It's useless, and creates a bogus warning in subprocess-posix.c.
Since I don't know which compilers might have it by default, just change
it to -Wno-redundant-decls.
mp_find_config_file() will print the filename lookup and its result in
verbose mode. This is wanted, but gets inconvenient when it is done for
every playlist entry (for resuming).
Lookup the watch_later subdir only once and cache the result instead.
This drops the logic for loading the resume file from other locations,
which should generally be unnecessary, though might lead to confusion if
the user has mixed old and new config paths (which the user shouldn't).
Also add a mp_find_user_config_file() function for a more
straightforward and reliable way to get actual local configpaths,
instead of possibly global and unwritable locations.
Also, for symmetry, check the resume option in mp_load_playback_resume()
just like mp_check_playlist_resume() does.
Share most of the implementation of config file lookup between
mp_find_all_config_files() and mp_find_config_file().
Also move the check for config path overrides to mp_get_platform_path()
directly. From the point of view of config file lookup, this is a bit
stupid, but on the other hand increases consistency, as user path
resolution exposes the mp_get_platform_path() functionality directly to
the user.
Now among other things panscan can be changed during playback.
Unfortunately, it flickers. The issue is that reconfig() clears the
framebuffer. Removing the clearing shows that the "unused" parts of
the picture are not cleared - even though OSD could render there. As
such, this is a separate issue.
When running with --panscan=1, this could crash - because the current
frame was reduced in size each time the image was redrawn, which would
result in a failed assertion the second time it's drawn.
This creates the window before the first file is loaded. This was
requested a bunch of times, but on the other hand a change to make this
behavior the default was reverted some time ago, because other users
hated it.
Drop mp_chmap_diff() (which is unused too now), and implement
mp_chmap_diffn() in a slightly simpler way. (Too bad there is no
standard function for counting set bits.)
Instead of somehow having 4 different cases with each their own weight,
do it with a single function that decides which channel layout is the
better fallback.
This is simpler, and also introduces new (fixed) semantics. The new test
added to test/chmap_sel.c actually works now. This is a mixed case with
no perfect upmix or downmix, but the better choice is the one which
loses the least channels from the original layout.
One test also changes. If the input is 7.1(wide-side), and the available
layouts are 7.1 and 5.1(side), the latter is now chosen instead of the
former. This makes sense: both layouts contain 6 out of 8 channels from
the original layout, but the 5.1(side) one is smaller. This follows the
general logic. The 7.1 layout has FLC/RLC speakers instead of BL/BR,
and judging by the names, "front left center" is completely different
from "back left". If these should be exchangeable, a separate exception
would have to be added.
Reuse MP_SPEAKER_ID_NA for this. If all mp_chmap entries are set to NA,
the channel layout has special "unknown channel layout" semantics, which
are used to deal with some corner cases.
The client API (libmpv) and encoding (--o) have slightly different
defaults from the command line player. Instead of doing a bunch of calls
to set the options explicitly, use profiles. This is simpler and has the
advantage that they can be listed on command line (instead of possibly
forcing the user to find and read the code to know all the details).
Sometimes, ALSA will return channel layouts with padded channels (NA
speakers). Use them instead of failing.
This still includes the old "braindeath" code to retry with a layout
without NA channels. This might be helpful for performance, and also the
padded channel layout string looks confusing.
To be fair, I have not encountered a case yet which would really need
this, and for which the old "braindeath" code did not fix it.
volatile barely means anything.
The polling is kind of bad too, but relatively harmless as device
opening/closing is a rare event, and the format change is not expected
to take long.
Remove the pointless talloc call too (must have been a leftover
from previous refactoring).
No reason to keep them separate. It's an artifact from the old
ao_coreaudio.c, which kept usage of two different APIs in the same file.
Removes a forward reference too.
This should fix some crashes due to dangling pointers.
The problem was that with_cocoa_lock_on_main_thread() is asynchronous.
It will not wait until it is finished. In the uninit case, this means
the VO could be deallocated and destroyed while cocoa was still running
uninit code.
So simply wait until it is done by using dispatch_sync(). There were
concerns that this could introduce a deadlock by the main thread trying
to wait for something on the VO thread. But from what I can see, this
never happens, and even if it does, it would crash anyway since the VO
is already gone.
One remaining worry is the video_resize_redraw_callback. From what I can
see, it still can mess things up, and will need a more elaborate fix.
Instead of trying to use af_format_conversion_score() (which tries to be
all kinds of clever), just compare the raw bits as a quality measure. Do
this because otherwise, weird formats like padded 24 bit formats will be
excluded, even though they might be the highest precision formats for
some hardware.
This means that for now, the user would have to check whether the format
is usable at all before calling ca_asbd_is_better(). But since this is
currently only used for ao_coreaudio.c and for the physical format, it
doesn't matter.
If coreaudio-exclusive should get PCM support, the best would be to
revert this change, and to add support for 24 bit formats directly.
Some time ago, a mechanism was added for automatically removing PCM-only
filters if the input format is spdif.
This could cause an infinite loop if the AO did not support spdif, but
was falling back to some PCM format. Then this code tried to remove the
last filter, which is a dummy filter for receiving and queuing filter
output. af_remove() simply fails gracefully in this case, so this
happens over and over again.
Fix by explicitly checking whether the filter to remove is a dummy
filter. (af_remove() also fails only if the dummy filters are attempted
to be removed - checking this directly is simpler.)
Move all of the channel map retrieval/negotiation code to a separate
file. This will (probably) be helpful when extending
ao_coreaudio_exclusive.c.
Nothing else changes, other than some minor cosmetics and renaming,
and changing some details for decoupling it from the ao_coreaudio.c
internals.
