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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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When calculating the properties of the frame, use sample_rate and
other information from the frame header instead of the stored
audio_format object.
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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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Make the function more generic by not passing "struct flac_data" to
it.
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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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That function diverts into various bit formats; it doesn't need a
typed pointer.
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Don't use audio_format_sample_size() for identifying the sample
format.
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The function flac_cue_track() first calls FLAC__metadata_object_new(),
then overwrites this pointer with FLAC__metadata_get_cuesheet(). This
allocate two FLAC__StreamMetadata objects, but the first pointer is
lost, and never freed.
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The FLAC replaygain parser used the "||" operator. This made the code
stop after the first value which was found.
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When one metadata check fails, return quickly. This removes 2 levels
of indent.
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This belongs into "git annotate" or AUTHORS.
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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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On 2009/03/17 Max Kellermann<max@duempel.org> wrote:
> There doesn't seem to be an "official" standard. I'd say: search for
> TITLE[1] first (the most explicit form), then TITLE1, and finally fall
> back to TITLE. This makes sure MPD supports every possible standard,
> without breaking.
I've also added some additional checks to make sure entry is long
enough.
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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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Parse the vorbis comments in libflac's metadata_callback and pass them
as tag struct to the decoder API.
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Renamed numOfItems to num_items.
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This patch allows mpd to recognise the albumartist tag in the way
foobar2000 and others write it to files.
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Splitted flac_copy_vorbis_comment() into flac_copy_comment() and
flac_copy_comment().
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Simplify flac_copy_vorbis_comment() by moving the comment
identification code out.
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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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Use GLib's G_GNUC_UNUSED instead of gcc.h's mpd_unused.
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Having an array instead of individual variables allows the use of the
replay_gain_mode enum as an array index.
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Renamed functions and variables.
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Don't pass the "seekable" flag with every decoder_data() invocation.
Since that flag won't change within the file, it is enough to pass it
to decoder_initialized() once per file.
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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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The last bit of CamelCase in audio_format.h. Additionally, rename a
bunch of local variables.
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Do full C99 integer type conversion in all modules which were not
touched by Eric's merged patch.
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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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Since flacSendChunk() is a trivial function and is only used in one
location, move the code there. The advantage is that calling
decoder_data() directly returns the decoder_command value, so we can
eliminate one decoder_get_command() call.
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Support for bit rates except 16 bits (and 8 bits on little endian) has
always been broken. Since we added optimized functions for 8, 16,
24/32 bits, we can remove the generic flac_convert() function.
Instead of removing it, convert it to a wrapper function for
flac_convert_*().
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Same optimization for 8 and 32 bit files, like the previous patch for
16 bit. Along the way, this patch adds 24 bit FLAC support!
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flac_convert_16() runs a lot faster than the generic (and quite buggy)
function flac_convert(). flac_convert_16() is only used for
non-stereo files, since there is already flac_convert_stereo16().
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By mistake, I casted the sample value to uint16_t, which is wrong.
This patch simplifies the code by using a int16_t pointer instead of
casting to int16_t* every time.
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