buffered_before_play was copied to struct player because it was used
to disable buffering when seeking. Instead of mainaining a copy of
this number, move just the flag to the player struct.
Renamed audio_configFormat to configured_audio_format. Renamed
audio_buffer.format to input_audio_format. Simplified its
initialization in openAudioDevice().
audio.c maintained one of MPD's many layers of audio buffers. It was
without any benefit, since playAudio() can simply send the source
buffer directly to the audio output plugin.
QUEUE adds a new song to the player's queue. CANCEL clears the queue.
These two commands replace the old and complex queueState and
queueLockState code.
Simplify and merge several if clauses before the clearPlayerQueue()
invocation. Call clearPlayerQueue() only if a song is actually
queued; add an assertion for that in clearPlayerQueue().
This variable is superfluous, it is only used to copy its value to
player_control.totalTime. Since the original source of this value
(song->tag->time) will still be available at this point, we can safely
remove fileTime.
Revert e4f5d6bd "re-enable-nonblocking, but sleep if busy".
Non-blocking mode with manual sleeping doesn't help at all (by the
way, the patch should have used snd_pcm_wait() instead of
my_usleep()). ALSA knows much more about the hardware quirks, so we
just let it do the job.
Leftover from the output API changes: oss_open_default() was changed
to return a void*, but it still returned "0" to report success.
Report the OssData pointer instead.
The decoder was woken up after each chunk which had been played. That
caused a lot of superfluous context switches. Wake up the decoder
only when a certain amount of the buffer has been consumed. This
formula is somewhat arbitrary, and has to be proven experimentally.
The mp3 plugin did not use the MAD_NCHANNELS() value correctly: when a
stream was not stereo, it was assumed to be mono, although the correct
number was passed to MPD. libmad doesn't support more than 2
channels, but this change allows gcc to optimize its inlining
strategy.
The dithering function audio_linear_dither() worked for signed 16 bits
only anyway, having a variable "bits" just disables important gcc
optimizations.
A frame contains one sample per channel, thus it is sample_size *
channels. This patch includes some cleanup for various locations
where the sample size for 24 bit audio was still 3 bytes (instead of
4).
There is only once update thread at a time. Make the "modified" flag
global and remove the return values of most functions. Propagating an
error is only useful for updateDirectory(), since updateInDirectory()
will delete failed subdirectories.
The documentation for directory_update_init() was incorrect: a job ID
must be positive, not non-negative. If the update queue is full and
no job was created, it makes more sense to return 0 instead of -1,
because it is more consistent with the return value of isUpdatingDB().
pthread_join() expects a "pointer to a pointer" parameter, but it got
a "pointer to an enum". On AMD64, an enum is smaller than a pointer,
leading to a buffer overflow.
In updateInDirectory(), add new directories immediately and
delete them when they turn out to be empty. This simplifies the code
and allows us to eliminate addSubDirectoryToDirectory().
If the user requests database update during startup, call
directory_update_init(). This should be changed to fully asynchronous
update later.
For this to work, main_notify has to be initialized before db_init().
The algorithm in addDirectoryPathToDB() can be simplified further if
it is combined with the function addParentPathToDB(). Since there is
no other caller of addDirectoryPathToDB(), we can do that. This saves
another large stack buffer.
This recursive function is very dangerous because it allocates a large
buffer on the stack in every iteration. That may be misused to
generate a stack overflow.
