Rename the "version" struct, because it seems to be a reserved name on
Solaris:
"src/decoder/mad_decoder_plugin.c", line 550: (enum) tag redeclared: version
cc: acomp failed for src/decoder/mad_decoder_plugin.c
Replaced all occurrences of g_error() with MPD_ERROR() located in a new header
file 'mpd_error.h'. This macro uses g_critical() to print the error message
and then exits gracefully in contrast to g_error() which would internally call
abort() to produce a core dump.
The macro name is distinctive and allows to find all places with dubious error
handling. The long-term goal is to get rid of MPD_ERROR() altogether. To
facilitate the eventual removal of this macro it was added in a new header
file rather than to an existing header file.
This fixes#2995 and #3007.
According to the mantis bug report 2847, there are several possible
variations of the "album artist" tag:
- "album artist"
- "album_artist"
- "albumartist"
This patch adds support for the latter two.
In libwildmidi 0.2.3, the function WildMidi_SampledSeek() was removed,
without changing the SO name. This patch adds an autoconf check for
that function. Fall back to WildMidi_FastSeek() if
WildMidi_SampledSeek() is not available anymore.
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.
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.
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.
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.
Pass sizeof(buf) to decoder_data(), not the number of samples (which
is half the size). At the same time, pass GME_BUF_SIZE to gme_play()
- libgme really wants to get the number of samples, not the number of
stereo frames. Previously, this plugin had been using only the first
half of the buffer.
Without libid3tag, we were trying to skip the ID3 frame (since
0.15.2). Its length however was not calculated at all, we were just
dropping everything from the current input buffer. This lead to the
first few seconds of the file being skipped. This patch attempts to
calculate the ID3v2 frame size with the formula from:
http://www.id3.org/id3v2.4.0-structure 3.1 and 6.2
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.
"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'"
"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"
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().
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.
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.
