| Commit message (Collapse) | Author | Age | Files | Lines |
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It makes no difference right now, but we're about to add an endianness
flag and will want to make sure it's correctly initialised every time.
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The first patch by Patrick didn't work, because his "#ifdef HAVE_OSX"
line would have required config.h.
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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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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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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 the PAUSE loop ends, re-check the next command before calling
ao_play() again.
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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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The GLib functions are more portable.
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Added another glue function in main().
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The glue_*() functions act as a glue between MPD's main() function and
its libraries. They handle disabled features, and pass validated
configuration options.
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No CamelCase.
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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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Renamed type, variables and functions.
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Add the "const" attribute to functions when their return value only
depends on parameters. This allows gcc to eliminate some function
calls.
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Parse the state file line by line, let each subsystem probe a line.
Only the playlist_state code gets the FILE pointer to read the
following lines.
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Return early from the destructor function when there is no configured
state file. Don't check the timer, don't call g_free(NULL).
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Print "Loading" instead of "Saving" in state_file_read(). Added debug
message to state_file_write().
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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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Those were only wrappers for playlist_state_{save,restore}(). Since
sf_callbacks has been removed, we can call the latter functions
directly.
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There are very few callbacks, and they are not meant to be pluggable.
Let's eliminate the array and call the load/save functions manually.
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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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The variables "success" and "error" are only used if SQLite support is
enabled.
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Very few lines to log a song URI when it has been entirely played.
Then mpd logs can be parsed to do statistics.
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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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The special-purpose function is used for saving/restore the software
volume control to the state file.
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This mixer plugin may be used instead of the traditional global
software mixer. It integrates with the "volume" filter plugin, and
can control the software volume of an audio output which has no
hardware mixer.
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Converted the range checks in volume_level_change() to assertions.
Changed all volume types to "unsigned", expect for those which must be
able to indicate error (-1).
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Since the "volume" command has been removed, nobody uses relative
volumes anymore.
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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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The "convert" filter must be the last filter in the chain. Ensure
that by doing its initialization at the very end of
audio_output_init().
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When song->mtime was not initialized properly, it was revealed that
strftime() might crash when gmtime_r() returns NULL due to an invalid
time_t input value.
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This patch adds initial filter support for audio outputs. Each audio
output gets a "filter" attribute, which is used by ao_play_chunk().
The PCM conversion is now performed by convert_filter_plugin.
audio_output.convert_state has been removed.
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REOPEN is called when the input audio format changes. The output
thread may be reconfigure the PCM converter.
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