With mono sound, jack_sample_size is smaller than frame_size (4 vs 2
bytes), and "space/jack_sample_size==0". That means mpd_jack_play()
will return 0, although no error has occurred.
Version 1.0.0 of the libao library added a new field to the
ao_sample_format struct. It is a char * named matrix. When
an ao_sample_format is allocated on the stack, this field contains
garbage. The proper course is to insure that is initialized to NULL.
NULL indicates that we do not want any mapping.
The struct is now initialized using a static initializer, and this
technique is compatible with all known versions of libao.
This fixes the following valgrind warning occuring on the first call of
httpd_output_read_page:
==20124== Conditional jump or move depends on uninitialised value(s)
==20124== at 0x425E65: httpd_output_read_page (httpd_output_plugin.c:240)
==20124== by 0x426087: httpd_output_open (httpd_output_plugin.c:279)
==20124== by 0x41D862: ao_open (output_plugin.h:206)
==20124== by 0x41E133: audio_output_task (output_thread.c:590)
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.
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
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.
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.
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.
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.
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.
Drop the required GLib version from 2.16 to 2.12, because many current
systems still don't have GLib 2.16. This requires several new
compatibility functions in glib_compat.h.
ALSA passes full period buffers to the hardware. If an application
doesn't finish writing a period, libasound will nonetheless send the
partial buffer (with undefined trailing data). This causes noise at
the end of playback. This patch attempts to track the current
position within the period buffer, and generates silence at the end,
before calling snd_pcm_drain().
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.
Same as the previous patch: create up to 16 configured source ports.
The plugin tries to do its best at guessing the right combination for
the given input file, the number of source and destination ports.
Implement the methods enable() and disable(). Bind the HTTP port in
the enable() method, but reject all incoming connections until the
output is opened.
After playback has stopped, the ring buffers may still contain
samples. These will be played when playback is started the next
time. We should clear the buffers each time.
jack_client_new() is deprecated. This requires libjack 0.100
(released nearly 5 years ago). We havn't been testing older libjack
versions anyway.
As a side effect, there is the new option "autostart".
Reintroduce a fix from commit 52a0653 (Warren Dukes): "don't call
snd_pcm_drain unless we're already in the RUNNING state". This prevents
ALSA with dmix from sometimes hanging when snd_pcm_drain is called, e.g.
when moving from one song to the next (as in mantis issue 2634).
drain() is the opposite of cancel(): it waits until all data in the
buffer has finished playing. Instead of implicitly draining in the
close() method like the ALSA plugin has been doing it forever, let the
output thread decide whether to drain or to cancel.
With these methods, an output plugin can allocate some global
resources only if it is actually enabled. The method enable() is
called after daemonization, which allows for more sophisticated
resource allocation during that method.
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".
Accidently, MPD has been using several GLib 2.16 functions for a
while, and nobody noticed yet. To simplify the code base, let's bump
the minimum GLib version for MPD to 2.16. That version is old enough,
and it's reasonable to expect users to have it.
