Removed local variable "sample_rate".

Not used by any plugin currently, but this eliminates the g_error()
call in input_plugin_config(), so it's worth it.

This is easier to traverse.

Wrap the AudioCompress library in a filter plugin.

Log to stderr, not to stdout (which broke PCM output).

Conflicts:
	src/decoder/ffmpeg_plugin.c

Use the signed C99 type int8_t instead.

Mixers with the "global" flag set aren't closed automatically when the
output device is closed.  Thus, they might still be open when MPD
shuts down.

This fixes an inconsistency in the stored playlist subsystem: when
obtaining the list of playlists (listplaylist, listplaylistinfo), the
file names in the playlist directory are converted to UTF-8 (according
to filesystem_charset), but when saving or loading playlists, the
filesystem_charset setting was ignored.

Call spl_valid_name() in spl_delete().

Don't allocate the file name before the playlist_dir==NULL check.

If both tags (stream and decoder) are present, we prefer the stream tag.
Fixes #2698, where ICY tag contained useful information, but was
overwritten with bogus decoder tag data.

This includes some default values of #defined constants used in the
code; it won't compile without it.

This patch prepares support for floating point samples (and probably
other formats).  It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.

Copied sources from
http://beesbuzz.biz/code/audiocompress/AudioCompress-2.0.tar.gz

[mk: created this patch under fluffy's name and fixed some gcc
signed/unsigned comparison warnings]

If we define id3v1_encoding, then the tags are not added to the
database.

The plugin code tried to force libavcodec to supply stereo samples.
That however has never actually worked.  By removing this code, we are
able to play surround files for the first time.

This fixes a regression due to a typo caused by "decoder: use
audio_format_init_checked()".