This is not done automatically by CoreAudio. I am told that it would a PITA
to have to switch back the format manually on the device (especially if the
same device is used for lpcm output).
b2f9e0610 introduced this functionality with code that was quite 'monolithic'.
Split the functionality over several functions and ose the new macros to get
array properties.
Introduce some macros to deal with properties. These allow to work around the
limitation of CoreAudio's API being `void **` based. The macros allow to keep
their client's code DRY, by not asking size and other details which can be
derived by the macro itself. I have no idea why Apple didn't design their API
like this in the first place.
* ao_coreaudio_utils: contains several utility function
* ao_coreaudio_properties: contains functions to set and get audio object
properties.
Conflicts:
audio/out/ao_coreaudio.c
The condition was checked wrongly on asbd which is the input format
description. This lead to the condition always being true, thus selecting lpcm
streams for digital input.
The initialization is split more clearly between compressed and lpcm case.
For the compressed case, format selection is simplified a lot and negotiation
removed. The way it was written it just passed back to the core the original
requested format, not what was found available on hardware.
Since this is most likely useless for the compressed case, I didn't bother
with this. In the future I'd like to split this AO in two one that only uses
the AUHAL and the other with direct access to the hardware so that even
passthrough of lcpm can be possible. This would decrease the latency,
audiophiles would like that.
Split out some utility functions that use the CoreAudio API but are not related
the main task of the AOs (which is to move data correctly to the ringbuffer).
These are mainly need for the verbosity of the CoreAudio API and are just
obscuring the 'real' code.
Read only the requested amount by the AUHAL (instead of all the buffered data).
No idea what the deal is with pausing the audio units if there is no audio to
play, maybe to avoid underruns of some sort. Anyway from my tests this
condition never occurred so I'm removing it all.
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.
Uses WASAPI in shared mode by default, add :exclusive flag to choose
exclusive mode (duh). WASAPI works somewhat different in shared mode:
the OS suggests the sample format to use, and the GetBuffer call is
done slightly differently.
The shared mode driver does not consume audio as fast as it notifies
the thread; we need to check how much we're allowed to write. Not doing
this correctly results in spamming the console with
AUDCLNT_E_BUFFER_TOO_LARGE errors.
When guessing formats for exclusive mode, try several sample size and
sample rate combinations instead of just falling back to s16le@44100hz.
If none of the rates are accepted, tries remixing >6 channels to 5.1
channels. Failing that, tries remixing to stereo. Failing everything,
including the CD Red Book format, what else is left to test?
Calculate buffer_block_size based on the configured channels and bytes
per sample; MSDN docs say nBlockAlign is not guaranteed to be set for
anything but integer PCM formats.
Adds the :list suboption to ao_wasapi0, which enumerates the audio endpoints
in the system.
Adds the :device=<n> suboption, which either takes an ID string (as output by
list) or a device number and uses the requested device instead of the system
default.
These two options were supported by ALSA and OSS only. Further, their
values were specific to the respective audio systems, so it doesn't make
sense to keep them as top-level options.
This changes how device names are handled. Before this commit, device
names were mangled in strange ways to avoid clashing with the option
parser syntax. "." was replaced with ",", and "=" with ":" (the user had
to do the inverse to get the correct device name).
The "new" option parser has multiple ways to escape option strings, so
we don't need this confusing hack anymore.
Add an explicit note to the manpage as well.
Seeking calls thread_reset, but doesn't call thread_play. thread_reset
would disable WASAPI events, but they would never get re-enabled unless
the user paused and then unpaused.
Keep track of whether the stream is paused or not (there already was a
field for that, but it was apparently unused), and if it's not paused,
call thread_play after thread_reset. Fixes mpv freezing after seeks.
Fixes format specifies that assume windows TYPEDEFS are as long as they look
like they are.
Remove calls to _beginthreadex and _endthreadex, these are only present on
microsoft's C runtimes. Replace by the otherwise identical CreateThread and
ExitThread calls.
This actually requires fixes to devicetopology.h, but the problem has been
(kinda) reported to mingw-w64:
<Kovensky> I see that those KSJACK* structs are supposedly declared in
devicetopology.h itself, but for some reason (some of?) the decls that use
them aren't seeing them?
<Kovensky> ok, it seems that it expects ks.h and ksmedia.h to declare those
structs, but it doesn't
<Kovensky> the included files declare KDATAFORMAT, KSIDENTIFIER and LUID (and
the associated pointer typedefs)
<Kovensky> but everything else is essentially inside #if 0
<Kovensky> changing the #ifndef _KS_ to only include KDATAFORMAT, KSIDENTIFIER
and LUID (and putting the KSJACK stuff outside that #ifndef) makes the
header compile
<Kovensky> it solves my immediate problem, but if that happened to begin with
there's probably something more wrong with the ks headers :S
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.
Remove these fields. In places where they are still needed, make them
private AO state.
Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
Some still do, because they use the value in other places of the init
function. ao_portaudio is tricky and reads ao->bps in the stream
thread, which might be started on initialization (not sure about that,
but better safe than sorry).
Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.
I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.
The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).
Also adapted ao_coreaudio to use this ringbuffer.
This is hopefully the start of something good. ca_ringbuffer_read and
ca_ringbuffer_write can probably cleaned up from all the NULL checks once
ao_coreaudio.c gets simplyfied.
Conflicts:
audio/out/ao_coreaudio.c
This was overlooked with commit 32a898f, because OSS4 volume control is
typically not available on Linux. BSD does have this feature, so the
broken code broke compilation there.
