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.
Fixes crashes when playing with certain numbers of channels. The core
assumes AOs accept data aligned on channels * samplesize, and ao_jack's
play() function broke that assumption:
mpv: core/mplayer.c:2348: fill_audio_out_buffers: Assertion `played % unitsize == 0' failed.
Fix by aligning the buffer and chunk sizes as needed.
GetTimer() is generally replaced with mp_time_us(). Both calls return
microseconds, but the latter uses int64_t, us defined to never wrap,
and never returns 0 or negative values.
GetTimerMS() has no direct replacement. Instead the other functions are
used.
For some code, switch to mp_time_sec(), which returns the time as double
float value in seconds. The returned time is offset to program start
time, so there is enough precision left to deliver microsecond
resolution for at least 100 years. Unless it's casted to a float
(or the CPU reduces precision), which is why we still use mp_time_us()
out of paranoia in places where precision is clearly needed.
Always switch to the correct time. The whole point of the new timer
calls is that they don't wrap, and storing microseconds in unsigned int
variables would negate this.
In some cases, remove wrap-around handling for time values.
The ALSA device was not closed when initialization failed.
The ALSA error handler (set with snd_lib_error_set_handler()) was not
unset when closing ao_alsa. If this is not done, the handler will still
be called when other libraries using ALSA cause errors, even though
ao_alsa was long closed. Since these messages were prefixed with
"[AO_ALSA]", they were misleading and implying ao_alsa was still used.
For some reason, our error handler is still called even after doing
snd_lib_error_set_handler(NULL), which should be impossible. Checking
with the debuggers, inserting printf(), as well as the alsa-lib source
code all suggest our error handler should not be called, but it still
happens. It's a complete mystery.
It turns out that ALSA's 4 channel layout is different from mpv's and
ffmpeg's 4.0 layout. Thus trying to do 4 channel output led to incorrect
remixing via lib{av,sw}resample.
Fix the default layouts for the internal filter chain as well, although
I'm not sure if it matters at all.
The snd_pcm_hw_params_test_format() call actually crashes in alsa-lib if
called with SND_PCM_FORMAT_UNKNOWN, so the already existing fallback
code won't work in this case.
Make all AOs use what has been introduced in the previous commit.
Note that even AOs which can handle all possible layouts (like ao_null)
use the new functions. This might be important if in the future
ao_select_champ() possibly honors global user options about downmixing
and so on.
The point is selecting a minimal fallback. The AOs will call this
through the AO API, so it will be possible to add options affecting
the general channel layout selection.
It provides the following mechanism to AOs:
- forcing the correct channel order
- downmixing to stereo if no layout is available
- allow 5.1 <-> 5.1(side) fallback
- handling "unknown" channel layouts
This is quite weak and lots of code/complexity for little gain. All AOs
already made sure the channel order was correct, and the fallback is of
little value, and could perhaps be done in the frontend instead, like
stereo downmixing with --channels=2 is handled. But I'm not really sure
how this stuff should _really_ work, and the new code will hopefully
provides enough flexibility to make radical changes to channel layout
negotiation easier.
Now mpv's channel map is used to map each channel to a speaker. This
allows in theory for playback of any layout for which ao_openal
actually has a speaker defined. Also add the back-center (BC) speaker,
which allows playback of 6.0 audio. Enabling more layouts by adding
other speakers would be possible, but I'm not sure about the speaker
positions.
This allows supporting 5 channel audio (which can be eother 5.0 or 4.1).
Fallback doesn't work yet. It will do nonsense if the channel layout
doesn't match perfectly, even though it's similar.
Add a CHECK_ALSA_ERROR macro to report ALSA errors. This is similar to
what vo_vdpau does. This removes lots of boiler plate, it almost gives
me the feeling the ao_alsa initialization code is now readable. This
change is squashed with the reformatting, because both changes are
just as noisy and useless.
Like most other AOs, ao_pulse set the channel count only, always using a
default layout. Try to set the exact layout.
For this, we need a big lookup table to map waveex/lavc/mpv speaker
position to PulseAudio's, since PA_CHANNEL_POSITION_ is apparently not
compatible to waveext, and I haven't seen any API functions that would
help mapping them.
Completely untested. (Let's leave that to someone else...)
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.
Also move the mp_audio struct to a the file audio.c.
We can remove a mysterious line of code from af.c:
in.format |= af_bits2fmt(in.bps * 8);
I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
Stefano Pigozzi
Diogo Franco (Kovensky)
wm4
Jonathan Yong
Rudolf Polzer
reimar