During the whole output thread, the audio_output object is locked, and
it is only unlocked while waiting for the GCond and while running a
plugin method. The error handler in ao_play_chunk() attempted to lock
the object again, which was code from MPD 0.15.x which should have
been removed a long time ago.
Until the decoder plugin has called decoder_initialized(), the player
may not submit seek commands. This however could occur with a slow
decoder and a CUE file with a virtual song offset. This patch adds
another check.
When you don't explicitly set an output sample rate, liblame tries to
guess an output sample rate from the input sample rate. You would
think that this "guessing" consists of just setting both equal, but
that is not the case. For 44.1kHz at 96kbit/s, liblame chooses
32kHz. This patch explicitly configures the output sample rate, to
stop the bad guessing.
This is a MPD 0.16 regression: when playing a 24 bit file, the switch
to 16 bit was made only partially, after mBytesPerPacket and
mBytesPerFrame had already been applied.
That means mBytesPerFrame referred to 24 bit, and mBitsPerChannel
referred to 16 bits. Of course, that cannot work.
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
Should be safe on OS X 10.4 (32-bit), since Apple's OSStatus boils
down to "signed long", and g_set_error() takes gint, which is really
just "int". Assigning "signed long" to "int" on 32-bit Unix should be
just fine, since both are signed 32-bit ints.
No idea if this is safe on 64-bit OS X.
Add new config parameter 'device' to audio_output type "osx":
- if not supplied or set to "default", open default device
- if set to "system", open system device
- otherwise 'device' should be an audio device name: mpd will find and
open the specified audio device, falling back to the default
device if it's not found
this is inconsistent with other commands (e.g. find) and seems wrong --
a song with no stickers attached is a perfectly valid state and an empty
list of stickers is also perfectly valid.
Fixes a regression: for output_plugin.delay(), we added a method to
the timer class which returns the delay in milliseconds. This fails
to detect negative values, because the unsigned integer is divided by
1000, and then casted to signed.
After popular demand, I've switched the order of "artist" and "title"
in the stream title. There is no standard, and there is no reliable
way to parse those from the stream title.
Call access() and print an extra error message when EACCES is
returned. Hopefully this will reduce the number of support requests
due to wrong file permissions.
When one song is played twice, and the decoder is working on the
second "instance", but the first should be seeked, the check in
player_seek_decoder() may assume that it can reuse the decoder without
exchanging pipes. The last thing was the mistake: the pipe pointer
was different, which led to an assertion failure. This patch adds
another check which exchanges the player pipe.
Change the assertion on "fail_timer==NULL" in OPEN to a runtime check.
This assertion crashed when the output thread failed while the player
thread was calling audio_output_open().
When you pass the flag AI_ADDRCONFIG to getaddrinfo(), it does not
consider address families on the loopback device. When run on a
machine without an external network card, just with "lo", it was
unable to look up any address.
Using libffado, to play on firewire audio devices.
Warning: this plugin was not tested successfully. I just couldn't
keep libffado2 from crashing. Use at your own risk.
For details, see my Debian bug reports:
http://bugs.debian.org/601657http://bugs.debian.org/601659
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.
Added support for a new optional configuration setting for the httpd output
named "bind_to_address". Setting it to a specific IP address (v4 or v6) will
cause the httpd output to bind to that address exclusively. Supporting
multiple addresses in parallel is future work.
This implements the feature requests #2998 and #2646.
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.
I've added PIPE_EVENT_SHUTDOWN because calling g_main_loop_quit() do not work when called from another thread.
Main thread was sleeping in g_poll() so I needed some way to wake it up.
By some strange reason call close(event_pipe[0]) in event_pipe_deinit() hangs.
In current implementation that code never reached so that was not a problem :-)
I've added a conditional to leave event_pipe[0] open on Win32.
The ReplayGain filter clamped the gain to max. 100 % even if the
algorithm determined the signal needed a boost. That would result in any
such tracks being played with too low volume, effectively defeating the
purpose of the filter.
Clear the notification before finishing the CANCEL command, so the
notify_wait() after that will always wait for the right notification,
sent by audio_output_all_cancel().
Unfortunately, there's no "optimized" implementation here. We can't
use Linux's proprietary system call dup3(), because it would require
us to specify the new descriptor.
If a song with an absolute path points inside the music directory,
print only the relative part. This happens when partial songs from a
playlist file were loaded.
I've already changed the "playlistinfo" command to hide HTTP
passwords, but forgot to do the same for the simpler "playlist"
command. This patch changes queue_print_uris() to use the code from
song_print_uri().
MPD doesn't have child processes anymore, and thus we're not expecting
to receive SIGCHLD very often. Since hard disk access isn't
interrupted by signals anyway, we don't need those excessive checks.
The function playlist_metadata_load() will overwrite the input buffer
before using the "name" parameter; since "name" points to the same
buffer, we'll get a corrupted string.
Some users reported that MPD crashes when using a new CURL version
with the threaded DNS resolver enabled. It seems that
curl_multi_fdset() returns no file descriptor when the DNS resolver
runs in another thread, so MPD does not have any event to wait for.
On the CURL mailing list, somebody suggested to sleep for a fixed
amount of time. This is not an elegant solution, because daemons
should never have to sleep without waiting for an event. I hope the
CURL developers will review the API and remove the threaded DNS
resolver.
Meanwhile, I'm removing the assertion in question, to allow those
unfortunate users running the latest CURL version to continue using
MPD.
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.
I've attached a patch that will make file URIs work on operating systems
that provide the getpeereid() function call to check the user ID of the
peer connected to a UNIX domain socket.
I took this tag name from a MusePack sample file I got from a user.
It is not documented in the APE specification:
http://wiki.hydrogenaudio.org/index.php?title=APE_key
People seem to be using undocumented extensions to the specification
anyway, and the best we can do is attempt to support them.
The new function playlist_open_any() combines playlist_mapper_open(),
playlist_list_open_uri() and playlist_list_open_stream(), providing an
easy API for all of them.
Merged both loops into playlist_list_open_stream(). This is needed
because playlist_list_open_stream() needs to know the MIME type, which
is only known after the stream has become "ready".
This buggy implementation failed to allow "..." sequences, because the
dot count was always zero. The usefulness of allowing "..." (or more
dots) is debatable, but since it's a valid file name, we allow it.
libcue's track_get_length() returns 0 for the last track, because that
information is not available in the CUE sheet. This makes MPD quit
playing the last track immediately. If we set "song.end_ms=0", MPD
will play the track until the end of the song file, which is what we
want.
I've attached a patch that will make file URIs work on operating systems
that provide the getpeereid() function call to check the user ID of the
peer connected to a UNIX domain socket.
this greatly improves performance of commands that return a lot
of data, e.g. search results or recursive content of a directory,
while being connected to local mpd via tcp/ip socket.
Memory leak fix. The input_stream object passed to
playlist_list_open_stream_suffix() must be closed by the caller - this
however never happens in playlist_list_open_path(), because it does
not return it to the caller.
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.
This is probably unsafe, and doesn't protect against symlink loops,
but we will eventually add this when we bring update*.c and inotify*.c
closer together.
This shouldn't really happen, but insane users might delete/rename the
music directory while MPD runs. What was even more insane was that
MPD crashed due to this. This is a workaround - there is currently
nothing useful we can do in this case; except maybe poll for the music
directory to reappear, but that's too much trouble for a user error.
I took these tag names from a MusePack sample file I got from a user.
These are not documented in the APE specification:
http://wiki.hydrogenaudio.org/index.php?title=APE_key
People seem to be using undocumented extensions to the specification
anyway, and the best we can do is attempt to support them.
Reduce the overhead. Most buffers used by MPD are around 2 to 4 kB.
8 kB seems large enough to keep heap fragmentation low.
Additionally, this patch fixes an off-by-one error in the alignment
formula.
On mingw32, snprintf() expects a 64 bit integer instead of a "long
int" for "%li" - this is not consistent with our expectation, so we're
using plain sprintf().
For some unknown reason, read() blocks on WIN32, even though it was
invoked inside the G_IO_IN callback. By switching to GIOChannel
functions, this problem is solved, and it works on both Linux and
Windows.
On WIN32, use g_io_channel_win32_new_fd() instead of
g_io_channel_unix_new(). There doesn't seem to be a practical
difference, but it seems more correct.
In mingw32, int16_t is not defined by sys/types.h, but it is by stdint.h,
and it is in the int16_t man page as being defined in stdint.h. Thanks to
mithi for help debugging.
Don't add it to the filter chain, because we need to apply replay gain
before cross-fading with the next song. Add a second replay_gain
filter which is used for the song being faded in (chunk->other).
This is useful at the maximum depth level, to update newly created
directories. It is however questionable if the hard-coded 5 seconds
delay is enough to create new directory trees with all of their files,
but we might make that delay configurable in the future.
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.
.. rather then append to the end of the previous one
Cuebreakpoints from the cuetools package has three modes of operation,
and the default is to append pregap (INDEX 00) to the end of the
previous track. This is the behavior most compliant to the existing
cue files.
Here is the patch which fixes the issue. I borrowed bits of
implementation from cuebreakpoints. I assumed that the whole audio
file must be covered by head-to-head going tracks, which is how
hardware CD players probably work. In cue_tag I changed rounding from
rounding up to rounding down because the thing in mpd which calculates
actual track duration (and current position) rounds it down, and I
didn't want to see in my playlist values different from whose in a
now-playing progress bar.
I've compared the resultant mpd behaviour with "mplayer -ss MM:SS.MS"
where the time was supplied by cuebreakpoints and noticed that mplayer
started each track a bit earlier then mpd, though this was the same
before the patch.
"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'"
Previously, tags of the new song being cross-faded in were sent
immediately. That can cause wrong information being displayed,
because the "previous" song might send its tag at the end again,
overriding the "next" song's tag. This patch saves & merges the tag
of the next song, and sends it when cross-fading is finished, and the
next song really starts.
When decoder->timestamp is calculated, the PCM data is already
converted to out_audio_format; using in_audio_format may cause funny
speedups/slowdowns.
"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"
When handle_update() was modified to use uri_safe_local(), suddently
"mpc update ''" and "mpc update '/'" stopped working, because both are
not a "safe" local URI. This patch adds a special case for these, to
retain backwards compatibility.
Did you ever accidently click "stop" while feeding a radio station?
This option sets the output device to "pause" to disable the "close"
method. It falls back to "pause" then, which is specific to the
plugin. Some plugins implement it by feeding silence.
With single+repeat enabled, it is expected that MPD repeats the
current song over andd over. With random mode also enabled, this
didn't work, because the song order was shuffled internally. This
patch adds a special check for this case.
This is a very basic check, which only ensures that the path does not
begin with a slash, doesn't have double slashes and the special names
"." and ".." are forbidden.
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().
Add an option for each audio output which enables the use of the
hardware mixer, instead of the software volume code.
This is hardware specific, and assumes linear volume control. This is
not the case for hardware mixers which were tested, making this patch
somewhat useless, but we will use it to experiment with the settings,
to find a good solution.
Apply the replay gain in the output thread. This means a new setting
will be active instantly, without going through the whole music pipe.
And we might have different replay gain settings for each audio output
device.
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.
The previous patch not only moved code, it also changed the check.
Negative gain values seem to be valid after all, there just was the
"magic" value 0.0 which means "not available". This patch changes the
"magic" value to "INFINITY", and uses the C99 function isinf() to
check. It might have been a better idea to use "NAN", but the "NAN"
macro is a GNU extension.
When all plugins have failed, MPD used to fall back to the "mad"
decoder plugin, to handle those radio streams without a Content-Type
response header. This however leads to unexpected results (garbage
being played) when the stream isn't really mp3. Since we care little
about "bad" streams, we shouldn't have hacks which have bad side
effects.
Let's get rid of this hack now! Only try to "mad" plugin if there was
no match at all (Content-Type, path suffix) and no other plugin has
been tried.
When enabling the pulse device fails, clear po->mainloop after
pa_threaded_mainloop_free() has finished. This is important for the
assertions.
Two wrong g_free() calls were also removed.
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.
The patch "input/file: don't fall back to parent directory" introduced
a regression: when trying to play a CUE track, decoder_run_song()
tries to open the file as a stream first, but this fails, because the
path is virtual.
This patch fixes decoder_run_song() (instead of reverting the previous
patch) to accept input_stream_open() failures if the song is a local
file. It passes the responsibility to handle non-existing files to
the decoder's file_decode() method.
Max Kellermann