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.
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.
Major API redesign: don't let the caller allocate the input_stream
object. Let each input plugin allocate its own (derived/extended)
input_stream pointer. The "data" attribute can now be removed, and
all input plugins simply cast the input_stream pointer to their own
structure (with an "struct input_stream base" as the first attribute).
This patch changes the following decoder plugins to implement
stream_tag() instead of tag_dup():
faad, ffmpeg, mad, modplug, mp4ff, mpcdec, oggflac
This simplifies their code, because they do not need to take care of
opening/closing the stream.
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.
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.
The plugin code tried to force libavcodec to supply stereo samples.
That however has never actually worked. By removing this code, we are
able to play surround files for the first time.
Removed the "vtrack" local variable (which triggered a gcc warning
because it was after the newly introduced NULL check), and run
strtol() on the original parameter.
On some platforms, libavcodec wants the output buffer aligned to 16
bytes (because it uses SSE/Altivec internally). It will segfault when
you don't obey this rule.
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.
MPD has been supporting 32 bit samples since version 0.15. This patch
changes one check, and removes the 32->24 conversion code.
Note that WavPack floating point samples have 32 bits, and MPD doesn't
have a special check for floating point - therefore, this WavPack
plugin still returns 24 bit integer samples as before (until we have
float support in the MPD core).
Call decoder_initialize() before entering the loop. We don't need to
call ov_read() before ov_info(). When the stream number changes,
check if the audio format is still the same.
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.
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.
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.
The decoder loop of flac_decode_internal(), flac_container_decode()
and flac_filedecode_internal() is merged into this one function. This
unifies the code, and uses the frame number to identify the end of a
CUE sub song.
If flac_container_decode() gets a seek destination which is out of
range, it ignores the SEEK command (never finishes it). This leads to
MPD lockup, because the player thread waits for completion.
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.
Convert the metadata with the libavformat function av_metadata_conv().
This ensures that canonical tag names are provided by libavformat, and
we can remove the "artist" vs "author" workaround.
svn r13289 of libvorbis introduced static callbacks (like OV_CALLBACKS_DEFAULT)
defined in "vorbisfile.h" header. First released version with this change is libvorbis-1.2.2.
In libversion-1.2.3 OV_EXCLUDE_STATIC_CALLBACKS define was added to avoid
warnings about unused static callbacks. Information on the OV_EXCLUDE_STATIC_CALLBACKS
can be found in http://svn.xiph.org/trunk/vorbis/CHANGES.
Don't initialize "vc" and "cs" with FLAC__metadata_object_new(); that
value is overwritten by FLAC__metadata_get_tags() and
FLAC__metadata_get_cuesheet().
The "off_t" type may change when you enable or disable large file
support on 32 bit platforms. This caused severe ABI problems within
MPD when we enabled LFS for the first time: two sources included
config.h and sys/types.h in different order, and had different off_t
sizes - leading to memory corruption because of ABI incompatibility.
This patch attempts to get rid of all public "off_t" uses: it removes
"off_t" from the input_stream ABI/API, and switches to GLib's 64 bit
"goffset" type. This may hurt 32 bit embedded platforms a tiny bit,
but that's not even measurable.
Usually, we read our "artist" tag from ffmpeg's "author" tag. In some
cases however (e.g. APE), this tag is named "artist". This patch
implements a fallback: if no "author" is found, MPD tries to use
"artist".
When the ID3 tag in an AAC file is larger than the current buffer, the
function decoder_buffer_consume() aborts. By using the new function
decoder_buffer_skip() instead, we can safely skip the ID3 tag.
