This encoder plugin is a replacement for the LAME encoder plugin for
those who prefer a "free" (non-patent encumbered) encoder library.
Most of the plugin source code is copied from the LAME encoder plugin,
since the LAME and TwoLAME APIs are nearly the same.

Fail when a feature is enabled, but the library is not found.

Don't append these to MPD_CFLAGS and MPD_LIBS.

Those were only wrappers for playlist_state_{save,restore}().  Since
sf_callbacks has been removed, we can call the latter functions
directly.

There are very few callbacks, and they are not meant to be pluggable.
Let's eliminate the array and call the load/save functions manually.

External cue sheet file for "file.flac" should be named as "file.flac.cue".

According to the ID3 2.4 documentation, "TOPE" is "Original
artist/performer", not "performer".  Removed "TOPE" support.  Instead,
map TPE3 ("Conductor/performer refinement") and TPE4 ("Interpreted,
remixed, or otherwise modified by") to "performer".

The tag_id3.c library supports both the documented "TSO2" tag, and the
inofficial TXXX/ALBUMARTISTSORT.

The Vorbis/FLAC decoder automatically supports the new tag, without
further change.

If the encoder plugin API is disabled at compile time, don't compile
test/run_encoder.c.

The variables "success" and "error" are only used if SQLite support is
enabled.

Very few lines to log a song URI when it has been entirely played.
Then mpd logs can be parsed to do statistics.

Do all the software volume stuff inside each output thread, not in the
player thread.  This allows one software mixer per output device, and
also allows the user to configure the mixer type (hardware or
software) for each audio output.

This moves the global "mixer_type" setting into the "audio_output"
section, deprecating the "mixer_enabled" flag.

The special-purpose function is used for saving/restore the software
volume control to the state file.

This mixer plugin may be used instead of the traditional global
software mixer.  It integrates with the "volume" filter plugin, and
can control the software volume of an audio output which has no
hardware mixer.

Converted the range checks in volume_level_change() to assertions.
Changed all volume types to "unsigned", expect for those which must be
able to indicate error (-1).

Since the "volume" command has been removed, nobody uses relative
volumes anymore.

This command has been deprecated more than 5 years ago (0.10.0).  Its
implementation is a kludge, let's remove it now.

The "convert" filter must be the last filter in the chain.  Ensure
that by doing its initialization at the very end of
audio_output_init().

When song->mtime was not initialized properly, it was revealed that
strftime() might crash when gmtime_r() returns NULL due to an invalid
time_t input value.

This patch adds initial filter support for audio outputs.  Each audio
output gets a "filter" attribute, which is used by ao_play_chunk().

The PCM conversion is now performed by convert_filter_plugin.
audio_output.convert_state has been removed.

REOPEN is called when the input audio format changes.  The output
thread may be reconfigure the PCM converter.

A recent change to the boolean parser introduced a bug: instead of
using the block_param's value with get_bool(), we passed param->value
(which is always NULL in this case).

Added public methods to get and set the current volume.

Some clients have visual feedback for "database update is running".
Using the "database" idle event is unreliable, because it is only
emitted when the database was actually modified.  This patch adds the
"update" event, which is emitted when the update is started, and again
when the update is finished, disregarding whether it has been
modified.

Added the response line "Last-Modified", which sends the modification
time in ISO 8601.  The same was already implemented for playlists.

We're going to extend the protocol now.  Update the protocol version.

Renamed functions.