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Since version 0.14, MPD has been logging to standard error instead of
standard output. The option name should reflect that. The old option
continues to work, we will remove it in a future MPD release.
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This encoder plugin is a replacement for the LAME encoder plugin for
those who prefer a "free" (non-patent encumbered) encoder library.
Most of the plugin source code is copied from the LAME encoder plugin,
since the LAME and TwoLAME APIs are nearly the same.
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External cue sheet file for "file.flac" should be named as "file.flac.cue".
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According to the ID3 2.4 documentation, "TOPE" is "Original
artist/performer", not "performer". Removed "TOPE" support. Instead,
map TPE3 ("Conductor/performer refinement") and TPE4 ("Interpreted,
remixed, or otherwise modified by") to "performer".
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The tag_id3.c library supports both the documented "TSO2" tag, and the
inofficial TXXX/ALBUMARTISTSORT.
The Vorbis/FLAC decoder automatically supports the new tag, without
further change.
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Initialize flac_data.tag right after flac_data_init(). This way, the
"goto fail" won't jump to the point where tag_free(NULL) can be
called.
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Do all the software volume stuff inside each output thread, not in the
player thread. This allows one software mixer per output device, and
also allows the user to configure the mixer type (hardware or
software) for each audio output.
This moves the global "mixer_type" setting into the "audio_output"
section, deprecating the "mixer_enabled" flag.
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This command has been deprecated more than 5 years ago (0.10.0). Its
implementation is a kludge, let's remove it now.
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On Mac OS X, the httpd plugin cannot be compiled, because OS X's
system headers do nto include sys/types.h, although they use
u_int32_t.
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Some clients have visual feedback for "database update is running".
Using the "database" idle event is unreliable, because it is only
emitted when the database was actually modified. This patch adds the
"update" event, which is emitted when the update is started, and again
when the update is finished, disregarding whether it has been
modified.
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Added the response line "Last-Modified", which sends the modification
time in ISO 8601. The same was already implemented for playlists.
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Don't free an internal configuration value in log_init(). Call
config_get_path() instead of manually calling parsePath().
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When the filesystem_charset is changed in mpd.conf, MPD should discard
the old database. In this error branch, MPD did not fill the GError
object properly, and logged a warning message instead, which caused a
segmentation fault.
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When MPD was paused, and the client sent the "stop" command (or
"clear"), a glitch caused MPD to continue playback for a split second.
This was because audio_output_all_cancel() calls
audio_output_all_update(), which reopens all output devices, and
re-ignites the playback loop.
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At the moment mpd doesn't store or restore the current track to/from
its state file when the daemon is stopped/started while in 'stopped'
state. I believe the preferred behaviour would be to store and
restore the current track even when the daemon is in stopped state
when shutting down.
I made a small patch to adapt this behaviour. If you believe this is
not the preferred behaviour, maybe this should be realized as a
configuration option. I'm not sure how to do this, but made a small
comment, where one would have to put the option.
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Call av_metadata_get() in a loop.
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The top-level "mixer_device" and "mixer_control" options have been
deprecated by MPD 0.15, and it's safe to remove them in MPD 0.16.
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Several users had problems with binding MPD to "localhost". The cause
was duplicate /etc/hosts entries: the resolver library returns
127.0.0.1 twice, and of course, MPD attempts to bind to "both" of
them. This patch makes failures non-fatal, given that at least one
address was bound successfully. This is a workaround; users should
rather fix their /etc/hosts file.
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This allows you to select controls with duplicate names.
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The old global settings "http_proxy_host", "http_proxy_port",
"http_proxy_user" and "http_proxy_password" continue to work.
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When all audio outputs have been closed due to failures, pause the
playback instead of stopping it. This way, the user may resume
at the current position after the problem has been dealt with.
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The "lastfm" input plugin is far from complete, because MPD does not
support nesting playlists yet. The "fluidsynth" decoder plugin
suffers from shortcomings in the libfluidsynth library:
http://www.mail-archive.com/fluid-dev@nongnu.org/msg01099.html
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Even if libsamplerate support is enabled, compile the fallback
resampler. When the user specifies the option
"samplerate_converter=internal", it is chosen in favor of
libsamplerate. This may help users with a weak FPU who don't want to
compile a custom MPD from source, because the fallback resampler does
not use floating point operations.
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After a seek, wait until enough new chunks are decoded before starting
playback. If this takes too long, send silence chunks to the audio
outputs meanwhile.
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This is similar to the MPD 0.14 patch "wait 10 seconds before
reopening a failed device", which only covered open() failures. This
patch adds the same feature for play().
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The default values for buffer_time and period_time were both capped by
the hardware limits on practically all chips. The result was a
period_time which was half as big as the buffer_time. On some chips,
this led to lots of underruns when using a high sample rate (192 kHz),
because MPD had very little time to send new samples to ALSA.
A period time which is one fourth of the buffer time turned out to be
much better. If no period_time is configured, see how much
buffer_time the hardware accepts, and try to configure one fourth of
it as period_time, instead of hard-coding the default period_time
value.
This is yet another attempt to provide a solution which is valid for
all sound chips. Using the SND_PCM_NONBLOCK flag also seemed to solve
the underruns, but put a lot more CPU load to MPD.
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The lastfm input plugin enables MPD to play lastfm:// URLs. This
plugin is not complete yet: it plays only the first song in the
last.fm playlist, and the playlist parser isn't even implemented
properly.
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Added a small RIFF parser library. Look for an "id3" chunk, and let
libid3tag parse it.
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This is the first patch in a series to enable 32 bit audio samples in
MPD. 32 bit samples are more tricky than 24 bit samples, because the
integer may overflow when you operate on a sample.
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Parse the vorbis comments in libflac's metadata_callback and pass them
as tag struct to the decoder API.
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Don't hard code the "bits" parameter to 16. Try to use the input's
sample format, if possible.
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The option "enabled" is on by default. If you specify "enabled no" in
an audio_output section, then this device is disabled by default.
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[mk: adapted to new output plugin API]
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