| Commit message (Collapse) | Author | Age | Files | Lines |
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Support 32 bit samples with software mixer.
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Conflicts:
NEWS
configure.ac
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The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
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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?).
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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.
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The new option "sample_rate" sets the sample rate for libmikmod.
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Remove the OPEN_2CH_MAX option. MPD's support for surround sound is
still clunky, but we're working on it.
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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).
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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.
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The oggflac plugin has been completely broken for quite a while and
nobody has noticed - maybe we should remove it?
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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.
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Conflicts:
src/input/lastfm_input_plugin.c
src/song_save.c
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We need the function zzip_file_stat().
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We're using API functions which are not available in 0.3.
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This function was not present in SQLite < 3.4.
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Should be "lastfm_user", not "lastfm_username".
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The line buffer had a fixed size of 5 kB, and was allocated on the
stack. This was too small for some users. As a hotfix, we're
increasing the buffer size to 32 kB now, allocated on the heap. In
MPD 0.16, we'll switch to dynamic allocation.
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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().
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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.
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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.
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Don't hold a file descriptor on root's tty when syslog is used for
logging.
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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.
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Be more clear which kind of port should be configured here.
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This new plugin parses extm3u files. Files without the "#EXTM3U"
header are still parsed by the plain old "m3u" plugin.
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Implement the methods enable() and disable(). Bind the HTTP port in
the enable() method, but reject all incoming connections until the
output is opened.
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When MPD plays a mono song (audio_format.channel==1), connect only one
source port to both destination ports.
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After playback has stopped, the ring buffers may still contain
samples. These will be played when playback is started the next
time. We should clear the buffers each time.
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jack_client_new() is deprecated. This requires libjack 0.100
(released nearly 5 years ago). We havn't been testing older libjack
versions anyway.
As a side effect, there is the new option "autostart".
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Instead of using MPD's audio output name (setting "name"), use a
separate configuration option. Change the default to "Music Player
Daemon".
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Store a list of supported tag items in the database. When loading a
database which does not have a matching list, we must rescan in order
to get the missing information.
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Use a single GString buffer object in all functions loading the
database. Enlarge it automatically for long lines. This eliminates
the maximum line length for tag values. There is still an upper limit
of 512 kB to prevent denial of service, but that's reasonable I guess.
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Conflicts:
NEWS
configure.ac
src/decoder/ffmpeg_plugin.c
src/update.c
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Convert the metadata with the libavformat function av_metadata_conv().
This ensures that canonical tag names are provided by libavformat, and
we can remove the "artist" vs "author" workaround.
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When you disable the "follow_outside_symlinks" or the
"follow_inside_symlinks" setting, the next update should remove the
now-ignored files from the database.
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Basically the same as the 0.15.5 patch "check again if output is open
on CANCEL". Same race condition, same fix.
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When the songs of two albums are in the same directory, all songs of
an album should be right next to each others.
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Always keep the audio_output object locked within the output thread,
unless a plugin method is called. This fixes several race conditions.
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The recovery is for nothing if we get CLOSE afterwards. Let's not
recover in the cancel() method, and let the next play() call sort it
out.
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Based on the XSPF playlist plugin.
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An asterisk means that this attribute should not be enforced, and
stays whatever it used to be. This way, some configuration values
work like masks.
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.. and keep up the JACK connection while MPD runs. Allocate the ring
buffers on the first open, and free them at MPD exit.
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Don't disconnect from JACK during pause.
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This is a complete rewrite of the PulseAudio output plugin. It uses
the asynchronous API, which gives us more control over everything.
Additionally, it connects to the PulseAudio server on startup, and
keeps this connection up while MPD runs. During pause, instead of
closing the stream, it enables "cork".
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This looks nicer in the PulseAudio manager than just "mpd".
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