When the update thread is started before MPD has forked (for
daemonization), it is killed, because threads do not survive a fork().
This induces an inconsistent state where MPD won't start any update
thread at all, because it thinks the thread is already running.
Move the "while" loop which checks for commands to the caller
ao_pause(). This simplifies the pause() method, and lets us remove
audio_output_is_pending().
If no ports are configured, don't overwrite the (NULL) configuration
with the port names of the first JACK server. If the server changes
after a JACK reconnect, MPD won't attempt to auto-detect again.
Currently, the JACK plugin manipulates the audio_format struct which
was passed to the open() method. This is very likely to break,
because the plugin must not permanently store this pointer. After
this patch, MPD ignores sample rate changes. It looks like other
software is doing the same, and I guess this is a non-issue.
This patch converts the audio_format pointer within jack_data into a
static audio_format struct.
Hi -
independently of libmikmod's other problems - there seems
to be a problem in mpd's wrapper: MikMod_Exit() is called
after the first file is decoded, which frees some ressources
within the mikmod library. An attempt to play a second file
leads to a crash. The appended patch fixes this for me.
(I don't know what the "dup" entry is good for - someone
who knows should review that too.)
best regards
Matthias
[mk: removed 3 more MikMod_Exit() invocations]
When there are duplicate slashes in the song paths, eliminate them;
example:
/var/lib/mpd/music//foo.mp3
becomes:
/var/lib/mpd/music/foo.mp3
The slash is only detected at the border between the music_directory
and the local part.
When the user configures a music_directory with a trailing slash, it
may break playlist loading, because MPD expects a double slash. Chop
off the trailing slash.
ffmpeg_tag_internal() does not look for a few tags that mpd
supports. Most noteably:
comment -> TAG_ITEM_COMMENT -> Description
genre -> TAG_ITEM_GENRE -> WM/Genre (not WM/GenreID)
year -> TAG_ITEM_DATE -> WM/Year
I *think* that this is the last of the tags that AVFormatContext() in
ffmpeg supports that mpd also uses.
Make those two methods optional to implement, and let input_stream.c
provide fallbacks. The buffer() method will be removed one day, and
there is now only one implementation left (input_curl.c).
The open_stream() method opens the input_stream. This allows the
archive plugin to do its own initialization, and it also allows it to
use input_stream.data. We can remove input_stream.archive now, which
was unnatural to have in the first place.
Preparation for supporting other channel numbers than stereo: use
loops instead of duplicating code for the second channel. Most
likely, gcc will unroll these loops, so the binary won't be any
different.
This patch implements the MMS protocol, by using libmms. It is quite
experimental: it does not support seeking yet, and it is currently
using synchronous I/O, which causes MPD to hang while waiting for the
server.
When the playlist is cleared, pc.errored_song is also cleared. This
causes pc_errored_song_uri() to crash, because it assumes that
pc.errored_song is set. Reset pc.error to fix that assumption.
When waiting for free space in the ring buffer, the JACK plugin
sleeped 10ms until there is enough space. This delay was too large
for low-latency setups (<10ms), and created a lot of xruns. Work
around that by reducing the sleep time to 1ms.
A proper solution for this would be to use an event based approach,
and we will do it, just not now.
When the connection failed once, you had to restart MPD, because it
never cleared the jack_data.shutdown flag. Instead, it refused to
play anything "because there is no client thread" (which is wrong at
that point).
GIOChannel is more portable than raw read()/write() calls. We're
using GIOChannel anyway, because we need it for plugging the client
into the GLib main loop.
Configure the GIOChannel to the bare minimum: no character set, no
buffering.
On some platforms, g_free() must be used for memory allocated by
GLib. This patch intends to correct a lot of occurrences, but is
probably not complete.
Both methods are always called together. There is no point in having
them separate. This simplifies the code, because the old configure()
method could be called more than once, and had to free old
allocations.
Reimplemented the legacy mixer configuration: copy the deprecated
configuration values into the audio_output section. Don't configure
the mixers twice (once for the audio_output, and a second time for the
legacy values).
This requires volume_init() to be called before initAudioDriver().
Return the default value in the conf_get_block_*() functions when
param==NULL was passed.
This simplifies a lot of code, because all initialization can be done
in one code path, regardless whether configuration is present.
Two bugs here led to a large number of interrupts being generated on the
sound card when ALSA output is being used. Because we specify no default
period_time, the sound card gives us 3000 interrupts/sec rather than a more
sane 20 or 30. This completes the revert of dd7711 already started by
4ca24f.
The larger bug was in the change to config_get_block_unsigned() and using 0
as the default value for both 'buffer_time' and 'period_time'. This means
any pre-setting of these options in newAlsaData() gets wiped out. Add a new
default for period_time, and ensure default values for buffer_time and
period_time are used if none are provided by the user.
Signed-off-by: Dan McGee <dan@archlinux.org>
[mk: set defaults in newAlsaData() to fix auto-configuration; renamed
"_MS" back to "_US" because ALSA expects microseconds, not milliseconds]
Signed-off-by: Max Kellermann <max@duempel.org>
Added all important id tags from the MusicBrainz wiki:
http://musicbrainz.org/doc/MusicBrainzTag
This should automatically enable its suport in the vorbis and flac
decoder plugins.
The input_stream API sets size to -1 when the size of the resource is
not known. The modplug decoder checked for size==0, which would be an
empty file.
You are allowed to call decoder_read() with decoder==NULL. It is a
convenience function provided by the decoder API. Don't manually fall
back to input_stream_read().
