The decoder_plugin struct is used by both the MPD core and the decoder
plugin implementations. Move it to a shared header file, to minimize
header dependencies.
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().
When you change the filesystem charset, discard the old database file
and create a new one. The old database file will most likely contain
stale or invalid information.
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)
When MPD is not playing, it may still remember which is the "current"
song. When you switch to "random" mode, MPD will always start playing
exactly this song. This defies the goal of "random" mode a little.
Clear the "current" song when MPD is not playing during the "random"
mode switch.
The output_command library provides a command interface to the audio
outputs. It assumes the input comes from an untrusted source
(i.e. the client) and verifies all parameters.
In addition to audio_format_valid(), provide functions which validate
only one attribute of an audio_format. These functions are reused by
audio_format_parse().
Added audio_format_parse() in a separate library, with a modern
interface: return a GError instead of logging errors. This allows the
caller to deal with the error.
When MPD explicitly starts playing, ignore the "REOPEN_AFTER" timeout.
This timeout was useful when MPD attempted to reopen a failed device
over and over, but it confuses users when they explicitly tell MPD to
start playing, while MPD insists to wait for the 10 seconds to pass.
Fix a memory leak: it was not guaranteed that pcm_convert_deinit() was
called for each pcm_convert_init(). This patch always (de)initializes
the pcm_convert library when the audio_output.open flag is flipped.
Pass the music chunk as a "const void *" to the encoder, instead of a
"const char *". Actually, both encoders currently expect 16 bit
samples, passing a 8-bit character is rather pointless.
The crossfading code shouldn't depend on the audio output code. Pass
the current audio format to cross_fade_calc() and let it compare
directly, instead of using isCurrentAudioFormat().
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.
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().
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]