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().
If no ports are configured, don't overwrite the (NULL) configuration
with the port names of the first JACK server. If the server changes
after a JACK reconnect, MPD won't attempt to auto-detect again.
Currently, the JACK plugin manipulates the audio_format struct which
was passed to the open() method. This is very likely to break,
because the plugin must not permanently store this pointer. After
this patch, MPD ignores sample rate changes. It looks like other
software is doing the same, and I guess this is a non-issue.
This patch converts the audio_format pointer within jack_data into a
static audio_format struct.
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]
When there are duplicate slashes in the song paths, eliminate them;
example:
/var/lib/mpd/music//foo.mp3
becomes:
/var/lib/mpd/music/foo.mp3
The slash is only detected at the border between the music_directory
and the local part.
When the user configures a music_directory with a trailing slash, it
may break playlist loading, because MPD expects a double slash. Chop
off the trailing slash.
ffmpeg_tag_internal() does not look for a few tags that mpd
supports. Most noteably:
comment -> TAG_ITEM_COMMENT -> Description
genre -> TAG_ITEM_GENRE -> WM/Genre (not WM/GenreID)
year -> TAG_ITEM_DATE -> WM/Year
I *think* that this is the last of the tags that AVFormatContext() in
ffmpeg supports that mpd also uses.
Make those two methods optional to implement, and let input_stream.c
provide fallbacks. The buffer() method will be removed one day, and
there is now only one implementation left (input_curl.c).
The open_stream() method opens the input_stream. This allows the
archive plugin to do its own initialization, and it also allows it to
use input_stream.data. We can remove input_stream.archive now, which
was unnatural to have in the first place.
Preparation for supporting other channel numbers than stereo: use
loops instead of duplicating code for the second channel. Most
likely, gcc will unroll these loops, so the binary won't be any
different.
Max Kellermann