player/Thread: calculate `buffered_before_play` based on a fixed duration

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.

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