Commit f78cddb4 introduced a regression: after a song was moved, the
order array was not updated (in random mode). This caused MPD to
think the "current" song has changed when you moved something to the
position of the current song.
Always assume the buffer is empty before calling the encoder. Always
flush the buffer immediately after there has been added something.
This reduces the risk of buffer overruns, because there will never be
a "rest" in the current buffer.
The non-blocking mode of libshout is sparsely documented, and MPD's
implementation had several bugs. Also removed connect throttling
code, that is done by the MPD core since 0.14.
After the state file has been loaded, the playlist version is still
"1", and "plchanges 1" returns the whole playlist. Fix this by
increasing the playlist version after the state file has been loaded.
Don't call syncPlaylistWithQueue() in nextSongInPlaylist() and
previousSongInPlaylist(). This is a relic from the time when there
was no event, and was a workaround to the timing problem.
Export the "g_playlist" variable, and pass it to all playlist
functions. This way, we can split playlist.c easier into separate
parts. The code which initializes the singleton variable is moved to
playlist_global.c.
Before every operation which modifies the playlist, remember a pointer
to the song struct. After the modification, determine the "next song"
again, and if it differs, dequeue and queue the new song.
This removes a lot of complexity from the playlist update code, and
makes it more robust.
The "current" variable is used for calculating the seek destination,
and was declared as "int". With very long song files, the 32 bit
integer can overflow. ffmpeg expects an int64_t, which is very
unlikely to overflow. Switch to int64_t.
If avcodec_decode_audio2() returns no output for an AVPacket,
libavcodec may buffer some data, and return a larger chunk of output
later. This patch disables a lot of bogus warnings.
Output the name of the codec as a debug message. During my tests,
ffmpeg never filled this struct member, but it may do so in the past,
and this debug message might become helpful.
The shout_mp3 encoder had two bugs: when no song was ever played, MPD
segfaulted during cleanup. Second bug: memory leak, each time the
shout device was opened, lame_init() was called again, and
lame_close() is only called once during shutdown.
Fix this by shutting down LAME each time the clear_encoder() method is
called.
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().
When the playlist was loaded from the state file, the order numbers
were the same as the positions. In random mode, we need to shuffle
the queue order. To accomplish that, call setPlaylistRandomStatus()
at the end of readPlaylistState(), and do a fresh shuffle.
When MPD is not playing while in random mode, and the client issues
the "clear" command, MPD crashes in stopPlaylist(), or more exactly,
in queue_order_to_position(-1). Exit from stopPlaylist() if MPD isn't
playing.
PlaylistInfo() (notice the capital 'P') sends a stored playlist to the
client. Move it to a separate library, where all the code which glues
the playlist and the MPD protocol together will live.
The playlist.c source is currently quite hard to understand. I have
managed to wrap my head around it, and this patch attempts to explain
it to the next guy.
The function playPlaylistIfPlayerStopped() is only called when the
player thread is stopped. Converted that runtime check into an
assertion, and remove one indent level.
One of the previous patches removed the "random" mode check from
nextSongInPlaylist(), which caused a shuffle whenever MPD wrapped to
the first song in "repeat" mode. Re-add that "random" check.
In playPlaylist(), the second "song==-1 && playing" check can never be
reached, because at this point, the function has already returned
(after unpausing).
When a song is deleted, start playing the next song immediately,
within deleteFromPlaylist(). This allows us to remove the ugly
playlist_noGoToNext flag, and the currentSongInPlaylist() function.
By calling queue_next_order() before playlist.current is invalidated
(by the deletion of a song), we get more robust results, and the code
becomes a little bit easier. incrPlaylistCurrent() is unused now, and
can be removed.
The function shuffles the virtual order of songs, but does not move
them physically. This is used in random mode.
The new function replaces playlist.c's randomizeOrder() function,
which was aware of playlist.current and playlist.queued. The latter
is always -1 anyway, and the former as preserved by the caller, by
converting playlist.current to a position, and then back to an order
number.
Add a "changed" check to setPlaylistRepeatStatus(): when the new
repeat mode is the same as the old one, don't do anything at all. No
more checks, no "idle" event.
When the random mode is toggled, MPD did not clear the queue. Because
of this, MPD continued with the next (random or non-random) song
according to the previous mode. Clear the queued song to fix that.
The function moveSongInPlaylist() attempted to read the position of
the current song, even if it was -1. Check that first. The same bug
was in shufflePlaylist().
The null plugin synchronizes the playback so it will happen in real
time. This patch adds a configuration option which disables this: the
playback will then be as fast as possible. This can be useful to
profile MPD.
It is possible that playlist.current is reset before the TAG event
handler playlist_tag_event() is called. Convert the assertion into a
run-time check.
Break from the loop instead of returning the function. This calls
player_stop_decoder(), which in turn emits the PLAYLIST event. This
allows the playlist to re-start the player.
Don't attempt to restart the player if it was stopped, but there were
still songs left on the playlist. This looks like it has been a
workaround for a bug which has been fixed long time ago.
The player_thread loop requests the next song from the playlist as
soon as the decoder finishes the song which is currently being played.
This is superfluous, and can lead to synchronization errors and wrong
results. The playlist already knows when the player starts playing
the next song (player_wait_for_decoder() triggers the PLAYLIST event),
and will then trigger the scheduler to provide the next song.
The "TAG" event is emitted by the player thread when the current
song's tag has changed. Split this event from "PLAYLIST" and make it
a separate callback, which is more efficient.
The "sticker" command allows clients to query or manipulate the
sticker database. This patch implements the sub-commands "get" and
"set"; more will follow soon (enumeration), as well as extended
"lsinfo" / "playlistinfo" versions.
When a song is deleted from the database, remove its sticker, too.
What's still missing is some sort of garbage collector after a fresh
database create (--create-db).
"Stickers" are pieces of information attached to existing MPD objects
(e.g. song files, directories, albums). Clients can create arbitrary
name/value pairs. MPD itself does not assume any special meaning in
them.
If a song is not within the music directory ("file:///..."), it has no
"parent directory". The archive code nonetheless dereferences the
parent pointer, causing a segmentation fault. Check parent!=NULL.
One of the previous patches made MPD consume 100% CPU in a busy wait:
when the music_pipe was full, it did not wait (with notify_wait()) for
free chunks, because a variable has a different meaning now. Always
pass "true" as the "wait" parameter.
Some plugins used the APE or ID3 tag loader as a fallback when their
own methods of loading tags did not work. Move this code out of all
decoder plugins, into song_file_update().
This new API gives the caller a writable buffer to the music pipe
chunk. This may allow the caller to eliminate several buffer copies,
because it may manipulate the returned buffer, until it calls
music_pipe_expand().
When libvorbis knows that a song is seekable, it seeks around like
crazy in the file before starting to decode it. This is very
expensive on remote HTTP resources, and delays MPD for 10 or 20
seconds.
This patch disables seeking on remote songs, because the advantages of
quickly playing a song seem to weigh more than the theoretical ability
of seeking for most MPD users. If users feel this feature is needed,
we will make a configuration option for that.
getBoolConfigParam() returns an int. It is not possible to check for
CONF_BOOL_UNSET after it has been assigned to a bool; use a temporary
int value for that.
Calling input_curl_select() after EOF has been reached causes an
assertion failure. This can happen if the HTTP response is empty.
Check c->eof before calling input_curl_select().
Path names in the directory and song structs are always encoded in
UTF-8. Don't use strcmp(), it cannot handle UTF-8 characters
properly. Use GLib's UTF-8 aware g_utf8_collate() function for that.
I was having problems with shoutcast stream outputs before applying
the attached patch, which enlarges the shoutcast output
buffer. Ideally, this should be configurable, but this resolves the
issue for my needs.
Max Kellermann