This reverts commit ff3e2c0514. The
check was necessary, after all, because this is what checked whether
the decoder had finished the current or the next song.
> The "queued" flag can only possibly be set if the decoder is still
> decoding the current song or if the decoder is stopped.
That was wrong because ProcessCommand() sets `queued=true` and also
starts the decoder (if it was idle).
> This is also what the following assert() checks.
That was also wrong, because the assert() has two conditions.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/566
If the decoder finishes decoding the current song between the two
IsIdle() checks, MPD stops playback instead of starting the decoder
for the next song.
This is usually not visible problem, because the main thread restarts
it via playlist::ResumePlayback(), but that way it, ignores "single"
mode.
As a workaround, this commit adds another "queued" check which
re-enters the player loop and checks again whether to start the
decoder.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/556
The "queued" flag can only possibly be set if the decoder is still
decoding the current song or if the decoder is stopped. This is also
what the following assert() checks. This check was not necessary.
Since we switched from autotools to Meson in commit
94592c1406, 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.
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.
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.
