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Same as ".m4a".
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libavformat 0.6 does not pass the original URI pointer to the "open"
method, which leads to a crash because MPD was using a dirty hack to
pass a pointer to that method.
This patch switches to av_open_input_stream() with a custom
ByteIOContext class, instead of doing the URI string hack with
av_open_input_file().
Loosely based on a patch from Jasper St. Pierre.
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Use the libavformat function av_probe_input_format() to probe the
AVInputFormat, instead of letting av_open_input_file() do it
implicitly. We will switch to av_open_input_stream() very soon, which
does not have the probing code.
Loosely based on a patch from Jasper St. Pierre.
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Free the string allocated by decoder_get_uri().
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Fix a memory leak in some code paths.
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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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Make it X_decoder_plugin.c.
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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 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.
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Support deprecated MIME types such as "audio/x-ogg". Support new
types such as "audio/flac".
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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.
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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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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.
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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.
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Let the audio_check library verify the audio format in all (relevant,
i.e. non-hardcoded) plugins.
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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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libavformat gives us the song artist in the "author" field. Since we
use av_metadata_conv(), we don't need to check for "artist".
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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.
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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".
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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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Call av_metadata_get() in a loop.
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This warning is useless. I assume the author added it for debugging
purposes.
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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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implemetation
function was implemented in the version we are comparing to so there must be higher or equal
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After much research[1][2][3] this should be the majority of currently
supported file extensions and mime-types for the currently supported
ffmpeg formats. This list maybe incomplete, but it's more complete
than anything else out there that I've been able to find. This list
needs to be updated every now and again as the ffmpeg sources support
more formats.
1. Sources
2. wiki.multimedia.cx
3. filext.com
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Preparing for per-plugin configuration sections in mpd.conf.
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The ffmpeg library supports the "True Audio Codec". The entry in
ffmpeg_suffixes was missing.
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The "current" variable is used for calculating the seek destination,
and was declared as "int". With very long song files, the 32 bit
integer can overflow. ffmpeg expects an int64_t, which is very
unlikely to overflow. Switch to int64_t.
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When ffmpeg cannot estimate the elapsed time, it sets
AVPacket.pts=AV_NOPTS_VALUE. Our ffmpeg decoder plugin did not check
for that special value.
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If avcodec_decode_audio2() returns no output for an AVPacket,
libavcodec may buffer some data, and return a larger chunk of output
later. This patch disables a lot of bogus warnings.
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Output the name of the codec as a debug message. During my tests,
ffmpeg never filled this struct member, but it may do so in the past,
and this debug message might become helpful.
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ffmpeg_tag_internal() does not look for a few tags that mpd
supports. Most noteably:
comment -> TAG_ITEM_COMMENT -> Description
genre -> TAG_ITEM_GENRE -> WM/Genre (not WM/GenreID)
year -> TAG_ITEM_DATE -> WM/Year
I *think* that this is the last of the tags that AVFormatContext() in
ffmpeg supports that mpd also uses.
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Use GLib's G_GNUC_UNUSED instead of gcc.h's mpd_unused.
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The old code casted it to a 32 bit integer, which cut off bits.
AVFormatContext.duration is a int64_t, so use this type.
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Use NULL instead. Found by sparse.
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Refuse to play audio formats which are not supported by MPD.
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