Work around -Werror=catch-value.

Tremor decoder is unusable since commit 2ee43c40. Sound is distorted to
the point where it's nothing but noise.

The data from vorbis_synthesis_pcmout() needs to be scaled and
clipped for 16 bit sample size. For reference see
http://lists.xiph.org/pipermail/tremor/2010-April/001642.html and
http://lists.xiph.org/pipermail/vorbis/2006-October/026513.html.
Signed-off-by: Marcin Jurkowski <marcin1j@gmail.com>

GME starts all track indexes at zero, but subtune prefixes
start at one. This fixes an off-by-one error during track
enumeration.

ffmpeg plugin: when decoded stream duration is unavailable, attempt fallback to container duration (fix MusicPlayerDaemon/MPD#110)

Addresses #82.

Previously, the ffmpeg decoder only checked for the "generic"
MusicBrainz metadata keys used in other metadata container formats.

Add ffmpeg decoder support for *.adx files (Sega game console)

Closes #60

This eliminates some overhead, because the compiler doesn't need to
consider these functions throwing.

Fixes a potential crash bug which is actually unreachable, because the
"pcm" plugin is never invoked when there is no (matching) MIME type.

After a seek failure, the buffer may still be full, and then
FillBuffer() aborts with assertion failure.

https://bugs.musicpd.org/view.php?id=4665

https://bugs.musicpd.org/view.php?id=4656 describes a crash due to
division by zero because frame.samples==0.  This should never happen,
but apparently can happen after seeking.  The best we can do is to
just ignore this frame.

https://bugs.musicpd.org/view.php?id=3876

From the avformat_open_input() API documentation:

 "Note that a user-supplied AVFormatContext will be freed on failure."

https://bugs.musicpd.org/view.php?id=4607

The libWavPack documentation says:

 "After a FALSE return the file should not be accessed again (other
 than to close it); this is a fatal error."

Technically, these samples are signed.  There was no practical
difference, only the declaration was formally wrong.

Since we don't use OPEN_2CH_MAX, this should be equal to
WavpackGetNumChannels() - but who knows, maybe a future libWavPack
version breaks this assumption.

Enable OPEN_DSD_AS_PCM if available.  No OPEN_DSD_NATIVE yet.

 https://bugs.musicpd.org/view.php?id=4606

Use MPD's internal APE tag code, which works just as well.  This
removes some duplicate code for reading tags and ReplayGain.

MPD uses a floating point range from -1 to +1 internally - why ask
libwavpack to use -32768..+32768 only to reduce it back to -1..+1
afterwards?