I took this tag name from a MusePack sample file I got from a user.
It is not documented in the APE specification:
http://wiki.hydrogenaudio.org/index.php?title=APE_key
People seem to be using undocumented extensions to the specification
anyway, and the best we can do is attempt to support them.
I took these tag names from a MusePack sample file I got from a user.
These are not documented in the APE specification:
http://wiki.hydrogenaudio.org/index.php?title=APE_key
People seem to be using undocumented extensions to the specification
anyway, and the best we can do is attempt to support them.
Without libid3tag, we were trying to skip the ID3 frame (since
0.15.2). Its length however was not calculated at all, we were just
dropping everything from the current input buffer. This lead to the
first few seconds of the file being skipped. This patch attempts to
calculate the ID3v2 frame size with the formula from:
http://www.id3.org/id3v2.4.0-structure 3.1 and 6.2
"When playing musepack files with mpd v0.15.8, rg seems to have no effect.
Using sample file below, mpd says 'computing ReplayGain album scale with gain 122.879997, peak 0.549150'.
One thing though, if I build mpd against old libmpcdec-1.2.6, rg works
as expected: 'computing ReplayGain album scale with gain 16.820000,
peak 0.099765'"
Previously, tags of the new song being cross-faded in were sent
immediately. That can cause wrong information being displayed,
because the "previous" song might send its tag at the end again,
overriding the "next" song's tag. This patch saves & merges the tag
of the next song, and sends it when cross-fading is finished, and the
next song really starts.
"There is a bug in fixed-point musepack (musepack_src_r435) playback.
In floating-point audio is OK but in fixed audio is distorted. I have
made a patch for this"
With single+repeat enabled, it is expected that MPD repeats the
current song over andd over. With random mode also enabled, this
didn't work, because the song order was shuffled internally. This
patch adds a special check for this case.
This is a very basic check, which only ensures that the path does not
begin with a slash, doesn't have double slashes and the special names
"." and ".." are forbidden.
Removed the decoder_command_finished() call at the end of
mp3_decode(). This is invalid, because decoder_command_finished() has
already been called in mp3_read().
Add an option for each audio output which enables the use of the
hardware mixer, instead of the software volume code.
This is hardware specific, and assumes linear volume control. This is
not the case for hardware mixers which were tested, making this patch
somewhat useless, but we will use it to experiment with the settings,
to find a good solution.
Apply the replay gain in the output thread. This means a new setting
will be active instantly, without going through the whole music pipe.
And we might have different replay gain settings for each audio output
device.
When all plugins have failed, MPD used to fall back to the "mad"
decoder plugin, to handle those radio streams without a Content-Type
response header. This however leads to unexpected results (garbage
being played) when the stream isn't really mp3. Since we care little
about "bad" streams, we shouldn't have hacks which have bad side
effects.
Let's get rid of this hack now! Only try to "mad" plugin if there was
no match at all (Content-Type, path suffix) and no other plugin has
been tried.
To allow libavformat to detect the format of the input file, append
the suffix of the input file to the URL of the virtual stream. This
specifically enables the "shorten" codec, which is supported by
libavformat/raw.c, detected only by the suffix.
The patch "input/file: don't fall back to parent directory" introduced
a regression: when trying to play a CUE track, decoder_run_song()
tries to open the file as a stream first, but this fails, because the
path is virtual.
This patch fixes decoder_run_song() (instead of reverting the previous
patch) to accept input_stream_open() failures if the song is a local
file. It passes the responsibility to handle non-existing files to
the decoder's file_decode() method.
This replaces the rewinding buffer code from the CURL input plugin.
It is more generic, and allows rewinding even when the server sends
Icy-Metadata (which would have been too difficult to implement within
the CURL plugin).
This is a rather complex patch for the stable branch (v0.15.x), but it
fixes a serious problem: the "vorbis" decoder plugin was unable to
play streams with Icy-Metadata, because it couldn't rewind the stream
after detecting the codec (Vorbis vs. FLAC).
When collecting tag values for the result set, add all of a song's tag
values of the searched type. This affects the "list" command.
Previously, "list" only considered the first tag value of a song.
Don't clear the music pipe when seeking has failed - check the
"seeking" flag instead of "command==SEEK". Clear the "seeking" flag
in decoder_seek_error().
Use the plugin instead of the glue code in normalize.c. This is used
wrapped inside a "autoconv" filter, to enable normalization for all
input file formats.
Fixes a memory leak: the "archive" input plugin opens the archive, but
never closes it. This patch moves the responsibility for doing that
to archive_plugin.open_stream(). This is an slight internal API
change, but it is the simplest and least intrusive fix for the memory
leak.
This fixes an inconsistency in the stored playlist subsystem: when
obtaining the list of playlists (listplaylist, listplaylistinfo), the
file names in the playlist directory are converted to UTF-8 (according
to filesystem_charset), but when saving or loading playlists, the
filesystem_charset setting was ignored.
The plugin code tried to force libavcodec to supply stereo samples.
That however has never actually worked. By removing this code, we are
able to play surround files for the first time.
Allow RIFF/AIFF ID3 tags up to 4 MB (old limit was 256 kB). This
might still be too small for some users, and when somebody complains,
we might do something more clever (like streaming the data into
libid3tag?).
On some platforms, libavcodec wants the output buffer aligned to 16
bytes (because it uses SSE/Altivec internally). It will segfault when
you don't obey this rule.
MPD has been supporting 32 bit samples since version 0.15. This patch
changes one check, and removes the 32->24 conversion code.
Note that WavPack floating point samples have 32 bits, and MPD doesn't
have a special check for floating point - therefore, this WavPack
plugin still returns 24 bit integer samples as before (until we have
float support in the MPD core).
If flac_container_decode() gets a seek destination which is out of
range, it ignores the SEEK command (never finishes it). This leads to
MPD lockup, because the player thread waits for completion.
Drop the required GLib version from 2.16 to 2.12, because many current
systems still don't have GLib 2.16. This requires several new
compatibility functions in glib_compat.h.
ALSA passes full period buffers to the hardware. If an application
doesn't finish writing a period, libasound will nonetheless send the
partial buffer (with undefined trailing data). This causes noise at
the end of playback. This patch attempts to track the current
position within the period buffer, and generates silence at the end,
before calling snd_pcm_drain().
When there's no queued song, and the current one has finished playing,
first make sure that the hardware outputs have really finished playing
the last chunk: call the drain() method in all audio outputs. Without
this patch, MPD stopped playback shortly before the ALSA sound card
had finished playing.
Added the "fd_util" library, which attempts to use the new thread-safe
Linux system calls pipe2(), accept4() and the options O_CLOEXEC,
SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is
not thread safe.
This is particularly important for the "pipe" output plugin (and
others, such as JACK/PulseAudio), because we were heavily leaking file
descriptors to child processes.
Same as the previous patch: create up to 16 configured source ports.
The plugin tries to do its best at guessing the right combination for
the given input file, the number of source and destination ports.
Max Kellermann