playback is stopped completely. This means the player process will no
longer have to wake up 100 times per second to see if it's been told to
start playing (the main process will just spawn a new player process when
it needs to). On the downside, this means an extra pair of forks() and the
re-initializing of the player and decode processes each time playback is
restarted.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6446 09075e82-0dd4-0310-85a5-a0d7c8717e4f
one now, and trying to call NULL was causing a segfault at exit.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6398 09075e82-0dd4-0310-85a5-a0d7c8717e4f
uninitialized variables and non-returning functions that return. Let's
tell it to stfu.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6277 09075e82-0dd4-0310-85a5-a0d7c8717e4f
because lsr may return less than the input buffer size, and the rest of the
audio code needs to know the new size. This fixes the clicking that was
introduced with recent changes to the lsr code. A huge thanks to remiss
for figuring this out.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6273 09075e82-0dd4-0310-85a5-a0d7c8717e4f
heavily modified both sets of code, and I think it's useful for people to
know who's responsible/familiar with what.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6272 09075e82-0dd4-0310-85a5-a0d7c8717e4f
relative paths in the DB or URLs. The main functional difference is that
playlistmove and playlistdelete will rewrite playlists in the correct
encoding and remove invalid lines instead of potentially modifying them.
The specific changes are:
appendSongToStoredPlaylist:
* Don't convert to FS charset
* Don't prepend music_directory if saving absolute paths
writeStoredPlaylistToPath:
* Convert to FS charset
* Prepend music_directory if saving absolute paths
loadStoredPlaylist:
* Make sure each line is either in the DB or a URL
loadPlaylist:
* Don't bother checking paths, since it's done in loadStoredPlaylist now
git-svn-id: https://svn.musicpd.org/mpd/trunk@6266 09075e82-0dd4-0310-85a5-a0d7c8717e4f
audio at once, so it won't work for us. The old full API code was still
heavily broken, as each call to pcm_convertSampleRate() used the same
state, even if it was processing two streams of audio. The new code keeps
a separate state for each audio stream that's being converted.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6255 09075e82-0dd4-0310-85a5-a0d7c8717e4f
number of channels is specified when the converter state is created.
Previously this was only done once, thus breaking horribly when the input
audio suddenly had a different channel count. A new state could be created
every time the number of channels changes, but this could happen many times
a second if resampling to two different formats at once. The simple API
doesn't have this problem, as channel count is an argument to the
conversion function itself.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6229 09075e82-0dd4-0310-85a5-a0d7c8717e4f
and samplerate conversion. This makes the code much easier to read, and
fixes a few bugs that were previously there.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6224 09075e82-0dd4-0310-85a5-a0d7c8717e4f
playlistadd, playlistdelete, etc. and would've caused the playlist to be
rewritten only up to the line with the error.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6133 09075e82-0dd4-0310-85a5-a0d7c8717e4f
