Apparently this can "sometimes" return an error. In my opinion, this
should never return an error: neither the semantics of the function,
nor the ALSA documentation or ALSA sample code seem to indicate that
a failure is to be expected. I'm not perfectly sure about this though
(I blame ALSA being a weird, big, underdocumented API).
Since it causes problems for some users, and since there is really no
reason why we should abort on such an error, turn it into a warning.
Fixes#1231.
Since the list associated with --audio-device is supposed to enable
simple user-selection, it doesn't make much sense to include overly
special things like ao_pcm or ao_null in the list. Specifically,
ao_pcm is harmful, because it will just dump all audio to a file
named audiodump.wav in the current working directory. The user can't
choose the filename (it can be customized, but not through this
option), and the working directory might be essentially random,
especially if this is used from a GUI.
Exclude "strange" entries. We reuse the fact that there's already a
simple list ordered by auto-probe priority in order to avoid having to
add an additional flag. This is also why coreaudio_exclusive was moved
above ao_null: ao_null ends auto-probing and marks the start of
"special" outputs, which don't show up on the device, but we want
coreaudio_exclusive to be selectable (I think).
Move it above ao_null, so that it can be selected during auto-probing
(even if it's only last). I see no reason why it should not be included,
and it makes the following commit slightly more elegant. (See
explanations there.)
Especially with other components (libavcodec, OSX stuff), the thread
list can get quite populated. Setting the thread name helps when
debugging.
Since this is not portable, we check the OS variants in waf configure.
old-configure just gets a special-case for glibc, since doing a full
check here would probably be a waste of effort.
While conceptually this sink stuff in PulseAudio does just the right
thing, actually listing the sinks is unbelievable complicated. Not only
is the idea that listing them should happen asynchronously completely
bullshit (who the fuck runs the PulseAudio server on a separate
computer), but the way this is done is full of bullshit too. Why
separate callbacks for each device? Why this obtuse mainloop shit?
Especially the mainloop shit makes it actively worse than doing things
manually with pthread primitives, and the reason for that (different
mainloop implementations for GUIs?) is laughable too. It's like they
chose the most complicated API possible just because they attempted
to "abstract" basic mechanisms in order to handle "everything". While
I don't claim to design the best APIs, this API is fucking terrible
without any excuse. (End of rant.)
All the dumb crap in pa_init_boilerplate() is needed to talk to the
audio server at all. Might also fix some subtle bugs in the init code
(which is strange, because the original file was contributed by the
devil himself).
The one in msg.c was mistakenly removed with commit e99a37f6.
I didn't actually test the change in ao_sndio.c (but obviously "ap"
shouldn't be static).
Don't wait after the audio thread has pushed the remaining audio to the
AO. Avoids hard hangs if the heuristic fails completely (could still
happen if get_delay returns absurd values).
CC: @mpv-player/stable
Since the internal AO driver API has no proper way to determine EOF, we
need to guess by querying get_delay. But some AOs (e.g. ao_pulse with
no-latency-hacks set) may never reach 0, maybe because they naively add
the latency to the buffer level. In this case our heuristic can break.
Fix by always using the delay to estimate the EOF time. It's not even
that important - it's mostly used to avoid blocking draining. So this
should be ok.
CC: @mpv-player/stable (maybe)
Unfortunately, ALSA is particularly bad with this, because mpv has to
add all sorts of magic crap to the device name to make things work. The
device selection overrides this, so explicitly selecting devices will
most likely break your audio. This has yet to be solved.
This function is available starting with PulseAudio 2.0, while we only
require 1.0. This broke compilation on Ubuntu 12.04.5 LTS.
Use our own function to calculate the buffer size, which is actually
simpler and needs slightly less code.
Hopefully fixes#1154.
CC: @mpv-player/stable
It was more complicated than it had to be: the audio thread already
determines whether audio has ended, so we can use that. Remove the
separate logic for draining.
Commit 957097 attempted to use PA_STREAM_FAIL_ON_SUSPEND to make
ao_pulse exit if the stream was started suspended.
Unfortunately, PA_STREAM_FAIL_ON_SUSPEND is active even during playback.
If you pause mpv, pulseaudio will close the actual audio device after a
while (or something like this), and unpausing won't work. Instead, it
will spam "Entity killed" error messages.
Undo this change and check for suspended audio manually during init.
CC: @mpv-player/stable
Sometimes, ao_pulse starts in suspended mode, which means playback is
essentially paused in pulseaudio. This gives the impression that mpv is
hanging, since it times video against the audio playback progress, and
audio never makes progress in this state.
I'm not sure if this will help - possibly it does with mixed
pulseaudio/alsa setups. However, if the alsa setup has the pulseaudio
plugin, alsa will hang too. But there's still a chance we get less
blame for pulseaudio messes.
libsndio has absolutely no mechanism to discard already written audio
(other than SIGKILLing the sound server). sio_stop() will always block
until all audio is played. This is a legitimate design bug.
In theory, we could just not stop it at all, so if the player is e.g.
paused, the remaining audio would be played. When resuming, we would
have to do something to ensure get_delay() returns the right value. But
I couldn't get it to work in all cases.
get_delay needs to report the current audio buffer status. It's
important for A/V sync that this information is current, but functions
which update it were called on play() or get_space() calls only.
This was in bytes, but it's more convenient to use samples (or frames;
in any case the smallest unit of audio that includes all channels).
Remove the ao->bps line too; it will be set after init() returns.
Digital pass-through was probably broken. Possibly fix it (no way to
test). This also should make the logic slightly saner.
Fortunately, it's unlikely that anyone who uses OSS has a spdif setup.
Commit 5b5a3d0c broke this. The really funny thing is that this code was
actually always under "#if BYTE_ORDER == BIG_ENDIAN". The breaking
commit just edited this code slightly, but it must have failed to
compile on big endian long before (since over 1 year ago, commit d3fb58).
Should be able to pass-through AC3, DTS, and others.
It seems PulseAudio wants players to fallback to PCM on certain events
signaled by the server, but we don't implement that. There's not much
documentation available anyway.
