| Commit message (Collapse) | Author | Files | Lines |
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Remove GLib dependency from some headers.
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Invoke decoder_initialized() in the libFLAC metadata callback. This
merges code from the FLAC and the OggFLAC decoder plugin into the
common library.
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This feature has been moved to the "flac" playlist plugin.
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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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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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Remove the audio_format attribute, add "frame_size" instead. The
audio_format initialization and check is moved both to
flac_data_get_audio_format().
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Don't update a float timestamp, this will make imprecisions add up
after a while. We already have the number of the current frame, let's
just calculate the float timestamp from that for every decoder_data()
command. For this, we need to add the attribute "first_frame", for
CUE sheet songs.
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Removed the "bit_rate" attribute from the flac_data struct. Pass the
number of bytes since the last call to flac_common_write(), and let
it calculate the bit rate.
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We don't want to work with floating point values if possible. Get the
integer number of frames from the FLAC__StreamMetadata_StreamInfo
object, and convert it into a float duration on demand. This patch
adds a check if the STREAMINFO packet has been received yet.
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We need this for more exact end-of-subsong detection for CUE files.
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Simple (up-rounding) integer division is good enough. We're casting
the result back to an integer anyway.
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This is a great simplification for flac_common_write(), because we can
convert and submit all of the buffer in one turn. No more partial
buffers with complicated formulas.
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Clean up tag and replay_gain_info there.
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The cue sheet embedded in a flac file doen't contain any information
about track titles and similar. There are three possibilities: Use an
external cue sheet that includes these information, use a tag CUESHEET
with a cue sheet including these information or use tags. I think the
latter is the best option and is already used by other projects.
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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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So far only seekpoints are supported, so no proper tagging yet
except for track number and track length.
Tagging should be done by parsing the cue sheet which
is often embedded as vorbis comment in flac files.
Furthermore the pathname should be configurable like "%A - %t - %T",
where %A means Artist, %t track number and %T Title or so.
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The plugin queries build-time configuration variables, and should
include config.h.
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Free the tag object when it turns out to be empty. This simplifies
several functions and APIs.
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Renamed types, functions, variables.
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Renamed functions and variables.
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"LOG_H" is a macro which is also used by ffmpeg/log.h. This is
ffmpeg's fault, because short macros should be reserved for
applications, but since it's always a good idea to choose prefixed
macro names, even for applications, we are going to do that in MPD.
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Everybody should use struct input_stream.
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These plugins are not input plugins, they are decoder plugins. No
CamelCase in the directory name.
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Don't compile the sources of disabled decoder plugins at all, and
don't attempt to register these.
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chunk_length can be converted to a local variable, because it is
always reset to 0 after it was used.
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There is still a lot of duplicated code in flac_plugin.c and
oggflac_plugin.c. Move code from flac_plugin.c to _flac_common.c, and
use the new function flac_common_write() also in oggflac_plugin.c,
porting lots of optimizations over to it.
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Get rid of CamelCase, and don't use a typedef, so we can
forward-declare it, and unclutter the include dependencies.
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Getting rid of CamelCase; not having typedefs also allows us to
forward-declare the structures.
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(Ab)use the decoder_command enumeration, which has nearly the same
values and the same meaning.
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Similar to the previous patch: pass total_time instead of manipulating
dc->totalTime directly.
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dc->audioFormat is set once by the decoder plugins before invoking
decoder_initialized(); hide dc->audioFormat and let the decoder pass
an AudioFormat pointer to decoder_initialized().
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Moved all of the player-waiting code to decoder_data(), to make
OutputBuffer more generic.
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decoder_initialized() sets the state to DECODE_STATE_DECODE and wakes
up the player thread. It is called by the decoder plugin after its
internal initialization is finished. More arguments will be added
later to prevent direct accesses to the DecoderControl struct.
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The decoder struct should later be made opaque to the decoder plugin,
because maintaining a stable struct ABI is quite difficult. The ABI
should only consist of a small number of stable functions.
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Include only headers which are really required. This speeds up
compilation and helps detect cross-layer accesses.
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We had functions names varied between
outputBufferFoo, fooOutputBuffer, and output_buffer_foo
That was too confusing for my little brain to handle.
And the global variable was somehow named 'cb' instead of
the more obvious 'ob'...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7355 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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All of our main singleton data structures are implicitly shared,
so there's no reason to keep passing them around and around in
the stack and making our internal API harder to deal with.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7354 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This at least makes the argument list to a lot of our plugin
functions shorter and removes a good amount of line nois^W^Wcode,
hopefully making things easier to read and follow.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7353 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7324 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7108 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Both mp4 and (ogg)flac inputPlugins got HTTP inputStream support
later in the game, so their calls to sendDataToOutputBuffer()
didn't get updated to support buffering while the outputBuffer
was full. This fixes it.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6873 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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