Replace select() usage with poll() (and reduce code duplication).
Also, while we're at it, drop --disable-audio-select, since it has the
wrong name anyway. And I have doubts that this is needed anywhere. If
it is, it should probably fallback to doing the right thing by default,
instead of requiring the user to do it manually. Since nobody has done
that yet, and since this configure option has been part of MPlayer ever
since ao_oss was added, it's probably safe to say it's not needed.
The '#ifdef SNDCTL_DSP_GETOSPACE' was pointless, since it's already used
unconditionally in another place.
1. Separate buffer and temporary file handling from the vo to make maintenance
and reading code easier
2. Skip resizing as much as possible if back buffer is still busy.
3. Detach and mark osd buffers for deletion if we want to redraw them and they
are still busy. This could be a possible case for the video buffers as
well. Maybe better than double buffering.
All the above steps made it possible to have resizing without any artifacts
even for subtitles. Also fixes dozen of bugs only I knew, like broken subtitles
for rgb565 buffers. I can now sleep at night again.
OSX is POSIX conformant, but it's a sad joke: it provides the
<semaphore.h> prototype as the standard demands, but they're empty
wrappers, and all functions just return ENOSYS.
Emulate them similar to how osdep/io.h emulate filesystem functions on
Windows. By including the header, working sem_* functions become
available.
To make it async-signal safe, use a pipe for wakeup (write() is AS-safe,
but mutexes can't be). Actually I'm not sure anymore if we really need
AS-safety, but for now the emulation can do it.
On Linux, the system provides a far more efficient and robust
implementation. We definitely want to avoid using the emulation if
possible, so this code is active on OSX only. For convenience we always
build the source file though, even if the implementation is disabled and
no actual code is generated.
(Linux provides working semaphores, but is formally not POSIX
conformant. On OSX it's the opposite. Is POSIX a complete joke?)
I'm not quite sure what we should actually do (maybe read input
commands?), but interpreting input as terminal key sequences is
definitely weird. So just do nothing.
Do terminal input with a thread, instead of using the central select()
loop. This also changes some details how SIGTERM is handled.
Part of my crusade against mp_input_add_fd().
An attempt at fixing #1070. Apparently something goes wrong if the
video size is equal to the screen size. Since the window decorations
add to the window size, it must actually be larger than the screen.
Actually I don't know what exactly is going wrong, but since this
commit also slightly improves the behavior otherwise, it's a win
anyway.
Try to keep the window size strictly below screen size, even accounting
for window decorations. Size it down and center the window so that it
fits (by either touching the left/right or top/bottom screen borders).
I haven't found any information on what is the maximum allowed size and
position of a window so that it doesn't collide with the task bar, so
assume that we can use the entire screen, minus 1 pixel to avoid
triggering fullscreen semantics (if that is even possible).
The purpose is making accessing the current playlist entry saner when
commands are executed during initialization, termination, or after
playlist navigation commands.
For example, the "playlist_remove current" command will invalidate
playlist->current - but some things still access the playlist entry even
on uninit. Until now, checking stop_play implicitly took care of it, so
it worked, but it was still messy.
Introduce the mpctx->playing field, which points to the current playlist
entry, even if the entry was removed and/or the playlist's current entry
was moved (e.g. due to playlist navigation).
To handle legacy commands, string replacement is used; the modified
string is returned by parse_cmd_str(), but it also frees all temporary
memory, which includes the replaced string.
Closes#1075.
Continues commit 348dfd93. Replace other places where input was manually
fetched with common code.
demux_was_interrupted() was a weird function; I'm not entirely sure
about its original purpose, but now we can just replace it with simpler
code as well. One difference is that we always look at the command
queue, rather than just when cache initialization failed. Also, instead
of discarding all but quit/playlist commands (aka abort command), run
all commands. This could possibly lead to unwanted side-effects, like
just ignoring commands that have no effect (consider pressing 'f' for
fullscreen right on start: since the window is not created yet, it would
get discarded). But playlist navigation still works as intended, and
some if not all these problems already existed before that in some
forms, so it should be ok.
reinit_window_state() will set VO_EVENT_RESIZE when it runs, so we
don't need to set it manually depending on the VOCTRL.
Probably avoids duplicated resize events. I don't expect this actually
fixes anything, but might help spotting other bugs easier (if there
are any).
This makes the player wait until each script is loaded. Do this to give
the script a chance to setup all its event handlers. It might also be
useful to allow a script to change options that matter for playback.
While waiting for a script to be loaded, the player actually accepts
input. This is needed because the scripts can execute player commands
anyway while they are being "loaded". The player won't react to most
commands though: it can't quit or navigate the playlist in this state.
For deciding whether a script is finally loaded, we use a cheap hack: if
mpv_wait_event() is called, it's considered loaded. Let's hope this is
good enough. I think it's better than introducing explicit API for this.
Although I'm sure this will turn out as too simplistic some time in the
future, the same would probably happen with a more explicit API.
Expose the central event handling functions explicitly, so that other
parts of the player can use them.
No functional changes. Preparation for the next commit.
This was kept in the codebase because it is slightly faster than --vo=opengl
on really old Intel cards (from the GMA era). Time to kill it, and let it rest.
Fixes#1061
With e.g --start=-3 --audio-buffer=10 the decoder entered EOF state
before the initial sync was finished, entered STATUS_EOF, and just
started playing audio from a random position.
This doesn't handle seeking outside of the file, which is a different
case. E.g. --start=30:00 with audio and video enabled in a file shorter
than 30:00 will play a random last part of audio. This could perhaps be
fixed by using the hr-seek target for cutting audio, instead of the
video PTS, but that would be kind of intrusive, so don't do it for now.
The simpler solution, assuming audio EOF on video EOF, wouldn't work,
because we allow audio to start before video, or to last after video.
Improve the logic how the audio thread decides how to wait until the AO
is ready for new data. The previous commit makes some of this easier,
although it turned out that it wasn't required, and we still can handle
AOs with bad get_space implementation (although the new code prints an
error message, and it might fail in obscure situations).
The new code is pretty similar to the old one, and the main thing that
changes is that complicated conditions are tweaked. AO waiting is now
used better (mainly instead of max>0, r>0 is used). Whether to wakeup
is reevaluated every time, instead of somehow doing the wrong thing
and compensating for it with a flag.
This fixes the specific situation when the device buffer is full, and
we don't want to buffer more data. In the old code, this wasn't handled
correctly: the AO went to sleep forever, because it prevented proper
wakeup by the AO driver, and as consequence never asked the core for new
data. Commit 4fa3ffeb was a hack-fix against this, and now that we have
a proper solution, this hack is removed as well.
Also make the refill threshold consistent and always use 1/4 of the
buffer. (The threshold is used for situations when an AO doesn't
support proper waiting or chunked processing.)
This commit will probably cause a bunch of regressions again.
Round get_space() results in the same way play() rounds the input size.
Some audio APIs do this for various reasons.
This affects only "push" based AOs. Some of these need no change,
because they either do it already right (like ao_openal), or they seem
not to have any such requirements (like ao_pulse).
Needed for the following commit.
"#" starts a comment, so the # key needs to be handled specially.
MPlayer has the same issue, but its input.conf is wrong. Or at least I
think it's wrong; looking at the MPlayer code it's doubtful they somehow
special-case and handle this.
Remove the unnecessary indirection through ao fields.
Also fix the inverted result of AOCONTROL_HAS_TEMP_VOLUME. Hopefully the
change is equivalent. But actually, it looks like the old code did it
wrong.
This is a deadlock caused by a lock order issue: sub/osd.c locks the OSD
first, then the subtitle decoder lock. player/sub.c does the reverse.
Fix this by discussing away the requirement for locking (see below),
which allows us to drop the broken sub lock. sub_get_text() still
acquires and releases the sub decoder lock, but it's not held at the
same time as the OSD lock anymore, so it should be fine.
Originally, the sub lock was acquired because sub_get_text() returns a
pointer to a mutable string. We simply declare that it's ok to call it
unlocked, as long as only 1 thread accesses it, which works out fine in
this case.
With --gapless-audio=no, changing from one file to the next apparently
made it hang, until the player was woken up by unrelated events like
input. The reason was that the AO doesn't notify the player of EOF
properly. the played was querying ao_eof_reached(), and then just went
to sleep, without anything waking it up.
Make it event-based: the AO wakes up the playloop if the EOF state
changes.
We could have fixed this in a simpler way by synchronously draining the
AO in these cases. But I think proper event handling is preferable.
Fixes: #1069
CC: @mpv-player/stable (perhaps)
It seems hrtf works in 48khz only - and if that wasn't the input, the
filter just exited with an error. Make it request the 48khz instead. The
player will insert a resampling filter.
Not sure why it wasn't done like this in the first place.
