To demonstrate the new I/O thread. libsoup is well-integrated into
the GLib main loop, which made this plugin pretty easy to write.
As a side effect, we have to initialize the I/O thread in all debug
programs that use the input API.
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
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 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.
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.
This replaces the rewinding buffer code from the CURL input plugin.
It is more generic, and allows rewinding even when the server sends
Icy-Metadata (which would have been too difficult to implement within
the CURL plugin).
This is a rather complex patch for the stable branch (v0.15.x), but it
fixes a serious problem: the "vorbis" decoder plugin was unable to
play streams with Icy-Metadata, because it couldn't rewind the stream
after detecting the codec (Vorbis vs. FLAC).
Use the plugin instead of the glue code in normalize.c. This is used
wrapped inside a "autoconv" filter, to enable normalization for all
input file formats.
This plugin is the groundwork for MPD's future generic CUE sheet
support. That's not complete yet, e.g. there is no way for a playlist
plugin to address an arbitrary position within a music file.
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.
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.
When using wave encoder with httpd audio output mpd can input this stream via http and audiofile decoder.
This for example opens simple way to configure lossless audio streaming port(like jack or pulseaudio does but without overhead).
Another possibility can be using it for gathering raw data for visualization plugins (If sync issue will be resolved)
Added the "fd_util" library, which attempts to use the new thread-safe
Linux system calls pipe2(), accept4() and the options O_CLOEXEC,
SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is
not thread safe.
This is particularly important for the "pipe" output plugin (and
others, such as JACK/PulseAudio), because we were heavily leaking file
descriptors to child processes.
Use a single GString buffer object in all functions loading the
database. Enlarge it automatically for long lines. This eliminates
the maximum line length for tag values. There is still an upper limit
of 512 kB to prevent denial of service, but that's reasonable I guess.
Don't let the mixer plugin "override" the libpulse callbacks.
Instead, add a "mixer" attribute to the pulse_output struct, and call
the mixer on all interesting events.
This is a complete rewrite of the PulseAudio output plugin. It uses
the asynchronous API, which gives us more control over everything.
Additionally, it connects to the PulseAudio server on startup, and
keeps this connection up while MPD runs. During pause, instead of
closing the stream, it enables "cork".
This has been replaced by the last.fm playlist plugin. The input
plugin has never worked well, and was just a playground to experiment
with the last.fm radio protocol.
Based on this API, we will add parsers for EXTM3U, PLS, ASX, last.fm
radio and others.
There is no integration into the MPD core yet. Right now, we have a
command line test program. This is work in progress.
This patch implements a light-weight inotify library, and watches all
directories below the music directory. It updates all directories
where files changed after a delay of 5 seconds.
The recorder plugin writes audio played by MPD to a file. This may be
useful for recording radio streams.
This implementation is incomplete, because support for tags is
missing, and MPD should be able to record each track to a different
file.
Instead of hard-coding the path "/etc/mpd.conf", use the configured
$(sysconfdir) path. This can be set with:
./configure --sysconfdir=/etc
Note that this changes the default path to "/usr/local/etc/mpd.conf",
given the default prefix "/usr/local". This is actually more correct
than the old default.
This encoder plugin is a replacement for the LAME encoder plugin for
those who prefer a "free" (non-patent encumbered) encoder library.
Most of the plugin source code is copied from the LAME encoder plugin,
since the LAME and TwoLAME APIs are nearly the same.
Do all the software volume stuff inside each output thread, not in the
player thread. This allows one software mixer per output device, and
also allows the user to configure the mixer type (hardware or
software) for each audio output.
This moves the global "mixer_type" setting into the "audio_output"
section, deprecating the "mixer_enabled" flag.
This mixer plugin may be used instead of the traditional global
software mixer. It integrates with the "volume" filter plugin, and
can control the software volume of an audio output which has no
hardware mixer.
This patch adds initial filter support for audio outputs. Each audio
output gets a "filter" attribute, which is used by ao_play_chunk().
The PCM conversion is now performed by convert_filter_plugin.
audio_output.convert_state has been removed.
The "volume" filter plugin will replace the current software volume
code. One "volume" filter may be attached to each output device.
This will allow the user to use hardware mixers for some devices, and
software mixers for other devices at the same time.
Currently, neither the filter API nor the "volume" plugin is
integrated into MPD.
This little program is used to test mixer plugins in an isolated
environment. This is ALSA-only currently, because we don't have a
real "plugin list" yet, and I'm too lazy to implement a switch.
[mk: folded with patch "Put icy related functions in extra source
files"; moved icy_server.c from HAVE_CURL to ENABLE_HTTPD_OUTPUT;
removed an unused variable]
Let's get rid of the "shout" plugin, and the awfully complicated
icecast daemon setup! MPD can do better if it's doing the HTTP server
stuff on its own. This new plugin has several advantages:
- easier to set up - only one daemon, no password settings, no mount
settings
- MPD controls the encoder and thus already knows the packet
boundaries - icecast has to parse them
- MPD doesn't bother to encode data while nobody is listening
This implementation is very experimental (no header parsing, ignores
request URI, no icy-metadata, ...). It should be able to suport
several encoders in parallel in the future (with different bit rates,
different codec, ...), to make MPD the perfect streaming server. Once
MPD gets multi-player support, we can even mount several different
radio stations on one server.
Even if libsamplerate support is enabled, compile the fallback
resampler. When the user specifies the option
"samplerate_converter=internal", it is chosen in favor of
libsamplerate. This may help users with a weak FPU who don't want to
compile a custom MPD from source, because the fallback resampler does
not use floating point operations.
Added diversion functions to pcm_resample.c. These check which
resampler is enabled at compile time (libsamplerate or fallback).
This prepares the following patch.
Passing libraries through LDFLAGS is a mistake that causes link to fail
when using --as-needed. Since the ld arguments are positional, so are
libtool's. Use the proper variable, thus, to pass the libraries.
This patch introduces the mixer for the pulse output.
Technically speaking, the pulse index is needed to get or set
the volume. You must define callback fonctions to get this index since
the pulse output in mpd is done using the simpe api. The pulse simple api
does not provide the index of the newly defined output.
So callback fonctions are associated to the pulse context.
The list of all the sink input is then retreived.
Then we select the name of the mpd pulse output and control
its volume by its associated index number.
Signed-off-by: Patrice Linel <patnathanael@gmail.com>
Signed-off-by: David Guibert <david.guibert@gmail.com>
[mk: fixed whitespace errors and broke long lines; removed
daemonization changes from main.c]
Turn the music_pipe into a simple music_chunk queue. The music_chunk
allocation code is moved to music_buffer, and is now managed with a
linked list instead of a ring buffer. Two separate music_pipe objects
are used by the decoder for the "current" and the "next" song, which
greatly simplifies the cross-fading code.
Added music_pipe_allocate(), music_pipe_push() and
music_pipe_cancel(). Those functions allow the caller (decoder thread
in this case) to do its own chunk management. The functions
music_pipe_flush() and music_pipe_tag() can now be removed.
The lastfm input plugin enables MPD to play lastfm:// URLs. This
plugin is not complete yet: it plays only the first song in the
last.fm playlist, and the playlist parser isn't even implemented
properly.
