| Commit message (Collapse) | Author | Files | Lines |
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After playback has stopped, the ring buffers may still contain
samples. These will be played when playback is started the next
time. We should clear the buffers each time.
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jack_client_new() is deprecated. This requires libjack 0.100
(released nearly 5 years ago). We havn't been testing older libjack
versions anyway.
As a side effect, there is the new option "autostart".
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Instead of using MPD's audio output name (setting "name"), use a
separate configuration option. Change the default to "Music Player
Daemon".
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When a song's tags could not be loaded during database update, log
this as a debug message. Same for a song being removed because its
updated tag could not be read.
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Store a list of supported tag items in the database. When loading a
database which does not have a matching list, we must rescan in order
to get the missing information.
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Convert a string into a tag_type enum.
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Clear the colon. This simplifies all attribute parsers, because they
can now use strcmp() instead of strncmp().
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If left uninitialized, then the decoder thread quits spuriously.
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Signed-off-by: Romain Bignon <romain@peerfuse.org>
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If no song was queued, then player_control.next_song might contain the
value for the next QUEUE command. We must not reset that.
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When the decoder finishes the "queued" song very quickly (before the
"current" song finishes playing), an assertion in do_play() fails
because it thinks that it should start decoding the queued song,
although that has in fact just finished.
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The assertion shouldn't access player_control.next_song without
locking it.
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Simplify several expressions.
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Don't access attributes without the lock.
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Don't access decoder_control attributes directly.
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Lock the player_control object when modifying its attributes.
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Asynchronous decoder startup is gone, and we don't need to check
command==DECODE_COMMAND_START anymore.
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These two variables are redundant, we need only one of them.
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It's not used if pc.error==PLAYER_ERROR_AUDIO.
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This is only a slight change to the previous locking behaviour: keep
the decoder unlocked during the loop, and lock it only while checking
decoder_control.command.
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The START command returns without blocking; we don't need the
asynchronous decoder start anymore.
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Return the result to the caller more quickly. This unifies error
handling: no error can be reported before the command is finished.
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They are just informational.
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Reintroduce a fix from commit 52a0653 (Warren Dukes): "don't call
snd_pcm_drain unless we're already in the RUNNING state". This prevents
ALSA with dmix from sometimes hanging when snd_pcm_drain is called, e.g.
when moving from one song to the next (as in mantis issue 2634).
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When the "next" chunk to be played is NULL, return from ao_play()
immediately, without going over the "while" loop (no-op).
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While paused, the player thread re-locks its mutex and waits for a
signal. This is racy: when the command is set while the thread is
waiting for the lock, it may wait forever. This patch adds another
command check before player_wait().
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After CANCEL, the output thread waits for another signal before it
continues playback, to synchronize with the caller. There were some
situations where this signal wasn't sent properly. This patch adds an
explicit g_cond_signal() at two code positions.
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That variable has been superseded by "remove_notify" (defined in
update_remove.c).
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Don't wake up the target thread in every iteration of the wait() loop.
Waking it up once, right after the command has been set, must be
enough.
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These parameters must be protected with a mutex, too. Wrap everything
inside player_lock()/player_unlock(), and use player_command_locked()
instead of player_command().
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After CANCEL, call g_cond_wait() only if the new command is still
NONE. Problem is that ao_command_finished() has to unlock the
audio_output object, and in the meantime, the player thread might have
submitted a new command.
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Fix a minor memory leak.
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Changed songvec_load() to song_load(). Added start and end markers
for each song. Removed the "key" line, it's redundant.
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The "music_root" global variable is allocated by db_init().
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Use a single GString buffer object in all functions loading the
database. Enlarge it automatically for long lines. This eliminates
the maximum line length for tag values. There is still an upper limit
of 512 kB to prevent denial of service, but that's reasonable I guess.
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Allocate the directory object after the "directory:" line. Assign the
mtime from the input file to this new object, instead of to the parent
directory.
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Fix a minor memory leak in the error handler.
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The old code tried to recover, but what's the point of that? If a
directory is duplicate, something is wrong with the database file.
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It's legal to pass decoder=NULL to decoder_read(). Add a check.
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Allocate a decoder_control object where needed, and pass it around.
This will allow more than one decoder thread one day.
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Simply use "return" instead of g_thread_exit().
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When the songs of two albums are in the same directory, all songs of
an album should be right next to each others.
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