Without this, the compiler may optimize accesses away.

Without this, the compiler may optimize accesses away.

This is useful in multiple mpd instances scenario, or multiple pulse outputs defined on the same mpd instance.
It is actually a more flexible way to route flows than the "sink" parameter, letting the PulseAudio routing do its job, but with the ability to isolate routing for each output.

If not specified, the role remains like it was before this commit, ie "music"

The JACK output plugin would not correctly upmix mono input files when exactly 2 output ports were configured. This fixes that.

Applying software volume to S16 samples means several bits of
precision are lost; at 25% volume, two bits are lost.  Additionally,
dithering adds some noise.

The problem gets worse when you apply the software volume code twice:
for the software mixer volume, and again for the replay gain.  This
loses some more precision and adds even more dithering noise, which
can become audible (see
https://github.com/MusicPlayerDaemon/MPD/issues/542).

By converting everything to 24 bit, we need to shift only two bits to
the right instead of ten, losing nearly no precision, and dithering is
not needed.  Even if the output device is unable to play S24 directly,
we can convert back to S16 with only one stage of dithering.

Closes https://github.com/MusicPlayerDaemon/MPD/issues/542

Just in case.

It's not a tri-state anymore since we introduced C++ exceptions.

Pass only the amount of data to PcmExport::Export() when its full
output fits into the ring buffer.  Using only a part of the
PcmExport::Export() result may cause data corruption because
PcmExport's internal state may contain partial blocks which would need
to be rolled back when only some of its output data was used.

As a side effect, this fixes an assertion failure because
PcmExport::CalcInputSize() considered partial block data and could
cause Play() to return a number larger than the "size" parameter.

If snd_pcm_writei() fails, throw an error and stop playback instead
of going into an endless busy loop.

MPD used to do that when this code lived in the player thread, but it
was removed by commit 98a7c62d; and
the replacement code in the ALSA output plugin didn't have it.

Without this timer, DispatchSockets() may disable the
MultiSocketMonitor and if Play() doesn't get called soon, it never
gets a chance to generate silence.  However if Play() gets called,
generating silence isn't necessary anymore...

Resulting from this misdesign (added by commit ccafe3f3 in 0.21.3),
the silence generator didn't work reliably.

In DispatchSockets(), when there was not enough data, but enough for
current playback, the method would disable the "active" flag so the
next Play() call would re-enable the MultiSocketMonitor.

This was an abuse of the flag which could result in a crash
in Cancel(), because that method asserts that the period_buffer is
empty, which it may be not.

The solution is to add anther flag called "waiting" which shares some
behavior with the old flag.

Really drain all data from the period_buffer in any case.

This assertion is not about whether all data has been consumed, but
about whether there has been any data at all.

Draining isn't finished just because the period_buffer has run empty.
It is only finished after snd_pcm_drain() has succeeded.

Apparently, if snd_pcm_drain() returns EAGAIN, it does not actually
want to be called again; the next call will snd_pcm_drain() will also
return EAGAIN, forever, even though the PCM state has meanwhile
switched to SND_PCM_STATE_SETUP.  This causes a busy loop; to fix
this, we should always check snd_pcm_state() to see if draining is
really required.