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What's happening is the `ptr' argument to that function is NULL for me
every time. `ptr' is unconditionally dereferenced to generate a log
message, and this is where mpd crashes.
Attached is a simple patch that tests for NULL and omits the log. With
this patch the crash disappeared and mpd went back to working well.
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Supports a number of videogame music formats, more info here:
http://www.fly.net/~ant/libs/audio.html
I wrote this plugin for the latest svn, get it here:
http://code.google.com/p/game-music-emu/source/checkout
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Adds mixrampdb and mixrampdelay commands. Reads MIXRAP_START and
MIXRAMP_END tags from FLAC files and overlaps instead of crossfading.
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"When playing musepack files with mpd v0.15.8, rg seems to have no effect.
Using sample file below, mpd says 'computing ReplayGain album scale with gain 122.879997, peak 0.549150'.
One thing though, if I build mpd against old libmpcdec-1.2.6, rg works
as expected: 'computing ReplayGain album scale with gain 16.820000,
peak 0.099765'"
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"There is a bug in fixed-point musepack (musepack_src_r435) playback.
In floating-point audio is OK but in fixed audio is distorted. I have
made a patch for this"
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Pass everything to the GLib logging library. No direct stderr access.
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The pointer is invalid if av_open_input_file() fails.
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Removed the decoder_command_finished() call at the end of
mp3_decode(). This is invalid, because decoder_command_finished() has
already been called in mp3_read().
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Don't allocate each replay_gain_info object on the heap. Those
objects who held a pointer now store a full replay_gain_info object.
This reduces the number of allocations and heap fragmentation.
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Pass the current URI to wavpack_open_wvc().
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Don't use the function ffmpeg_helper(), don't initialize the codec.
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Fix a memory leak in some code paths.
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Use input_stream.uri.
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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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Easier to reuse the function.
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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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Make this code is reusable.
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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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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).
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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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This is like tag_dup(), but works with an input_stream object instead
of a file path.
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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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First multiply the floating point return value of
decoder_seek_where(), then cast to integer.
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Removed local variable "sample_rate".
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Use the signed C99 type int8_t instead.
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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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This fixes a regression due to a typo caused by "decoder: use
audio_format_init_checked()".
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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.
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The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
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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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Pass the audiofile_setup_sample_format() result to
audio_format_init_checked().
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