Previously, there was the setting `buffered_before_play` which
specified a percentage of the audio buffer, defaulting to `10%`.  That
was working well enough for quite some time, until high-quality audio
formats became common.

At 44.1 kHz, 16 bit stereo, MPD collected 2.3 seconds worth of data in
the buffer before starting playback.  With the same default settings
and 192 kHz, 24 bit stereo, that was only 0.27 seconds.

Making this depend on the byte size only leads to high latency at low
quality, and too little data at high quality.  The natural choice
would be to use a duration instead of a byte size, which should give
the same good experience with all audio formats.

Since the `buffered_before_play` configuration setting was not
understood well by users and caused more harm than good, this commit
deprecates it.  It has now no effect.

Fixes integer truncation bugs (closes #359).

It is already inherited by the base class, and we don't need to import
it explicitly.

Simplify the formula, and I guess this makes the formula more
reliable.  Imagine somebody configured `buffered_before_play` larger
than 25%; then the decoder would be woken up all the time.  This
doesn't seem logical.  On the other hand, it's easy to understand that
the decoder should be woken up below 75% buffer fill.

Calculate the value only once.

Shouldn't ever happen, but who knows...

The `SliceBuffer` constructor can throw.