| Commit message (Collapse) | Author | Age | Files | Lines |
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Improves quality by not squeezing 32 bit samples down to 16 bit, and
then back to 32 bit floating point. Reduces CPU usage by skipping a
conversion step.
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If there are multiple pages, the last partial page must be flushed.
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Conflicts:
src/output/osx_plugin.c
src/text_input_stream.c
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Add the encoder_plugin method end(). This is important for the
recorder plugin.
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Don't reset the ogg_stream_state object, because this discards the
end-of-stream packet that was just added.
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For simplicity, the MPD core should not have to deal with packing. It
is rarely used, and those plugins that need it should use the
pcm_export library instead.
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Conflicts:
Makefile.am
NEWS
configure.ac
src/encoder/flac_encoder.c
src/log.c
src/pcm_buffer.c
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Convert to padded 24 bit samples, instead of falling back to 16 bit.
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This fixes a buffer corruption bug; pcm_buffer is not designed to be a
persistent buffers, and will discard anything between two consecutive
calls.
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This fixes a buffer corruption bug; pcm_buffer is not designed to be a
persistent buffers, and will discard anything between two consecutive
calls.
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This fixes a buffer corruption bug; pcm_buffer is not designed to be a
persistent buffers, and will discard anything between two consecutive
calls.
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Conflicts:
src/player_thread.c
src/playlist_control.c
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In the "vorbis" plugin, this is a copy of the old flush() method,
while flush() gets a lot of code remove, it just sets the "flush" flag
and nothing else. It doesn't start a new stream now, which should fix
a few problems in some players.
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These fixes were mostly generated with `codespell' [0] and manually
reviewed.
[0] http://git.profusion.mobi/cgit.cgi/lucas/codespell/
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Conflicts:
Makefile.am
NEWS
configure.ac
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Without the ogg_stream_reset() call, the "e_o_s" flag never gets
reset, and libogg writes EOS packets over and over.
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Remove trailing whitespace found by this command:
find -name '*.[ch]' | xargs grep "[[:space:]]$"
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Conflicts:
NEWS
configure.ac
src/directory.h
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When you don't explicitly set an output sample rate, liblame tries to
guess an output sample rate from the input sample rate. You would
think that this "guessing" consists of just setting both equal, but
that is not the case. For 44.1kHz at 96kbit/s, liblame chooses
32kHz. This patch explicitly configures the output sample rate, to
stop the bad guessing.
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Conflicts:
NEWS
configure.ac
src/input/rewind_input_plugin.c
src/output/httpd_output_plugin.c
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Replaced all occurrences of g_error() with MPD_ERROR() located in a new header
file 'mpd_error.h'. This macro uses g_critical() to print the error message
and then exits gracefully in contrast to g_error() which would internally call
abort() to produce a core dump.
The macro name is distinctive and allows to find all places with dubious error
handling. The long-term goal is to get rid of MPD_ERROR() altogether. To
facilitate the eventual removal of this macro it was added in a new header
file rather than to an existing header file.
This fixes #2995 and #3007.
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This patch prepares support for floating point samples (and probably
other formats). It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.
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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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When using wave encoder with httpd audio output mpd can input this stream via http and audiofile decoder.
This for example opens simple way to configure lossless audio streaming port(like jack or pulseaudio does but without overhead).
Another possibility can be using it for gathering raw data for visualization plugins (If sync issue will be resolved)
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That's an optional method.
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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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When a new tag is set, end the current stream and begin a new one.
Use vorbis_analysis_headerout() to write a full ogg header. This
fixes a problem with icecast: after a song change in MPD, icecast
stops forwarding ogg packets to its clients.
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libvorbis goes into a very long loop if we try to add data after a
flush was invoked by vorbis_analysis_wrote(0). This seems to be a
problem with the internal end-of-stream marker. Thus, we cannot reuse
the vorbis_dsp_state object.
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Copy all tags know to MPD to the vorbis_comment.
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Allocate the vorbis_comment object when it's used. It is not used
anymore in vorbis_encoder_tag().
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Don't reinitialize the encoder with every tag.
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Prepare the removal of vorbis_encoder.vc.
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Converted the ogg_page attribute from the vorbis_encoder struct to a
local function of vorbis_encoder_read(). This simplifies some code,
because we don't need to check the page anymore before using it.
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