Removed a superfluous decoder_get_command() call.

The audio_format variable is only used and initialized for
decoder_initialized().  Move it into that block to save some bytes on
the stack.

aac_stream_decode() was basically copy+pasted from aac_decode().
Since stream_decode() can also decode files, eliminate aac_decode().

Check whether enough data has been read yet.

If the stream is not seekable, don't try to decode all frames to find
out the total song time.

Use input_stream_eof() instead.

According to the documentation, mpc_decoder_decode() returns an
mpc_uint32_t.  Since the special return value (mpc_uint32_t)-1
translates to a very large long integer, this may cause segmentation
faults if not interpreted properly.

Don't hard-code the factor "2".

Removed 3 superfluous decoder_get_command() invocations.

mpc_to_mpd_buffer() converts a whole chunk at a time.  This eliminates
3 local variables in mpc_decode().

Don't split the buffer conversion loop.  When libmpcdec returns a
chunk, convert and send the whole chunk at a time.  This moves several
checks out of the loop, and greatly improves performance.

Don't use deprecated MPD libraries.

The buffer is always casted to int32_t* anyway; declare it as int32_t
array, and remove the cast.

Since each "eof=true" is followed by "break", the variable is
superfluous.

Due to an unnoticed merge error, there was a superfluous "}".  Remove
it.

Move the reponsibility for freeing the file handle to libvorbis.

decoder_file_decode() should check for plugin->file_decode, not
plugin->stream_decode().

Eliminate two decoder_get_command() invocations from the main loop.

Don't let the buffer grow until it is full, flush it whenever there is
data available.

Parse ID3 tags, even when they are in the middle of the stream.  Very
few streams provide embedded ID3 tags.  Most of them send only
Shoutcast "icy" tags, which limits the practical usefulness of this
patch.

When a command is received, decode_next_frame_header() and
decodeNextFrame() return DECODE_BREAK.  This is already checked by
both callers, which means that we can eliminate lots of
decoder_get_command() checks.

When a tag is updated, the old tag was freed before the new one was
created.  Reverse the order to be sure that other threads always see a
valid pointer.

This still leaves a possible race condition, but it will be addressed
later.

When the tag of the current song changes (e.g. a new tag was sent in
the stream), update the playlist, so clients pick up the new tag.

The stream_decode() and file_decode() methods returned a boolean,
indicating whether they were able to decode the song.  This is
redundant, since we already know that: if decoder_initialized() has
been called (and dc.state==DECODE), the plugin succeeded.  Change both
methods to return void.

os_compat.h is deprecated and should be replaced with well-defined
standard headers.

Eliminated deprecated libraries.

The currently replay_gain_apply() implementation duplicates code from
pcm_volume(), except that it uses a floating point scale.  Eliminate
all duplicated code from and make it utilize the pcm_volume() library
function.  This introduces replay gain support for 24 bit audio.

Allow pcm_volume() to increase volume.

It may be desirable to change the range of integer volume levels
(e.g. to 1024, which may utilize shifts instead of expensive integer
divisions).  Introduce the constant PCM_VOLUME_1 which describes the
integer value for "100% volume".  This is currently 1000.

Having an array instead of individual variables allows the use of the
replay_gain_mode enum as an array index.

The function simplifies wavpack_replaygain(), because it already
contains the float parser, and it works with a fixed buffer instead of
doing expensive heap allocations.

This allows us to remove the "static char[]" hack.

Some code simplification.  Avoid pointers to pointers.

Prefer C enums over CPP macros.

The author's name is already mentioned in the copyright declaration.

Renamed functions and variables.