anymore (which would be required to deal with multiple
segments). Move the cache to struct track.
(Whether it's worth caching the subtitle state just for the situation
when subtitle tracks get reselected is questionable. But for now, it's
nice to have the subtitles immediately show up when reselecting a
subtitle.)
For files with only 1 chapter, the "cycle" command was ignored. Reenable
it, but don't let it terminate playback of the file.
For the full story, see #2550.
Fixes#2550.
OK, this made the --sub-paths and --audio-file-paths synonyms, which is
not what we wanted. Actually restrict the type of file loaded as well.
Really fixes#2632.
Requested. It works like --sub-paths. This will also load audio files
from a "audio" sub directory in the config file (because the same code
as for subtitles is used, and it also had such a feature).
Fixes#2632.
When crossing timeline boundaries (such as switching to a new segment or
chapter with ordered chapters), clear the internal text subtitle list.
This breaks the sub-seek command, but is otherwise not too harmful.
Fixes Sub-OC-test-final7.mkv. (The internal text subtitle list is
basically a cache to make subtitles show up at the right time when
seeking back.)
I suspect this was caused by 76fcef61. The sample file times subtitles
slightly before the video frame when it should show up. This is to avoid
problems with subtitles showing up a frame later than intended. It also
means that a subtitle which is supposed to show up on the start of a
timeline part boundary actually might first be shown in a different
part. Since we now manipulate the packet timestamps, instead of
manipulating timestamps after the subtitle decoder, this means this
subtitle event would have 2 timestamps, which our code of course does
not handle.
If the two parts come one after another, this would actually work (since
the subtitle would have the same timestamps in the old and new part),
but it breaks if the new part (which follows the old part in the
physical file) is has a completely different start time in the timeline.
Essentially, the trick used to time subtitles correctly is incompatible
with the way we cache subtitles (to make them survive seeks).
The simple solution is just clearing the cached subtitles when crossing
chapter boundaries.
See #2609:
"When eof is reached it would be shown on the OSD and in the console.
Next try seeking to the middle. Seeking to the middle of the file will
only result in the OSD message being updated. Lua seems to fail to
observe the change in the property until the video is unpaused."
The demuxer infrastructure was originally single-threaded. To make it
suitable for multithreading (specifically, demuxing and decoding on
separate threads), some sort of tripple-buffering was introduced. There
are separate "struct demuxer" allocations. The demuxer thread sets the
state on d_thread. If anything changes, the state is copied to d_buffer
(the copy is protected by a lock), and the decoder thread is notified.
Then the decoder thread copies the state from d_buffer to d_user (again
while holding a lock). This avoids the need for locking in the
demuxer/decoder code itself (only demux.c needs an internal, "invisible"
lock.)
Remove the streams/num_streams fields from this tripple-buffering
schema. Move them to the internal struct, and protect them with the
internal lock. Use accessors for read access outside of demux.c.
Other than replacing all field accesses with accessors, this separates
allocating and adding sh_streams. This is needed to avoid race
conditions. Before this change, this was awkwardly handled by first
initializing the sh_stream, and then sending a stream change event. Now
the stream is allocated, then initialized, and then declared as
immutable and added (at which point it becomes visible to the decoder
thread immediately).
This change is useful for PR #2626. And eventually, we should probably
get entirely of the tripple buffering, and this makes a nice first step.
The "script-binding" command is used by the Lua scripting wrapper to
register key bindings on the fly. It's also the only way to get fine-
grained information about key events (such as separate key up/down
events). This information is sent via a "key-binding" message when the
state of a key changes.
Extend it to send name of the mapped key itself. Previously, it was
assumed that the user just uses an unique identifier for the binding's
name, so it wasn't needed. With this change, a user can map exactly the
same command to multiple keys, which is useful especially with the next
commit.
Part of #2612.
This is for the sake of command.c and the "deinterlace" option/property.
Instead of forcing certain "better" defaults when inserting yadif,
change the actual "yadif" defaults.
I pondered not changing vf_yadif, and instead adding a trivial "yadif-
auto" wrapper filter, which would merely have different defaults. But
thinking about it, it doesn't make any sense for "deinterlace" to have
different defaults from vf_yadif, with vf_yadif having the "worse"
defaults. If someone wants the old behavior, the old behavior can be
forced in a backward and forward compatible way by setting the
suboptions.
Fixes#2539 (kind of).
MPlayer traditionally always used the display aspect ratio, e.g. 16:9,
while FFmpeg uses the sample (aka pixel) aspect ratio.
Both have a bunch of advantages and disadvantages. Actually, it seems
using sample aspect ratio is generally nicer. The main reason for the
change is making mpv closer to how FFmpeg works in order to make life
easier. It's also nice that everything uses integer fractions instead
of floats now (except --video-aspect option/property).
Note that there is at least 1 user-visible change: vf_dsize now does
not set the display size, only the display aspect ratio. This is
because the image_params d_w/d_h fields did not just set the display
aspect, but also the size (except in encoding mode).
Apparently, this was replaced by the SD_CTRL_SET_VIDEO_PARAMS set
dimensions. But I can't find out when this happened - possibly, these
fields were never used by sd_lavc.c, and only by the (long removed)
MPlayer dvdsub decoder.
Until now, feeding packets to the decoder in advance was done for text
subtitles only. This was possible because libass buffers all subtitle
data anyway (in ASS_Track). sd_lavc, responsible for bitmap subs, does
not do this. But it can buffer a small number of subtitle frames ahead.
Enable this.
Repurpose the sub_accept_packets_in_advance(). Instead of "can take all
packets" it means "can take 1 packet" now. (The old meaning is still
needed locally in dec_sub.c; keep it there.) It asks the decoder whether
there is place for at least 1 subtitle packet. sd_lavc implements it and
returns true if its internal fixed-size subtitle queue still has a free
slot. (The implementation of this in dec_sub.c isn't entirely clean.
For one, decode_chain() ignores this mechanism, so it's implied that
bitmap subtitles do not use the subtitle filter chain in any advanced
way.)
Also fix 2 bugs in the sd_lavc queue handling. Subtitles must be checked
in reverse, because the first entry will often have endpts==NOPTS, which
would always match. alloc_sub() must cycle the queue buffer, because it
reuses memory allocations (like sub.imgs) by design.
Helps with files that have occasional broken timestamps. For larger
discontinuities, e.g. caused by actual timestamp resets, we still want
to realign audio.
(I guess in general, this should be removed and replaced by a more
general resync-on-desync logic, but not now.)
This makes no sense, because the client is obligated to react to this
event.
This also happens to fix a deadlock with JSON IPC clients sending
"disable_event all", because MPV_EVENT_SHUTDOWN was used to stop the
thread driving the socket connection (fixes#2558).
Requested. Don't overwrite permanent OSD text set with e.g. --osd-msg1.
Instead, append the OSD message to it (on the next line).
Note that with --osd-msg1, seeking will still overwrite the OSD with the
playback status for a while. If you do not want this, use --osd-msg3
--osd-level=3 instead.
At least I hope so.
Deriving the duration from the pts was not really correct. It doesn't
include speed adjustments, and becomes completely wrong of the user e.g.
changes the playback speed by a huge amount. Pass through the accurate
duration value by adding a new vo_frame field.
The value for vsync_offset was not correct either. We don't need the
error for the next frame, but the error for the current one. This wasn't
noticed because it makes no difference in symmetric cases, like 24 fps
on 60 Hz.
I'm still not entirely confident in the correctness of this, but it sure
is an improvement.
Also, remove the MP_STATS() calls - they're not really useful to debug
anything anymore.
This was just converting back and forth between int64_t/microseconds and
double/seconds. Remove this stupidity. The pts/duration fields are still
in microseconds, but they have no meaning in the display-sync case (also
drop printing the pts field from opengl/video.c - it's always 0).
Instead of periodically trying to enable it again. There are two cases
that can happen:
1. A random discontinuity messed everything up,
2. Things are just broken and will desync all the time
Until now, it tried to deal with case 1 - but maybe this is really rare,
and we don't really need to care about it. On the other hand, case 2 is
kind of hard to diagnose if the user doesn't use the terminal.
Seeking will reenable display-sync, so you can fix playback if case 1
happens, but still get predictable behavior in case 2.
This is simply the average refresh rate. Including "bad" samples is
actually an advantage, because the property exists only for
informational purposes, and will reflect problems such as the driver
skipping a vsync.
Also export the standard deviation of the vsync frame duration
(normalized to the range 0-1) as vsync-jitter property.
This was used with --no-sub-ass (aka --no-ass). This option (which is
not yet removed) strips all styling from the subtitles, and renders them
as plaintext only. For some reason, it originally seemed convenient to
reuse all the OSD text rendering code (osd_libass.c). While this was
indeed simple, it had a bad influence on the rest of the code. For
example, it had to decide whether to go through the OSD code path, or
the proper subtitle renderer in sd_ass.c.
Kill the OSD subtitle renderer. Reimplement --no-sub-ass and also
"secondary" subtitles in sd_ass.c. fill_plaintext() contains some rather
minor code duplication with osd_libass.c for setting up a dummy
ASS_Event and escaping the stripped text. Since sd_ass.c already has to
handle "normal" text subtitles, and has code for stripping ASS tags,
this remains all relatively simple.
Remove all the unnecessary crap from the rest of the code.
Use the demux_set_ts_offset() added in the previous commit to base each
timeline segment to use timestamps according to its relative position
within the overall timeline. As a consequence we don't need to care
about these timestamps anymore, and everything becomes simpler.
(Another minor but delicious nugget of sanity.)
Most of this is explained in the DOCS additions.
This gives us slightly more sanity, because there is less interaction
between the various parts. The goal is getting rid of the video_offset
entirely.
The simplification extends to the user API. In particular, we don't need
to fix missing parts in the API, such as the lack for a seek command
that seeks relatively to the start time. All these things are now
transparent.
(If someone really wants to know the real timestamps/start time, new
properties would have to be added.)
This adds support for the progress indicator taskbar extension
that was introduced with Windows 7 and Windows Server 2008 R2.
I don’t like this solution because it keeps its own state and
introduces another VOCTRL, but I couldn’t come up with anything
less messy.
closes#2399
If the player sends a frame with duration==0 to the VO, it can trivially
underrun. Don't panic, but keep the correct time.
Also, returning the absolute time from vo_get_next_frame_start_time()
just to turn it into a float with relative time was silly. Rename it and
make it return what the caller needs.
80ms allowable desync was a bit too much. It'd allow for a range of
160ms, which everyone can notice. It might also be a bother to apply
compensation resampling speed for that long.
We always let audio slowly desync until a threshold is reached, and then
pushed it back by applying a maximum compensation speed. Refine what
comes afterwards: instead of playing with the nominal video speed, use
the actual required audio speed for keeping sync as measured by the A/V
difference. (The "actual" speed is the ideal speed with A/V differences
added.)
Although this works in theory, it's somewhat questionable how much this
works in practice. The ideal time value is actually not exact, but is
the time at which the frame is scheduled (could be compensated by using
the time_left calculations in handle_display_sync_frame()). It doesn't
account for speed changes or catastrophic discontinuities. It uses only
10 past frames.
As long as it's within the desync tolerance, do not change the audio
speed at all for resampling. This reduces speed changes which might be
caused by jittering timestamps and similar cases.
(While in theory you could just not care and change speed every single
frame, I'm afraid that such changes could possibly cause audio
artifacts. So better just avoid it in the first place.)
This is very "illustrative", unlike the video-speed-correction
property, and thus useful. It can also be used to observe scheduling
errors, which are not detected by the core. (These happen due to
rounding errors; possibly not evne our fault, but coming from
files with rounded timestamps and so on.)
Instead of looking at the current frame duration for the intended
speedup, look at all past frames, and find a good average speed. This
ties in with not wanting to average _all_ frame durations, which
doesn't make sense in VFR situations.
This is currently done in the most naive way possible, but already sort
of works for VFR which switches between frame durations that are
integer multiples of a base rate. Certainly more improvements could
be made, such as trying to adjust directly on FPS changes, instead of
averaging everything, but for now this is not needed at all.
Helps somewhat with muxer-rounded timestamps.
There is some danger that this introduces a timestamp drift. But since
they are averaged values (unlike as when using an incorrect container
framerate hint), any potential drift shouldn't be too brutal, or
compensate itself soon. So I won't bother yet with comparing the results
with the real timestamp, unless we run into actual problems.
Of course we still prefer potentially real timestamps over the
approximated ones. But unless the timestamps match the container FPS,
we can't know whether they are (no, checking whether the they have
microsecond components would be cheating). Perhaps in future, we could
let the demuxer export the timebase - if the timebase is not 1000 (or
divisible by it), we know that millisecond-rounded timestamps won't
happen.
Get rid of get_past_frame_durations(), which was a bit too messy. Add
a past_frames array, which contains the same information in a more
reasonable way. This also means that we can get the exact current and
past frame durations without going through awful stuff. (The main
problem is that vo_pts_history contains future frames as well, which is
needed for frame backstepping etc., but gets in the way here.)
Also disable the automatic disabling of display-sync if the frame
duration changes, and extend the frame durations allowed for display
sync. To allow arbitrarily high durations, vo.c needs to be changed
to pause and potentially redraw OSD while showing a single frame, so
they're still limited.
In an attempt to deal with VFR, calculate the overall speed using the
average FPS. The frame scheduling itself does not use the average FPS,
but the duration of the current frame. This does not work too well,
but provides a good base for further improvements.
Where this commit actually helps a lot is dealing with rounded
timestamps, e.g. if the container framerate is wrong or unknown, or
if the muxer wrote incorrectly rounded timestamps. While the rounding
errors apparently can't be get rid of completely in the general case,
this is still much better than e.g. disabling display-sync completely
just because some frame durations go out of bounds.
We need a frame duration even on start, because the number of vsyncs
the frame is shown is predetermined. (vo_opengl actually makes use of
this property in certain cases.)
"Missed" implies the frame was dropped, but what really happens is that
the following frame will be shown later than intended (due to the
current frame skipping a vsync).
(As of this commit, this property is still inactive and always
returns 0. See git blame for details.)
When the audio format is not known yet and the audio chain is still
initializing, filter reinit will fail. Normally, attempts to
reinitialize filters at this stage should be rare (e.g. user commands
editing the filter chain). But it sometimes happened with track
switching in combination with the video code calling
update_playback_speed() at arbitrary times.
Get rid of the message by not trying to change the filters for the sake
of playback speed update while decoding is still being initialized.
Has the same function as setting the option.
This commit changes the property in a bunch of other ways. For example
if the VO is not created, it will return the option value.
This check disables the display-sync resample method. If the filters
convert PCM to AC3, we can still insert a filter to change speed. This
is because filters are inserted at the beginning of the filter chain.
Actually, it didn't really require that before (most work was avoided),
but some bits had to be run anyway. Separate the speed change into a
light-weight function, which merely updates already created filters, and
a heavy-weight one which messes with filter insertion.
This also happens to fix the case where the filters would "forget" the
current speed (force resampling, change speed, hit a volume control to
force af_volume insertion - it will reset speed and desync).
Since we now always run the light-weight function, remove the
af_scaletempo verbose message that is printed on speed setting. Other
than that, all setters are cheap.
Move it (in a cosmetic sense), and also move its invocation to below all
the video handling.
All other changes remain cosmetic, including moving the framedrop
calculation code, and getting rid of the video_speed_correction
variable.
We still have a sample-based buffer between filters and audio outputs.
In order to avoid cutting frames into half (which can upset receivers),
we strictly need to align the boundaries on which we cut the audio.
Update msg.c state immediately if a terminal or logging setting is set.
Until now, this was delayed until mp[v]_initialize() was called. When
using the client API, you could easily miss logged error messages, even
when logging was initialized early on by calling
mpv_request_log_messages().
(Properties can't be used for this either, because properties do not
work before mpv_initialize().)
Discontinuities (like toggling fullscreen) can cause multiple frames to
be dropped in succession, which sounds very weird. It's better to drop
some video frames instead to compensate for larger desyncs.
We roughly base it on the maximum allowed speed changes (audio change is
"additional" to the video change to account for deviations when playing
at max. video speed change).
update_av_diff() works on the timestamps, while time_left is in real
time. When playing at not-1 speed, these are very different, and cause
the A/V difference to jitter. Fix this by scaling the expected A/V
desync to the correct range.
This didn't show up with cases where the frame pattern has a cycle of 1
or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show
up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default
settings.)
In this case, we must not add the timing error of the next frame to the
A/V difference estimation of the current frame. Use the previous timing
error instead.
This is another bug resulting from the confusion about whether we
calculate parameters for the currently playing frame, or the one we're
about to queue.
Commit a1315c76 broke this slightly. Frame drops got counted multiple
times, and also vo.c was actually trying to "render" the dropped frame
over and over again (normally not a problem, since frames are always
queued "tightly" in display-sync mode, but could have caused 100% CPU
usage in some rare corner cases).
Do not repeat already dropped frames, but still treat new frames with
num_vsyncs==0 as dropped frames. Also, strictly count dropped frames in
the VO. This means we don't count "soft" dropped frames anymore (frames
that are shown, but for fewer vsyncs than intended). This will be
adjusted in the next commit.
Bump it to 80, and 2 vsyncs. This is another measure against vsync
jitter. Admittedly this is a bit simplistic (and we should probably
estimate a stable estimated vsync phase instead), but for now this will
do.
It's not needed, because the additional data is not appended, but is the
total size of the audio buffer. The maximum size is the static audio
drop size (or twice, if the audio is duplicated).
Calculate the A/V difference directly in the display sync code, instead
of the awkward current way, which reuses the fields for audio sync.
We still set time_frame, because it makes falling back to audio sync
somewhat smoother.
When dropping or repeating frames, we essentially influence when the
frame after the next frame will be shown, not the next frame. This led
to dropping/repeating frames 2 times, because the A/V difference had a
delay of one frame. Compensate it with the expected value.
This is all kinds of stupid - update_avsync_after_frame() will multiply
this value with the speed at a later point, and we only update this
field for this function. (This should be refactored.)
This makes the bitrate properties unavailable, instead of
returning 0 when:
1. No track is selected, or
2. Not enough packets have been read to have a bitrate estimate yet
Some mkv files can have this. The chapter times are still timestamps
(and thus not affected by the start time), but it misplaces the OSD
chapter ticks.
Apparently this function caused weird problems to me. I have no idea
why. The usage of the function looks perfectly fine to me, and even
rounding issues can be excluded. In any case, getting rid of this solved
my problem, and makes the code actually more readable.
Let's hope this doesn't confuse client API users too much. It's still
the best solution to get rid of corner cases where it actually return
the wrong timestamp on start, and then suddenly jump.
This adjustment is supposed to improve the audio speed calculation in
case of unexpected desync. The flipped sign made it actually worse,
although the total impact of this bug was very minor.
