state (when users press stop, previous snd_pcm_drop(), then
snd_pcm_drain() was called. this would lockup dmix)
git-svn-id: https://svn.musicpd.org/mpd/trunk@6517 09075e82-0dd4-0310-85a5-a0d7c8717e4f
completely stopped. Instead, send them SIGSTOP to pause the process until
they're needed again. Then send them SIGCONT instead of re-spawning them.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6485 09075e82-0dd4-0310-85a5-a0d7c8717e4f
some versions of shoutcast send "content-type" in all lowercase, and I
don't trust other servers to get the case right for the rest of the headers
we look for.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6482 09075e82-0dd4-0310-85a5-a0d7c8717e4f
have any effect until the aac and mp4 input plugins actually support a
stream decoding API.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6481 09075e82-0dd4-0310-85a5-a0d7c8717e4f
been redirected. This prevents zeroconf from blocking daemonization, and
makes sure any errors get sent to the logs and not stdout.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6477 09075e82-0dd4-0310-85a5-a0d7c8717e4f
header. While this is odd for an HTTP header, it's actually quite common
for streaming clients to send it without a space. Some clients do send
with a space as well, but without one has always worked fine and may in
fact be more compatible.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6472 09075e82-0dd4-0310-85a5-a0d7c8717e4f
silence a warning about an unused variable without using stupid checks for
HAVE_AVAHI || HAVE_BONJOUR.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6471 09075e82-0dd4-0310-85a5-a0d7c8717e4f
I couldn't test mDNSResponder support on Linux, as Debian doesn't include it - but should work as well.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6453 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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
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
