Move a condition somewhere else, which makes it conceptually simpler.
Also, the assignment to full_audio_buffers removed with this commit was
dead, and its value never used.
Fatal errors in the vidoe chain (such as failing to initialize the video
chain) disable video decoding. Restart the playloop, instead of just
continuing the current iteration.
The resulting behavior should be the same, but it gets rid of possible
corner cases.
Commit dc00b146, which disables polling by default, missed another
instance of polling: when the player pauses automatically on low cache.
This could lead to apparent freezes when playing network streams.
In my opinion this is not really necessary, since there's only a single
user of update_video(), but others reading this code would probably hate
me for using magic integer values instead of symbolic constants.
This should be a purely cosmetic commit; any changes in behavior are
bugs.
Instead of blocking on the demuxer when reading a packet, let packets be
read asynchronously. Basically, it polls whether a packet is available,
and if not, the playloop goes to sleep until the demuxer thread wakes it
up.
Note that the player will still block for I/O, because audio is still
read synchronously. It's much harder to do the same change for audio
(because of the design of the audio decoding path and especially
initialization), so audio will have to be done later.
Mouse cursor handling, --heartbeat-cmd, and OSD messages basically
relied on polling. For this reason, the playloop always used a small
timeout (not more than 500ms).
Fix these cases, and raise the timeout to 100 seconds. There is no
reason behind this number; for this specific purpose it's as close to
infinity as any other number.
On MS Windows, or if vo_sdl is used, the timeout remains very small.
In these cases the GUI code doesn't do proper event handling in the
first place, and fixing it requires much more effort.
getch2_poll() still does polling, because as far as I'm aware no event-
based way to detect this state change exists.
This adds a thread to the demuxer which reads packets asynchronously.
It will do so until a configurable minimum packet queue size is
reached. (See options.rst additions.)
For now, the thread is disabled by default. There are some corner cases
that have to be fixed, such as fixing cache behavior with webradios.
Note that most interaction with the demuxer is still blocking, so if
e.g. network dies, the player will still freeze. But this change will
make it possible to remove most causes for freezing.
Most of the new code in demux.c actually consists of weird caches to
compensate for thread-safety issues (with the previously single-threaded
design), or to avoid blocking by having to wait on the demuxer thread.
Most of the changes in the player are due to the fact that we must not
access the source stream directly. the demuxer thread already accesses
it, and the stream stuff is not thread-safe.
For timeline stuff (like ordered chapters), we enable the thread for the
current segment only. We also clear its packet queue on seek, so that
the remaining (unconsumed) readahead buffer doesn't waste memory.
Keep in mind that insane subtitles (such as ASS typesetting muxed into
mkv files) will practically disable the readahead, because the total
queue size is considered when checking whether the minimum queue size
was reached.
demux_seek() actually doesn't return seek success. Instead, it fails if
the demuxer is flagged as unseekable (but this is checked explicitly at
the beginning of this function), or if the seek target PTS is
MP_NOPTS_VALUE (which can never happen).
This should be unneeded, and the packet position is already sufficient
for this case.
Accessing the stream position directly is going to be a problem when the
stream is accessed from another thread later.
Let the VOs draw the OSD on their own, instead of making OSD drawing a
separate VO driver call. Further, let it be the VOs responsibility to
request subtitles with the correct PTS. We also basically allow the VO
to request OSD/subtitles at any time.
OSX changes untested.
Stop using it in most places, and prefer STREAM_CTRL_GET_SIZE. The
advantage is that always the correct size will be used. There can be no
doubt anymore whether the end_pos value is outdated (as it happens often
with files that are being downloaded).
Some streams still use end_pos. They don't change size, and it's easier
to emulate STREAM_CTRL_GET_SIZE using end_pos, instead of adding a
STREAM_CTRL_GET_SIZE implementation to these streams.
Make sure int64_t is always used for STREAM_CTRL_GET_SIZE (it was
uint64_t before).
Remove the seek flags mess, and replace them with a seekable flag. Every
stream must set it consistently now, and an assertion in stream.c checks
this. Don't distinguish between streams that can only be forward or
backwards seeked, since we have no such stream types.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
Cover art is treated like video, but is not really video. In one case,
the audio sync code was accidentally still active. Fixes cover art
playback with --ao=null. (This is due to ao_null's latency emulation.
Although it's not very clear whether that is actually correct...)
Some options change from percentages to number of kilobytes; there are
no cache options using percentages anymore.
Raise the default values. The cache is now 25000 kilobytes, although if
your connection is slow enough, the maximum is probably never reached.
(Although all the memory will still be used as seekback-cache.)
Remove the separate --audio-file-cache option, and use the cache default
settings for it.
Until recently, the VO was an unavoidable part of the seeking code path.
This was because vdpau deinterlacing could double the framerate, and hr-
seek and framestepping etc. all had to "see" the additional frames. But
we've removed the frame doubling from the vdpau VO and moved it into a
video filter (vf_vdpaupp), and there's no reason left why the VO should
participate in seeking.
Instead of queuing frames to the VO during seek and skipping them
afterwards, drop the frames early.
This actually might make seeking with vo_vdpau and software decoding
faster, although I haven't measured it.
Now we avoid calling update_video() twice on reconfig (once to check
whether there are still new frames, and again to actually do the
reconfig). Instead, we check whether there's still something going on
before calling update_video() at all, and depending on that
update_video() will be allowed to reconfig or not.
This will simplify some things later.
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
This wasn't really fine, and could (perhaps) cause weird corner cases on
reinit or when the player was paused.
Before eb9d20, video_left was also set to true if vo->frame_loaded was
set, and this variable basically indicated whether the previous
update_video() call was successful. This was overlooked when changing
everything. Simply always call update_video(), it should be equivalent.
Change how the video decoding loop works. The structure should now be a
bit easier to follow. The interactions on format changes are (probably)
simpler. This also aligns the decoding loop with future planned changes,
such as moving various things to separate threads.
This was part of osdep/threads.c out of laziness. But it doesn't contain
anything OS dependent. Note that the rest of threads.c actually isn't
all that OS dependent either (just some minor ifdeffery to work around
the lack of clock_gettime() on OSX).
For some reason, the buffered_audio variable was used to "cache" the
ao_get_delay() result. But I can't really see any reason why this should
be done, and it just seems to complicate everything.
One reason might be that the value should be checked only if the AO
buffers have been recently filled (as otherwise the delay could go low
and trigger an accidental EOF condition), but this didn't work anyway,
since buffered_audio is set from ao_get_delay() anyway at a later point
if it was unset. And in both cases, the value is used _after_ filling
the audio buffers anyway.
Simplify it. Also, move the audio EOF condition to a separate function.
(Note that ao_eof_reached() probably could/should whether the last
ao_play() call had AOPLAY_FINAL_CHUNK set to avoid accidental EOF on
underflows, but for now let's keep the code equivalent.)
This should probably be an AO function, but since the playloop still has
some strange stuff (using the buffered_audio variable instead of calling
ao_get_delay() directly), just leave it and make it more explicit.
This collects statistics and other things. The option dumps raw data
into a file. A script to visualize this data is included too.
Litter some of the player code with calls that generate these
statistics.
In general, this will be helpful to debug timing dependent issues, such
as A/V sync problems. Normally, one could argue that this is the task of
a real profiler, but then we'd have a hard time to include extra
information like audio/video PTS differences. We could also just
hardcode all statistics collection and processing in the player code,
but then we'd end up with something like mplayer's status line, which
was cluttered and required a centralized approach (i.e. getting the data
to the status line; so it was all in mplayer.c). Some players can
visualize such statistics on OSD, but that sounds even more complicated.
So the approach added with this commit sounds sensible.
The stats-conv.py script is rather primitive at the moment and its
output is semi-ugly. It uses matplotlib, so it could probably be
extended to do a lot, so it's not a dead-end.
The audio subsystem now wakes up the playback thread explicitly, and we
don't need this anymore.
It still could cause dropouts and such if there are bugs in the recently
introduced audio changes, so this is a thing to watch out for.
And slightly adjust the semantics of MPV_EVENT_PAUSE/MPV_EVENT_UNPAUSE.
The real pause state can now be queried with the "core-idle" property,
the user pause state with the "pause" property, whether the player is
paused due to cache with "paused-for-cache", and the keep open event can
be guessed with the "eof-reached" property.
This property is set to "yes" if playback was paused due to --keep-open.
The change notification might not always be perfect; maybe that should
be improved.
And consistently use MP_NOPTS_VALUE as error value for the users of this
function. This is better than using -1, especially because negative
values can be valid timestamps.
Instead of comparing the current chapter every time, set the playback
end timestamp to the chapter end. Likewise, don't execute an extra seek
for the start chapter.
Maybe we could also use the timeline facility to restrict playback to
the given chapter range, but this would be strange when using
--chapter=N to start playback at a given chapter. Then you couldn't seek
back, which is possibly not what the user wants.
Instead, always use the mpctx->chapters array. Before this commit, this
array was used only for ordered chapters and such, but now it's always
populated if there are chapters.
Stream-level chapters (like DVD etc.) did potentially not have
timestamps for each chapter, so STREAM_CTRL_SEEK_TO_CHAPTER and
STREAM_CTRL_GET_CURRENT_CHAPTER were needed to navigate chapters. We've
switched everything to use timestamps and that seems to work, so we can
simplify the code and remove this old mechanism.
This reverts commit 75dd3ec210.
This broke seeking with ordered chapters in some situations. While
the reverted commit was perfectly fine for playback of normal files,
it overlooked that in the ordered chapters case switching segments
actually reinitialized the audio chain completely, including the
decoder. And decoders still read packets on initialization. We can
restore the original commit as soon as decoders stop doing this.
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
The OSC used significant CPU time while the player was paused. It turned
out that the "tick" event sent during pause is the problem. The OSC
accesses the player core when receiving a tick event, which in turn will
cause the core to send another tick event, leading to infinite feedback.
Fix this by sending an idle tick only every 500ms. This is not very
proper, but the idea behind the tick event isn't very clean to begin
with (and the OSC should use timers instead).
Not sure about this... might redo.
At least this provides a case of a broadcasted event, which requires
per-event data allocation.
See github issue #576.
M_OPT_PARSE_ESCAPES was pretty stupid, and broke the (useful) assumption
that string variables contain exactly the same value as set by the
option. Simplify it, and move escape handling to the place where it's
used.
Escape handling itself is not terribly useful, but still allows useful
things like multiline custom OSD with "\n".
This is partial only, and it still accesses some MPContext internals.
Specifically, chapter and track lists are still read directly, and OSD
access is special-cased too.
The OSC seems to work fine, except using the fast-forward/backward
buttons. These buttons behave differently, because the OSC code had
certain assumptions how often its update code is called.
The Lua interface changes slightly.
Note that this has the odd property that Lua script and video start
at the same time, asynchronously. If this becomes an issue, explicit
synchronization could be added.
Add a client API, which is intended to be a stable API to get some rough
control over the player. Basically, it reflects what can be done with
input.conf commands or the old slavemode. It will replace the old
slavemode (and enable the implementation of a new slave protocol).
The code removed from handle_input_and_seek_coalesce() did two things:
1. If there's a queued seek, stop accepting non-seek commands, and delay
them to the next playloop iteration.
2. If a seek is executing (i.e. the seek was unqueued, and now it's
trying to decode and display the first video frame), stop accepting
seek commands (and in fact all commands that were queued after the
first seek command). This logic is disabled if seeking started longer
than 300ms ago. (To avoid starvation.)
I'm not sure why 1. would be needed. It's still possible that a command
immediately executed after a seek command sees a "seeking in progress"
state, because it affects queued seeks only, and not seeks in progress.
Drop this code, since it can easily lead to input starvation, and I'm
not aware of any disadvantages.
The logic in 2. is good to make seeking behave much better, as it
guarantees that the video display is updated frequently. Keep the core
idea, but implement it differently. Now this logic is applied to seeks
only. Commands after the seek can execute freely, and like with 1., I
don't see a reason why they couldn't. However, in some cases, seeks are
supposed to be executed instantly, so queue_seek() needs an additional
parameter to signal the need for immediate update.
One nice thing is that commands like sub_seek automatically profit from
the seek delay logic. On the other hand, hitting chapter seek multiple
times still does not update the video on chapter boundaries (as it
should be).
Note that the main goal of this commit is actually simplification of the
input processing logic and to allow all commands to be executed
immediately.
Do two things:
1. add locking to struct osd_state
2. make struct osd_state opaque
While 1. is somewhat simple, 2. is quite horrible. Lots of code accesses
lots of osd_state (and osd_object) members. To make sure everything is
accessed synchronously, I prefer making osd_state opaque, even if it
means adding pretty dumb accessors.
All of this is meant to allow running VO in their own threads.
Eventually, VOs will request OSD on their own, which means osd_state
will be accessed from foreign threads.
The terminal OSD code includes the handling of the terminal status line,
showing player OSD messages on the terminal, and showing subtitles on
terminal (the latter two only if there is no video window, or if
terminal OSD is forced).
This didn't handle some corner cases correctly. For example, showing an
OSD message on the terminal always cleared the previous line, even if
the line was an important message (or even just the command prompt, if
most other messages were silenced).
Attempt to handle this correctly by keeping track of how many lines the
terminal OSD currently consists of. Since there could be race conditions
with other messages being printed, implement this in msg.c. Now msg.c
expects that MSGL_STATUS messages rewrite the status line, so the caller
is forced to use a single mp_msg() call to set the status line.
Instead of littering print_status() all over the place, update the
status only once per playloop iteration in update_osd_msg(). In audio-
only mode, the status line might now be a little bit off, but it's
perhaps ok.
Print the status line only if it has changed, or if another message was
printed. This might help with extremely slow terminals, although in
audio+video mode, it'll still be updated very often (A-V sync display
changes on every frame).
Instead of hardcoding the terminal sequences, use
terminfo/termcap to get the sequences. Remove the --term-osd-esc option,
which allowed to override the hardcoded escapes - it's useless now.
The fallback for terminals with no escape sequences for moving the
cursor and clearing a line is removed. This somewhat breaks status line
display on these terminals, including the MS Windows console: instead of
querying the terminal size and clearing the line manually by padding the
output with spaces, the line is simply not cleared. I don't expect this
to be a problem on UNIX, and on MS Windows we could emulate escape
sequences. Note that terminal OSD (other than the status line) was
broken anyway on these terminals.
In osd.c, the function get_term_width() is not used anymore, so remove
it. To remind us that the MS Windows console apparently adds a line
break when writint the last column, adjust screen_width in terminal-
win.c accordingly.
Insane .ass subtitle scripts can cause severe slowdown (depending on the
speed of the machine, or the insanity of the script), so mention how to
test without subtitles. This is mainly to make the user aware that
subtitle rendering can be a problem. For longwinded explanation, there
isn't enough space.
This is relatively hacky, but it's Christmas, so it's ok. This does two
things: 1. allow selecting two subtitle tracks, and 2. include a hack
that renders the second subtitle always as toptitle. See manpage
additions how to use this.
For some reason, this checked whether there are external tracks at all
before doing any seeks. Possibly this was to avoid multiple
get_main_demux_pts() calls, but calling this multiple times shouldn't be
too bad.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
wm4
Tsukasa OMOTO
Martin Herkt
Sebastian Morr
xylosper