The buffer_size may be updated before the process callback is called for
the first time. Or, the connection graph could change, which changes the
latency of the pipeline after mpv's output. Ensure we keep on top of
these changes by registering callbacks to update our latency estimation.
It appears some device can be missing if we filter too many. In
particular, I've seen devices starting with "front" and "sysdefault"
being mapped to different hardware. I conclude that it's not sane trying
to present a nice device list to users in ALSA. It's fucked. (Although
kodi appears to attempt some intense "beautification" of the device
list, which includes parsing parameters from the device name and such.
Well, let's not.)
No other audio API requires such ridiculous acrobatics.
Apparently POLLERR can be set if poll is called while the device is in
the SND_PCM_STATE_PREPARED state. So assume that we can simply call
snd_pcm_status() to check whether the error is because the device went
away (i.e. we expect it to return ENODEV if this happened).
This avoids sporadic device lost warnings and AO reloads. The actual
device lost case is untested.
For example, previously, --audio-device='alsa/' would provide ao->device="" to
the alsa driver in spite of the fact that this is an already parsed option. To
avoid requiring a check of ao->device[0] in every driver, make sure this never
happens.
Fall back on PATH_DEV_DSP if nothing is set.
This mirrors the behaviour of --audio-device / --alsa-device.
There doesn't appear to be a general way to list devices with oss, so
--audio-device=help doesn't list oss devices except for the default one if the
file exists.
Previously --audio-device was ignored entirely by ao_oss.
fixes#4122
See: https://msdn.microsoft.com/en-us/library/windows/desktop/dd743946.aspx
Microsoft example code often uses a SAFE_RELEASE macro like the one in
the above link. This makes it easier to avoid errors when releasing COM
interfaces. It also reduces noise in COM-heavy code.
ao_wasapi.h also had a macro called SAFE_RELEASE, though unlike the
version above, its SAFE_RELEASE macro accepted a second parameter which
allowed it to destroy arbitrary objects other than just COM interfaces.
This renames ao_wasapi's SAFE_RELEASE to SAFE_DESTROY, which should more
accurately reflect what it does and prevent confusion with the Microsoft
version.
We log a large number of formats, but we rarely log the result of the
probing. Change this.
The logic in try_format_exclusive() changes slightly, but should be
equivalent. EXIT_ON_ERROR() checks for FAILED(), which should be
exclusive to SUCCEEDED().
Long planned. Leads to some sanity.
There still are some rather gross things. Especially g_groups is ugly,
and a hack that can hopefully be removed. (There is a plan for it, but
whether it's implemented depends on how much energy is left.)
Until now, this was only implemented for ao_alsa and AOs not using
push.c. ao_alsa.c relied on enabling funny underrun semantics for
avoiding resets on lower levels, while other AOs using push.c didn't do
anything.
Change this and at least make push.c copy silent data to the AO. This
still isn't perfect as keeping track of how much silence was played when
seems complex, so we don't do it. The consequence is that frame-stepping
will essentially randomize the A/V offset (it'll recover immediately
when unpausing, but still ugly). Also, in order to empty the currently
buffered audio on seeks etc., we still call ao_driver->reset and so on,
so the AO driver will still need to handle this specially.
The intent is to make behavior with ALSA less weird (for one we can
remove the code in ao_alsa.c that tries to trigger an initial
underflow). Also might help with #3754.
The "default" entry (which is and always was mpv/mplayer's default) does
not have a description set in the ALSA API. (While "sysdefault"
strangely has.)
Instead of an empty description, this should show something nice, so
reuse the ao.c code for naming default devices (see previous commit).
It's still a bit ugly that audio-device-list will have a default entry
for "Autoselect device" and "Default (alsa)", but then again we probably
want to allow the user to force ALSA (i.e. prevent fallbacks to other
AOs) just because ALSA is so flaky and makes this a legitimate feature.
This will make it easier for AOs to add explicit default device entries.
(See next commit.)
Hopefully this change doesn't lead accidentally to bogus "Default"
entries to appear, but then it can only happen if the device ID is
empty, which would mean the underlying audio API returned bogus entries.
Use the device name as fallback. This is ugly, but still better than
skipping the description entirely. This can be an issue on ALSA, where
the API can return entries without proper description.
This happens when ALSA gives us more channels than we asked for, for
whatever reasons. It looks like this wasn't handled correctly. The mpv
and ALSA channel counts could mismatch, which would lead to UB.
I couldn't actually trigger this case, though. I'm fairly sure that
drivers or plugins exist that do it anyway. (Inofficial ALSA motto: if
it can be broken, then why not break it?)
If the input is already mono or stereo, or if channel map selection
results in mono or stereo, then disable further use of the champ ALSA
API (or rather, stop trusting its results). Then we behave like a simple
application that only wants to output mono or stereo.
See #3045 and #2905. I couldn't actually test these cases, but this
commit is supposed to fix them.
set_chmap() skipped _setting_ the ALSA chmap if chmap use was requested
to be disabled by setting dev_chmap.num=0 by the caller, but it still
queried the current ALSA channel map. We don't trust it that much, so
disable that as well.
But we still query and log it, because that could be helpful for
debugging. Otherwise we could skip the entire set_chmap() call in these
cases.
If the "default" device refuses to be opened as spdif device (i.e. it
errors due to the AES0 etc. parameters), we were falling back to the
iec958 device. This is needed on some systems for smooth operation with
PCM vs. spdif.
Now change it to try "hdmi" before "iec958", which supposedly helps in
other situations.
Better suggestions welcome. Apparently kodi does this too, although I
didn't check directly.
The player tries to avoid splitting frames with spdif (sample alignment
stuff). This can in certain corner cases with certain drivers lead to
the situation that ao_get_space() returns a number higher than 0 and
lower than the audio frame size. The playloop will round this down to 0
bytes and do nothing, leading to a missed wakeup. This can lead to
underruns or playback completely getting stuck.
It can be reproduced by playing AC3 passthrough with no video and:
--ao=null --ao-null-buffer=0.256 --ao-null-outburst=6100
This commit attempts to fix it by allowing the playloop to write some
additional data (to get a complete frame), that will be buffered within
the AO ringbuffer even if the audio device doesn't want it.
We always want to use __declspec(selectany) to declare GUIDs, but
manually including <initguid.h> in every file that used GUIDs was
error-prone. Since all <initguid.h> does is define INITGUID and include
<guiddef.h>, we can remove all references to <initguid.h> and just
compile with -DINITGUID to get the same effect.
Also, this partially reverts 622bcb0 by re-adding libuuid.a to the
build, since apparently some GUIDs (such as GUID_NULL) are not declared
in the source file, even when INITGUID is set.
There were multiple values under M_OPT_EXIT (M_OPT_EXIT-n for n>=0).
Somehow M_OPT_EXIT-n either meant error code n (with n==0 no error?), or
the number of option valus consumed (0 or 1). The latter is MPlayer
legacy, which left it to the option type parsers to determine whether an
option took a value or not. All of this was changed in mpv, by requiring
the user to use explicit syntax ("--opt=val" instead of "-opt val").
In any case, the n value wasn't even used (anymore), so rip this all
out. Now M_OPT_EXIT-1 doesn't mean anything, and could be used by a new
error code.
Currently, calling mp_input_wakeup() will wake up the core thread (also
called the playloop). This seems odd, but currently the core indeed
calls mp_input_wait() when it has nothing more to do. It's done this way
because MPlayer used input_ctx as central "mainloop".
This is probably going to change. Remove direct calls to this function,
and replace it with mp_wakeup_core() calls. ao and vo are changed to use
opaque callbacks and not use input_ctx for this purpose. Other code
already uses opaque callbacks, or has legitimate reasons to use
input_ctx directly (such as sending actual user input).
The first one is printed even if the user disabled video (or there's no
video), so just remove it. The second one uses deprecated sub-option
syntax, so remove that as well.
And introduce a global option which does this. Or more precisely, this
deprecates the global wasapi and coreaudio options, and adds a new one
that merges their functionality. (Due to the way the sub-option
deprecation mechanism works, this is simpler.)
I decided that it's too much work to convert all the VO/AOs to the new
option system manually at once. So here's a shitty hack instead, which
achieves almost the same thing. (The only user-visible difference is
that e.g. --vo=name:help will list the sub-options normally, instead of
showing them as deprecation placeholders. Also, the sub-option parser
will verify each option normally, instead of deferring to the global
option parser.)
Another advantage is that once we drop the deprecated options,
converting the remaining things will be easier, because we obviously
don't need to add the compatibility hacks.
Using this mechanism is separate in the next commit to keep the diff
noise down.
Instead of requiring each VO or AO to manually add members to MPOpts and
the global option table, make it possible to register them automatically
via vo_driver/ao_driver.global_opts members. This avoids modifying
options.c/options.h every time, including having to duplicate the exact
ifdeffery used to enable a driver.
Normally I'd prefer a bunch of smaller functions with fewer parameters
over a single function with a lot of parameters. But future changes will
require messing with the parameters in a slightly more complex way, so a
combined function will be needed anyway. The now-unused "global"
parameter is required for later as well.
Positional parameters cause problems because they can be ambiguous with
flag options. If a flag option is removed or turned into a non-flag
option, it'll usually be interpreted as value for the first sub-option
(as positional parameter), resulting in very confusing error messages.
This changes it into a simple "option not found" error.
I don't expect that anyone really used positional parameters with --vo
or --ao. Although the docs for --ao=pulse seem to encourage positional
parameters for the host/sink options, which means it could possibly
annoy some PulseAudio users.
--vf and --af are still mostly used with positional parameters, so this
must be a configurable option in the option parser.
With the previous commit, ao_alsa.c now has 3 possible ways to pause
playback. Actually all 3 of them need get_delay() to fake its return
value, so don't duplicate that code.
Also much of the code looks a bit questionable when considering
inconsistent pause/resume calls from outside, so ignore redundant calls.
push.c does not handle this automatically, and AOs using push.c have to
handle it themselves. Also, ALSA is low-level enough that it needs
explicit support in user code. At least I haven't found any option that
does this.
We still can get away relatively cheaply by abusing underflow-handling
for this. ao_alsa.c already configures ALSA to handle underflows by
playing silence. So we purposely induce an underflow when opening the
device, as well as when pausing or resetting the device.
This introduces minor misbehavior: it doesn't account for the additional
delay the initial silence adds, unless the device has fully played the
fragment of silence when the player starts sending data to it. But
nobody cares.
Cheng Sun
wm4
Philip Sequeira
Kevin Mitchell
James Ross-Gowan
pavelxdd
Aman Gupta
Josh de Kock