audio: make buffer size configurable

Really only for testing.
malloc+memset(0) to calloc

Signed-off-by: wm4 <wm4@nowhere>
manpage: fix sub_add description

It is in fact selected. The manpage wasn't updated when this was
changed.
audio: fix random crashes on uninit

I added this non-sense earlier this day. Oops.
win32: add tmpfile() replacement

The Windows version of tmpfile is actually pretty broken. It tries to
create the file in the root directory of the current drive, which means
on Vista and up, it normally fails due to insufficient permissions.
Replace it with a version that uses GetTempPath.

Also remove the Windows-specific note about automatic deletion of the
cache file. FILE_FLAG_DELETE_ON_CLOSE is available in NT, and it should
be pretty reliable.
player: check for null in resume config search

Signed-off-by: wm4 <wm4@nowhere>

CC: @mpv-player/stable
audio: correctly initialize output buffer

Just like the previous commit, this takes care of fallout from commit
7ab228, which fixed a bug, but introduced some new ones.

CC: @mpv-player/stable
af_hrtf: cosmetics: reindent misaligned code block
af_hrtf: request required samplerate, instead of erroring out

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.
vo_sdl: fix typo
audio/out: make EOF handling properly event-based

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)
sub: fix possible deadlock with --no-sub-ass and similar

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.
audio/out: remove old things

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.
mplayer-input.conf: fix broken binding

"#" 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.
ao_oss: minor simplification

Equivalent code.
ao_pcm: minor simplification
ao_sndio: fix a comment

Whether this code was written with the correct assumptions in mind, I
don't know.
audio/out: always round get_space on period size

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.
audio/out/push: redo audio waiting

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.
audio: fix initial sync with huge AO buffer

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.