Even if libsamplerate support is enabled, compile the fallback
resampler.  When the user specifies the option
"samplerate_converter=internal", it is chosen in favor of
libsamplerate.  This may help users with a weak FPU who don't want to
compile a custom MPD from source, because the fallback resampler does
not use floating point operations.

After a seek, wait until enough new chunks are decoded before starting
playback.  If this takes too long, send silence chunks to the audio
outputs meanwhile.

This is similar to the MPD 0.14 patch "wait 10 seconds before
reopening a failed device", which only covered open() failures.  This
patch adds the same feature for play().

The default values for buffer_time and period_time were both capped by
the hardware limits on practically all chips.  The result was a
period_time which was half as big as the buffer_time.  On some chips,
this led to lots of underruns when using a high sample rate (192 kHz),
because MPD had very little time to send new samples to ALSA.

A period time which is one fourth of the buffer time turned out to be
much better.  If no period_time is configured, see how much
buffer_time the hardware accepts, and try to configure one fourth of
it as period_time, instead of hard-coding the default period_time
value.

This is yet another attempt to provide a solution which is valid for
all sound chips.  Using the SND_PCM_NONBLOCK flag also seemed to solve
the underruns, but put a lot more CPU load to MPD.

The lastfm input plugin enables MPD to play lastfm:// URLs.  This
plugin is not complete yet: it plays only the first song in the
last.fm playlist, and the playlist parser isn't even implemented
properly.

Added a small RIFF parser library.  Look for an "id3" chunk, and let
libid3tag parse it.

This is the first patch in a series to enable 32 bit audio samples in
MPD.  32 bit samples are more tricky than 24 bit samples, because the
integer may overflow when you operate on a sample.

Parse the vorbis comments in libflac's metadata_callback and pass them
as tag struct to the decoder API.

Don't hard code the "bits" parameter to 16.  Try to use the input's
sample format, if possible.

The option "enabled" is on by default.  If you specify "enabled no" in
an audio_output section, then this device is disabled by default.

[mk: adapted to new output plugin API]

The generic sockaddr struct is too small for some addresses.  For
accept(), we have to allocate a sockaddr_storage struct on the stack,
which is large enough for all addresses.

Added the uri_remove_auth() library function which strips username
and password from a HTTP URI, and use it in song_print_url().  This
allows you to add HTTP URIs to the playlist including secret username
and password, without disclosing it to all MPD clients.

The check "open()!=0" is wrong, you have to write "open()>=0", because
-1 means error, and 0 is a valid file handle.

When the MVP device has been closed in the cancel() method, and the
play() method attempts to reopen it, check for errors.

Looks like the MVP audio output only supports 16 and 24 bit audio
samples.  If MPD generates any other sample formats, force it to use
16 bit.

When the channel count is greater than 2, fall back to stereo sound.

If mpd.conf specifies a user, and MPD is invoked by exactly this user,
ignore the "user" setting.  Don't bother to look up its groups and
don't attempt to change uid, it won't work anyway.

Use delete_directory() for removing sub directories instead of
dirvec_clear().  This ensures that all memory occupied by
subdirectories of deleted directories is freed.

When a directory is deleted, MPD deleted only the directory from the
database; it did not bother to walk the full tree to free all memory
and to remove deleted songs from the playlist.  Replace a
dirvec_delete() with delete_directory().

Pass the input_stream object to decoder_data().  Without it, the MPD
core does not see stream tags.

There are a few problems left in this plugin:

- fluidsynth decodes in real time, while MPD prefers to buffer as
  quickly as possible; as a workaround, this plugin uses a timer
  object to synchronize with real-time playback

- I don't know yet how fluidsynth tells me when the song has ended

- the "soundfont" configuration setting is not yet documented, and it
  will likely change soon (in favor of a per-decoder configuration
  block)

The ffmpeg library supports the "True Audio Codec".  The entry in
ffmpeg_suffixes was missing.

Don't define HAVE_FFMPEG if the ffmpeg libraries were found via
pkg-config, but ffmpeg support was disabled because
avcodec_decode_audio2() is not available.

Always assume the buffer is empty before calling the encoder.  Always
flush the buffer immediately after there has been added something.
This reduces the risk of buffer overruns, because there will never be
a "rest" in the current buffer.

Don't duplicate the tag received by the send_metadata() method - send
it to the shout server directly.

Removed the manual timer synchronization from the shout plugin.
libshout's shout_sync() function does it for us.

The non-blocking mode of libshout is sparsely documented, and MPD's
implementation had several bugs.  Also removed connect throttling
code, that is done by the MPD core since 0.14.

The "current" variable is used for calculating the seek destination,
and was declared as "int".  With very long song files, the 32 bit
integer can overflow.  ffmpeg expects an int64_t, which is very
unlikely to overflow.  Switch to int64_t.

When ffmpeg cannot estimate the elapsed time, it sets
AVPacket.pts=AV_NOPTS_VALUE.  Our ffmpeg decoder plugin did not check
for that special value.

If avcodec_decode_audio2() returns no output for an AVPacket,
libavcodec may buffer some data, and return a larger chunk of output
later.  This patch disables a lot of bogus warnings.

The shout_mp3 encoder had two bugs: when no song was ever played, MPD
segfaulted during cleanup.  Second bug: memory leak, each time the
shout device was opened, lame_init() was called again, and
lame_close() is only called once during shutdown.

Fix this by shutting down LAME each time the clear_encoder() method is
called.

When the output device fails to play a chunk, set pc.error to
PLAYER_ERROR_AUDIO.  This way, the playlist knows that it should not
queue the next song.

Hi -
independently of libmikmod's other problems - there seems
to be a problem in mpd's wrapper: MikMod_Exit() is called
after the first file is decoded, which frees some ressources
within the mikmod library. An attempt to play a second file
leads to a crash. The appended patch fixes this for me.
(I don't know what the "dup" entry is good for - someone
who knows should review that too.)
best regards
Matthias

[mk: removed 3 more MikMod_Exit() invocations]

The wavpack library seems to use the .wvc stream even if the OPEN_WVC
flag is not set.  In this case, pass NULL to be sure libwavpack won't
use it.