Added mpd.conf options for disabling automatic resamling, sample
format and channel conversion. This way, users may choose to override
ALSA's automatic resampling, and use libsamplerate instead.
This git branch has become a real MPD fork now. Time to change the
package name to the code name "mpd-mk". Set the version number to
"0.14~git" to mark this as a non-released version.
Don't follow relative symlinks which point into the music directory.
This allows you to organize music with symbolic links, without MPD
managing separate copies of each song.
The mapper library maps directory and song objects to file system
paths. With this central library, the code mixture in path.c should
be cleaned up, and we will be able to add neat features like aliasing.
isMusic() used to be a very inefficient function: with every
invocation, it did another stat() on the specified file. There is
only one caller, do the stat() there manually and use hasMusicSuffix()
instead of isMusic().
By always creating the parent directory, we can use delete_name_in()
without further lookups. The parents which may non exist will be
pruned later. An update request for a non-existing or empty directory
should be quite unusual, so this doesn't add any measurable overhead.
In order to optimize buffer usage, pass only the base file name to
updateInDirectory(). This way, updateInDirectory() may choose when to
allocate a larger buffer for the full path.
It is invalid to pass a path with the wrong dirname to dirvec_find().
To be able to find a subdirectory only by its basename, compare only
the basename of both paths.
The only caller of deletePlaylist() appends PLAYLIST_FILE_SUFFIX, so
we can be sure it's already there. We don't need to stat the file,
since unlink() does all the checking.
Commit 80a2c937 broke resume after pause: it cleared the
input_audio_format when it attempted to simplify a complicated
expression. Don't clear it, just assign input_audio_format if a new
format was specified.
We only need to lock sv->nr changes to prevent traversals ( why
it's called "nr_lock"). free(3) is a "slow" function on my
system; so we can avoid unnecessarily holding a lock long for
longer than needed.
If the sample format isn't supported by the device (i.e. 24 bit on
low-end sound chips), fall back to 16 bit output. There is code in
pcm_utils.c which converts PCM data to 16 bit.
Convert any number of channels to stereo. In fact, this isn't really
stereo, it's rater mono blown up to stereo. This patch should only
make it possible to play 5.1 files at all; "real" conversion to stereo
should be implemented, but for now, this is better than nothing.
In order to be able to deal with non-trivial conversions,
pcm_convertChannels() needs to know both the input and the output
channel count. Simplify buffer allocation in that function.
Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2()
and pcm_convert_channels_2_to_1(). Improved the quality of
pcm_convert_channels_2_to_1() by calculating the arithmetic mean value
of both samples.
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.
When a directory failed to update, it was removed from the database,
without freeing all children and songs (memory leak), and without
locking (race condition). Introduce the functions clear_directory()
and delete_directory(), which do both.
Don't use db_get_directory() and traverse the full path with every
directory being loaded. Just see if the current parent contains the
entry. Everything else would be invalid anyway..
A manipulated database could trigger an assertion failure, because the
parent didn't match. Do a proper check if the new directory is within
the parent's. This uses FATAL() to bail out, so MPD still dies, but
it doesn't crash.
Remove clutter from directory.c. Everything which saves or loads
to/from the hard disk goes to directory_save.c, and code which sends
directory information to the client is moved into directory_print.c.
Having an array with disabled entries sucks. Removed that
DISABLED_SHOUT_ENCODER_PLUGIN macro, and fill the plugin list only
with plugins which are actually enabled. This should be done for all
plugin types.
"volume" was passed as an unsigned integer, which is correct. It's
just that when it was multiplied with the sample value, the whole
operation was changed to unsigned, breaking the algorithm (and Qball's
ears). Internally change "volume" to signed.
With commit 6dcd7fea (if I am not mistaken) the error returned when
you try to save to an existing playlist is wrong. Instead of
MPD_ACK_ERROR_EXIST, MPD_ACK_ERROR_NO_EXIST is returned. This is
obviously wrong and breaks gmpc.
Commit 0bfe7802 broke update for new files in the root directory,
because music_root->path was an empty string and not NULL. There were
some NULL tests missing. Change them to !isRootDirectory(path)
instead of path!=NULL.
