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Adds mixrampdb and mixrampdelay commands. Reads MIXRAP_START and
MIXRAMP_END tags from FLAC files and overlaps instead of crossfading.
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When decoder->timestamp is calculated, the PCM data is already
converted to out_audio_format; using in_audio_format may cause funny
speedups/slowdowns.
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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.
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Use input_stream.uri.
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The replay_gain_state struct holds the precalculated scale factor,
which is removed from struct replay_gain_info.
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This function replaces the replay_gain_info parameter for
decoder_data(). This allows the decoder to announce replay gain
changes, instead of having to pass the same object over and over.
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Seek the decoder to the start of the range before beginning with
playback. Stop the decoder when the end of the range has been
reached. Add the start position to the seek position. Expose the
duration of the range, not the full song file.
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Remove the data_time parameter from decoder_data(). This patch
eliminates the timestamp counting in most decoder plugins, because the
MPD core will do it automatically by default.
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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().
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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.
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If both tags (stream and decoder) are present, we prefer the stream tag.
Fixes #2698, where ICY tag contained useful information, but was
overwritten with bogus decoder tag data.
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Unified function for converting an audio_format object to a string,
for log messages and for the "status" command.
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After we've been hit by Large File Support problems several times in
the past week (which only occur on 32 bit platforms, which I don't
have), this is yet another attempt to fix the issue.
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These two variables are redundant, we need only one of them.
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It's legal to pass decoder=NULL to decoder_read(). Add a check.
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Allocate a decoder_control object where needed, and pass it around.
This will allow more than one decoder thread one day.
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Use GMutex/GCond instead of the notify library. Manually lock the
player_control object before accessing the protected attributes. Use
the GCond object to notify the player thread and the main thread.
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Replace decoder_control.notify with decoder_control.mutex and
decoder_control.cond. Lock the mutex on all accesses to
decoder_control.command and decoder_control.state.
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Don't abort the whole MPD process when the conversion fails. This has
been a denial-of-service attack vector for years.
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Database update was broken due to the dc.pipe!=NULL assertion. This
assertion is only valid while MPD decodes a song, not during database
update.
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dc.pipe must be non-NULL while the decoder thread is running. Ensure
that with a load of assertions.
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When a new song starts playing, send its tag (song->tag) to the music
pipe. This allows output plugins to render tags for all songs, not
only those with embedded tags understood by the decoder plugin.
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This updates the copyright header to all be the same, which is
pretty much an update of where to mail request for a copy of the GPL
and the years of the MPD project. This also puts all committers under
'The Music Player Project' umbrella. These entries should go
individually in the AUTHORS file, for consistancy.
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Turn the music_pipe into a simple music_chunk queue. The music_chunk
allocation code is moved to music_buffer, and is now managed with a
linked list instead of a ring buffer. Two separate music_pipe objects
are used by the decoder for the "current" and the "next" song, which
greatly simplifies the cross-fading code.
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Added music_pipe_allocate(), music_pipe_push() and
music_pipe_cancel(). Those functions allow the caller (decoder thread
in this case) to do its own chunk management. The functions
music_pipe_flush() and music_pipe_tag() can now be removed.
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To aid debugging, print the audio format of the decoder plugin in a
debug message, and print information about PCM conversion.
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The parameter name "wait" overlaps with the POSIX wait() function.
Rename it.
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Now that I've found this nice function in the GLib docs, we can
finally remove our custom sleep function. Still all those callers of
g_usleep() have to be migrated one day to use events, instead of
regular polling.
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The decoder_plugin struct is used by both the MPD core and the decoder
plugin implementations. Move it to a shared header file, to minimize
header dependencies.
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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.
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Removed yet another superfluous buffer layer: return the PCM buffer
from pcm_convert() instead of copying PCM data into the
caller-supplied buffer.
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Copy PCM data to the music_pipe_write() buffer, and apply replay gain
/ normalization to it, instead of manipulating the source buffer.
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Decoder plugins must not send partial frames.
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In NDEBUG, the parameter "decoder" is not used.
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When the decoder thread is waiting for free chunks in the music pipe,
don't ignore the STOP command. Just return dc.command without further
checks.
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Free memory allocated by libsamplerate when the output or the decoder
is closed.
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The tag() method reads a tag from the stream. This replaces the
meta_name and meta_title attributes.
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This patch fixes a minor memory leak: when decoder_tag() attempted to
send a merged tag object (created by tag_add_stream_tags()), and was
interrupted by a decoder command, it did not free the temporary merged
tag object.
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It is illegal to pass an empty audio buffer around. pcm_resample()
sometimes seems to result in 0 samples, maybe related to
libsamplerate. To work around that problem, add special checks after
both pcm_convert() invocations. Removed the pcm_resample()==0 checks
from pcm_convert().
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Refuse to play audio formats which are not supported by MPD.
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