When MPD is stopped, but the last song is still the "current song",
and you delete it, playlist->current is not updated, and becomes an
invalid value. Fix this by catching "!playlist->playing &&
playlist->current == (int)songOrder".
audio_output_config_count() returns the number of audio outputs in the
configuration file. It is only used by initAudioDriver(). The public
function audio_output_count() now returns audioOutputArraySize.
When we reset pc.next_song if there is no song queued, this might
cause a race condition: the next song to be played is cleared, while
pc.command was already set. Clear the "next_song" only if there is a
song queued for the current do_play() invocation.
If a new song is queued before calling playerSeek(), then the player
and the playlist enter an inconsistent state, because the player
discards the playlist's "queued" song in favor of the seeked song.
Call playlist_update_queued_song() after playerSeek().
After a player command (successful or not), reset pc.next_song,
because the queue is supposed to be empty then. Otherwise,
playlist.queued and pc.next_song may disagree, which triggers an
assertion failure.
Commit f78cddb4 introduced a regression: after a song was moved, the
order array was not updated (in random mode). This caused MPD to
think the "current" song has changed when you moved something to the
position of the current song.
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.
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.
After the state file has been loaded, the playlist version is still
"1", and "plchanges 1" returns the whole playlist. Fix this by
increasing the playlist version after the state file has been loaded.
Don't call syncPlaylistWithQueue() in nextSongInPlaylist() and
previousSongInPlaylist(). This is a relic from the time when there
was no event, and was a workaround to the timing problem.
Export the "g_playlist" variable, and pass it to all playlist
functions. This way, we can split playlist.c easier into separate
parts. The code which initializes the singleton variable is moved to
playlist_global.c.
Before every operation which modifies the playlist, remember a pointer
to the song struct. After the modification, determine the "next song"
again, and if it differs, dequeue and queue the new song.
This removes a lot of complexity from the playlist update code, and
makes it more robust.
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.
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.
Output the name of the codec as a debug message. During my tests,
ffmpeg never filled this struct member, but it may do so in the past,
and this debug message might become helpful.
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 update thread is started before MPD has forked (for
daemonization), it is killed, because threads do not survive a fork().
This induces an inconsistent state where MPD won't start any update
thread at all, because it thinks the thread is already running.
Move the "while" loop which checks for commands to the caller
ao_pause(). This simplifies the pause() method, and lets us remove
audio_output_is_pending().