When the audio thread fails to properly init, it signals failure
to the main thread, AND THEN starts to clean up. For this to work,
ao_init callback must not return until the thread's cleanup is finished.
This is correctly handled in the ao_uninit callback by waiting for
the thread to exit, so just call that to clean up the main thread.
I have no idea why I didn't do this in the first place.
Even with change notifications, there are still (rare) cases when the
feed thread gets AUDCLIENT_DEVICE_INVALIDATED. So handle failures in
thread_feed by requesting ao_reload.
on changes to PKEY_AudioEngine_DeviceFormat, device status, and default device.
call ao_reload directly in the change_notify "methods".
this requires keeping a device enumerator around for the duration of
execution, rather than just for initially querying devices
Before, failures, particularly in the thread loop init, could lead to a
bad state for the duration of mpvs execution. Make sure that
everything that was initialized gets properly and safely
uninitialized.
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.
There were subtle and minor race conditions in the pull.c code, and AOs
using it (jack, portaudio, sdl, wasapi). Attempt to remove these.
There was at least a race condition in the ao_reset() implementation:
mp_ring_reset() was called concurrently to the audio callback. While the
ringbuffer uses atomics to allow concurrent access, the reset function
wasn't concurrency-safe (and can't easily be made to).
Fix this by stopping the audio callback before doing a reset. After
that, we can do anything without needing synchronization. The callback
is resumed when resuming playback at a later point.
Don't call driver->pause, and make driver->resume and driver->reset
start/stop the audio callback. In the initial state, the audio callback
must be disabled.
JackAudio of course is different. Maybe there is no way to suspend the
audio callback without "disconnecting" it (what jack_deactivate() would
do), so I'm not trying my luck, and implemented a really bad hack doing
active waiting until we get the audio callback into a state where it
won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we
can be sure that the callback doesn't access the ringbuffer or anything
else anymore. Since both sched_yield() and pthread_yield() apparently
are not always available, use mp_sleep_us(1) to avoid burning CPU during
active waiting.
The ao_jack.c change also removes a race condition: apparently we didn't
initialize _all_ ao fields before starting the audio callback.
In ao_wasapi.c, I'm not sure whether reset really waits for the audio
callback to return. Kovensky says it's not guaranteed, so disable the
reset callback - for now the behavior of ao_wasapi.c is like with
ao_jack.c, and active waiting is used to deal with the audio callback.
In my opinion, we shouldn't use atomics at all, but ok.
This switches the mpv code to use C11 stdatomic.h, and for compilers
that don't support stdatomic.h yet, we emulate the subset used by mpv
using the builtins commonly provided by gcc and clang.
This supersedes an earlier similar attempt by Kovensky. That attempt
unfortunately relied on a big copypasted freebsd header (which also
depended on much more highly compiler-specific functionality, defined
reserved symbols, etc.), so it had to be NIH'ed.
Some issues:
- C11 says default initialization of atomics "produces a valid state",
but it's not sure whether the stored value is really 0. But we rely on
this.
- I'm pretty sure our use of the __atomic... builtins is/was incorrect.
We don't use atomic load/store intrinsics, and access stuff directly.
- Our wrapper actually does stricter typechecking than the stdatomic.h
implementation by gcc 4.9. We make the atomic types incompatible with
normal types by wrapping them into structs. (The FreeBSD wrapper does
the same.)
- I couldn't test on MinGW.
The volume controls in mpv now affect the session's volume (the
application's volume in the mixer). Since we do not request a
non-persistent session, the volume and mute status persist across mpv
invocations and system reboots.
In exclusive mode, WASAPI doesn't have access to a mixer so the endpoint
(sound card)'s master volume is modified instead. Since by definition
mpv is the only thing outputting audio in exclusive mode, this causes no
conflict, and ao_wasapi restores the last user-set volume when it's
uninitialized.
Due to the COM Single-Threaded Apartment model, the thread owning the
objects will still do all the actual method calls (in the form of
message dispatches), but at least this will be COM's problem rather than
having to set up several handles and adding extra code to the event
thread.
Since the event thread still needs to own the WASAPI handles to avoid
waiting on another thread to dispatch the messages, the init and uninit
code still has to run in the thread.
This also removes a broken drain implementation and removes unused
headers from each of the files split from the original ao_wasapi.c.
ao_wasapi.c was almost entirely init code mixed with option code and
occasionally actual audio handling code. Split most things to
ao_wasapi_utils.c and keep the audio handling code in ao_wasapi.c.
Gets rid of the internal ring buffer and get_buffer. Corrects an
implementation error in thread_reset.
There is still a possible race condition on reset, and a few refactors
left to do. If feasible, the thread that handles everything
WASAPI-related will be made to only handle feed events.
The uint64_t math would cause overflow at long enough system uptimes
(...such as 3 days), and any precision error given by the double math will
be under one milisecond.
Until now, this was always conflated with uninit. This was ugly, and
also many AOs emulated this manually (or just ignored it). Make draining
an explicit operation, so AOs which support it can provide it, and for
all others generic code will emulate it.
For ao_wasapi, we keep it simple and basically disable the internal
draining implementation (maybe it should be restored later).
Tested on Linux only.
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
Use QueryPerformanceCounter to improve the accuracy of
IAudioClock::GetPosition.
While this is mainly for "realtime correctness" (usually the delay is a
single sample or less), there are cases where IAudioClock::GetPosition
takes a long time to return from its call (though the documentation doesn't
define what a "long time" is), so correcting its value might be important in
case the documented possible delay happens.
The lack of device latency made get_delay report latencies shorter than
they should; on systems with fast enough drivers, the delay is not
perceptible, but high enough invisible delays would cause desyncs.
I'm not yet completely sure whether this is 100% accurate, there are
some issues involved when repeatedly pausing+unpausing (the delay might
jump around by several dozen miliseconds), but seeking seems to be
working correctly now.
Windows applications that use LoadLibrary are vulnerable to DLL
preloading attacks if a malicious DLL with the same name as a system DLL
is placed in the current directory. mpv had some code to avoid this in
ao_wasapi.c. This commit just moves it to main.c, since there's no
reason it can't be used process-wide.
This change can affect how plugins are loaded in AviSynth, but it
shouldn't be a problem since MPC-HC also does this and it's a very
popular AviSynth client.
Remove the nonsensical print_lock too.
Things that are called from the option validator are not converted yet,
because the option parser doesn't provide a log context yet.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
The previous RING_BUFFER_COUNT value, 64, would have ao_wasapi buffer 64
frames of audio in the ring buffer; a delay of 1280ms, which is clearly
overkill for everything. A value of 8 buffers 8 frames for a total of
160ms.
When get_space was converted to returning samples instead of bytes, a
unit type mismatch in get_delay's calculation returned bogus values. Fix
by converting get_space's value back to bytes.
Fixes playback with ao_wasapi when reaching EOF, or seeking past it.
This comes with two internal AO API changes:
1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.
2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)
Change all AOs to use the new API.
The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.
Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
MSDN tells me to multiply the samplerates by 4 (for setting up the S/PDIF
signal frequency), but doesn't mention that I'm only supposed to do it
on the new, NT6.1+ IEC 61937 structs. Works on my Realtek Digital Output,
but as I can't connect any hardware to it I can't hear the result.
Also, always ask for little-endian AC3. I'm not sure if this is supposed
to be LE or NE, but Windows is LE on all platforms, so we go with LE.
Entirely untested as this troper has no S/PDIF hardware.
Refuses trying any other format if we can't use passthrough, or we would
end up sending white noise at the user.
Do an strstr match against the device description and, if we have only
a single match, take it. This works as long as the devices in the system
don't change, but it's not supposed to be reliable; if one wants
reliability, one uses the device ID string.
Formatting.
This could turn valid parameters into syntax errors by the mere presence
or abscence of a device (e.g. USB audio devices), so don't do that.
We do validate that, if the parameter is an integer, it is not negative.
We also respond to the "help" parameter, which does the same as the "list"
suboption but exits after listing.
Demote the validation logging to MSGL_DBG2.
Validates by trying to pick the device using the device enumerator and
aborting with out of range on failure.
Refactors find_and_load_device to not use the wasapi_state; it might be
called during validation. Adds missing CoInitialize/CoUninitialize calls.
Remove unused variables (the SAFE_RELEASE macros keep them referenced so
compiler warnings don't help finding them...).
Remove the IMMDeviceEnumerator from the wasapi_state, it's only needed
during initialization and initialization is now well factored enough to
get rid of it.
Try and connect to unplugged devices as well when using the device ID
string.
Omit "{0.0.0.00000000}." on devices that start with that substring,
re-add when searching for devices by ID.
Log the device ID of the default device.
Log the friendly name of the used device.
Consistently refer to endpoints/devices as devices, as this is more
consistent with mpv terminology.
Kevin Mitchell
wm4
Diogo Franco (Kovensky)
James Ross-Gowan
Stefano Pigozzi
Jonathan Yong