Since we switched from autotools to Meson in commit
94592c14, we don't need to include
`config.h` early to properly enable large file support.  Meson passes
the required macros on the compiler command line instead of defining
them in `config.h`.

This means we can include `config.h` at any time, whenever we want to
check its macros, and there are no ordering constraints.

Works around a problem where MPD goes into a busy loop because
snd_pcm_drain() always returns `-EAGAIN` without making any progress
(fixes #425).

This problem was triggered by snd_pcm_drain() after snd_pcm_cancel()
and snd_pcm_prepare(), but without submitting any data with
snd_pcm_writei().

I believe this is a kernel bug: in non-blocking mode, the kernel's
snd_pcm_drain() function returns early.  In this mode, it only checks
whether snd_pcm_drain_done() has been called already, but
snd_pcm_drain_done() is never called if no data was submitted.

In blocking mode, the following `for` loop detects this condition, so
snd_pcm_drain_done() is not necessary, but without this extra check,
we get `-EAGAIN` forever.

This fixes a problem which caused a failure with snd_pcm_writei()
because snd_pcm_drain() had already been called in the previous
iteration.  This commit makes sure that snd_pcm_drain() is only called
after the final snd_pcm_writei() call.

This fixes discarded samples at the end of playback.

When a playback error has occurred, MPD would never recover until one
restarts MPD.

CancelInternal() doesn't need to be protected because it is called
synchronously from Cancel().

Fixes a theoretical race condition which could occur in Drain() (but
was extremely unlikely).

If our `ring_buffer` is smaller than the ALSA-PCM buffer (if the
latter has more than the 4 periods we allocate), it can happen that
the start threshold is crossed and ALSA switches to
`SND_PCM_STATE_RUNNING`, but the `ring_buffer` is empty.  In this
case, MPDD will generate silence, even though the ALSA-PCM buffer has
enough data.  This causes stuttering (#420).

This commit amends an older workaround for a similar problem (commit
e08598e7) by adding a snd_pcm_avail()
check, and only generate silence if there is less than one period of
data in the ALSA-PCM buffer.

Fixes #420

The method Cancel() assumes that the `period_buffer` must be empty
when `active==false`, but that is not the case when Play() fails.

Of course the assertion in Cancel() is not 100% correct, but I decided
to rather fix this in LockCaughtError() because the `period_buffer`
should only be accessed from within the RTIO thread, and this is the
only code path where `active` can be set to `false` with a non-empty
`period_buffer`.

Fixes #423

This implements real error handling, and avoids calling
CancelInternal() from this code path.

alsa-lib doesn't set errno, it returns errors as negative integers.

This check was added 9 years ago in commit
4dc25d39 to work around a dmix bug
which I assume has been fixed long ago.

Removing this fixes another corner case: if draining is requested
before the start threshold is reached, the PCM is still in
SND_PCM_STATE_PREPARED but not yet SND_PCM_STATE_RUNNING, which means
the submitted data will never be played.  This corner case is
realistic when playing songs shorter than the ALSA buffer (if the
buffer is very large).

This fixes a corner case which has probably never occurred and
probably never will: if Cancel() is called, and then Play() followed
by Drain(), the plugin should really play that data.  However
currently, this never happens, because snd_pcm_prepare() is never
called.

This call was missing, causing very high CPU usage when the ALSA
output plugin was used with dmix.

Closes #391

When `metadata_sent` is `false`, the plugin assumes there is metadata
which must be sent, even if no metadata page was passed to the plugin.
Initializing it to `true` avoids dereferencing this `nullptr`.

Fixes #412

Bugs in libroar which broke the MPD build have been annoying me for
quite some time, and the newest bug has now hit my main build machine:

 https://github.com/MusicPlayerDaemon/MPD/issues/377

Problem is the usage of the typedef `_IO_off64_t` in libroar's
`vio_stdio.h`:

 int roar_vio_to_stdio_lseek (void *__cookie, _IO_off64_t *__pos, int __w);

This `_IO_off64_t` is an internal implementation detail of glibc and
was removed in version 2.28.  Nobody must ever use it.  Why the ****
did the RoarAudio developers use it?  Not using internal typedefs
isn't exactly rocket science.

This annoys me enough to finally remove the plugin.  Anyway, I've
never heard of anybody using RoarAudio, so my best guess is that
nobody will notice.

So long, autotools!  This is my last MPD related project to migrate
away from it.  It has its strengths, but also very obvious weaknesses
and weirdnesses.  Today, many of its quirks are not needed anymore,
and are cumbersome and slow.  Now welcome our new Meson overlords!

the most notable bugs are 

1. osx_output_set_device_format should use the target asbd rather than AudioFormat. This is because asbd's sample rate calculation reflects the real dop target rate of the DAC, white AudioFormat's sample rate is the original DSD format rate.

2. the original code value the highest rate that's the multiple of the target rate. This cause DOP always have the wrong rate chosen. This is also not necessary for PCM playback --- MPD's goal is bit perfect, and it's meaningless to raise to two or four times the PCM sample rate.

3.  if sample_rate cannot be synchronized, the test for falling back to PCM is wrong. If the file format is in DSD format such fallback is necessary, whatever the params.dop setting is.

the code here tried to guard DSD features behind ENABLE_DSD. However, the sample rate setting should be shared between two scenarios. 

https://github.com/MusicPlayerDaemon/MPD/commit/40a1ebee295c569521ea17ffdedc641d1aedd9cb#diff-ce7ecec9ea9ca3df90d9c290cb3ef9d4R795


The code runs fine if the dac supports the sample rate, as Mac OS  will use the device rate if stream rate is 0. 
However, when DAC is uncapable of processing the sample rate, a wrong rate (device rate) will be used for the stream rate.

This method was added in Boost 1.58.

some device seems to have issue with setting kAudioDevicePropertyVolumeScalar  with kAudioObjectPropertyElementMaster. Use AudioToolbox 's kAudioHardwareServiceDeviceProperty_VirtualMasterVolume instead. 

Ideally, we should get the steoro channels first, and set the kAudioDevicePropertyVolumeScalar for each channel, which is doable as presented in https://github.com/cmus/cmus/blob/master/op/coreaudio.c. I will do a follow up PR after refactor PR.

1 sec for pause is too long. we wait for the same amount of time as when ring buffer is not available for writing.

This PR will fix #271.

special thanks to @coroner21 who contributed a nice way to score hardware supported format in #292

Also, The DSD related code are all guarded with ENABLE_DSD  flag.

- Update the mixer to set on device property instead of audio unit property. When user choose "hardware" as mixer type, they will be able to change the hardware device volume instead of the software (AudioUnit) volume. 
- We don't use square root scale in volume calculation as previous code did. This will make the volume level in line with system volume meter --- That is, MPD will have the same percentage volume reading compared to System Setting (Either in "System Preference" or in "Audio Midi Setup" app)

This code was added in 21851c06 but
looks completely broken:

- the status code is "206 OK" but "206" would be "Partial Content"

- the "Content-Length" header has a bogus value

- the "Content-RangeX" parameter has different bogus values (why
  "Content-RangeX" anyway and not "Content-Range"?)

Apart from that, there are strange undocumented non-standard headers
which are probably there to work around bugs/expectations in one
broken proprietary client product.  But these days, MPD doesn't bend
over to support broken clients.  So let's kill this code.

Closes #304