Our AudioObjectGetPropertyDataT() wrapper throws exception on error,
and calling it from OSXOutput::Disable() can cause MPD crash due to
std::terminate().

Closes https://github.com/MusicPlayerDaemon/MPD/issues/932

Wait until there is data in the ring buffer.

Wait until there is some data; don't let our render callback be
invoked without any data.

We shouldn't restart the AudioUnit while the ring buffer is empty, or
else our render callback may emit noise.

Closes https://github.com/MusicPlayerDaemon/MPD/issues/771

This will keep track of AudioOutputUnitStart() and
AudioOutputUnitStop().  This will provide some separation between "not
(yet) (re)started" and "paused".

This shall be done by Close(), which will be called automatically
after an error.

The formula in osx_output_score_sample_rate() to detect multiples of
the source sample rate was broken: when given a 44.1 kHz input file,
it preferred 16 kHz over 48 kHz, because its `frac_portion(16)=0.75`
is smaller than `frac_portion(48)=0.91`.

That formula, introduced by commit 40a1ebee, looks completely
wrong.  It doesn't do what the code comment pretends it does.

Instead of using that `frac_portion` to calculate a score, this patch
adds to the score only if `frac_portion` is nearly `0` or `1`.  This
means that the factor is nearly integer.

Closes https://github.com/MusicPlayerDaemon/MPD/issues/904

Fixes `-Wpedantic`.

Currently it falls back to system default device (either internal speaker or headphone) when device not found. 
I believe it is a better to fail in this case, to make it better aligned with platforms (such as alsa).

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.

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.