| Commit message (Collapse) | Author | Age | Files | Lines |
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.. rather then append to the end of the previous one
Cuebreakpoints from the cuetools package has three modes of operation,
and the default is to append pregap (INDEX 00) to the end of the
previous track. This is the behavior most compliant to the existing
cue files.
Here is the patch which fixes the issue. I borrowed bits of
implementation from cuebreakpoints. I assumed that the whole audio
file must be covered by head-to-head going tracks, which is how
hardware CD players probably work. In cue_tag I changed rounding from
rounding up to rounding down because the thing in mpd which calculates
actual track duration (and current position) rounds it down, and I
didn't want to see in my playlist values different from whose in a
now-playing progress bar.
I've compared the resultant mpd behaviour with "mplayer -ss MM:SS.MS"
where the time was supplied by cuebreakpoints and noticed that mplayer
started each track a bit earlier then mpd, though this was the same
before the patch.
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Conflicts:
NEWS
configure.ac
src/cue/cue_tag.c
src/decoder/mpcdec_decoder_plugin.c
src/player_thread.c
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"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'"
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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.
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Like tag_merge(), but can deal with NULL parameters, and frees both
tag objects.
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"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"
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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.
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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().
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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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For int16_t.
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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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When handle_update() was modified to use uri_safe_local(), suddently
"mpc update ''" and "mpc update '/'" stopped working, because both are
not a "safe" local URI. This patch adds a special case for these, to
retain backwards compatibility.
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Did you ever accidently click "stop" while feeding a radio station?
This option sets the output device to "pause" to disable the "close"
method. It falls back to "pause" then, which is specific to the
plugin. Some plugins implement it by feeding silence.
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If we're not doing this, and a new song is played after pause ends,
then you will hear the rest of the previous song.
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Always use the same number of samples from each channel's ring
buffer. This ensures that all channels are kept in sync.
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Pass everything to the GLib logging library. No direct stderr access.
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The pointer is invalid if av_open_input_file() fails.
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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.
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Dots at the beginning of an URI segment are ok, as long as the special
names "." and ".." are not used.
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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.
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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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Don't allocate each replay_gain_info object on the heap. Those
objects who held a pointer now store a full replay_gain_info object.
This reduces the number of allocations and heap fragmentation.
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The previous patch not only moved code, it also changed the check.
Negative gain values seem to be valid after all, there just was the
"magic" value 0.0 which means "not available". This patch changes the
"magic" value to "INFINITY", and uses the C99 function isinf() to
check. It might have been a better idea to use "NAN", but the "NAN"
macro is a GNU extension.
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Moved (and renamed) the function calc_replay_gain_scale() to
replay_gain_info.c.
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This function determines whether replay gain data is available.
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Same for "listplaylistinfo".
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The only "return" statement always returns 0.
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Pass the current URI to wavpack_open_wvc().
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Don't use the function ffmpeg_helper(), don't initialize the codec.
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Fix a memory leak in some code paths.
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Use input_stream.uri.
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All close() implementations must call this method.
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Conflicts:
Makefile.am
NEWS
configure.ac
src/decoder/ffmpeg_decoder_plugin.c
src/decoder_thread.c
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Taken from the ffmpeg sources.
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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.
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