This updates the copyright header to all be the same, which is
pretty much an update of where to mail request for a copy of the GPL
and the years of the MPD project. This also puts all committers under
'The Music Player Project' umbrella. These entries should go
individually in the AUTHORS file, for consistancy.
If the PCM handle gets disconnected, don't close and clear it in
alsa_recover(). The MPD core will call alsa_close() anyway. This
way, we can always assume that alsa_data.pcm is always valid.
This patch fixes a theoretical (but practically impossible) flaw: the
variable "buffer_time" may be uninitialized when it is used.
Initialize the variable with snd_pcm_hw_params_get_buffer_time().
The default values for buffer_time and period_time were both capped by
the hardware limits on practically all chips. The result was a
period_time which was half as big as the buffer_time. On some chips,
this led to lots of underruns when using a high sample rate (192 kHz),
because MPD had very little time to send new samples to ALSA.
A period time which is one fourth of the buffer time turned out to be
much better. If no period_time is configured, see how much
buffer_time the hardware accepts, and try to configure one fourth of
it as period_time, instead of hard-coding the default period_time
value.
This is yet another attempt to provide a solution which is valid for
all sound chips. Using the SND_PCM_NONBLOCK flag also seemed to solve
the underruns, but put a lot more CPU load to MPD.
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]
Log the real period and buffer size. This might be useful when
debugging xruns. Note that the same information is available in
/proc/asound/card*/pcm*p/sub*/hw_params
Some sound chips/drivers (e.g. Intel HDA) don't support 24 bit
samples, they want to get 32 bit instead. Now that MPD's PCM library
supports 32 bit, add a 32 bit fallback when 24 bit is not supported.
Use GLib's GError library for reporting output device failures.
Note that some init() methods don't clean up properly after a failure,
but that's ok for now, because the MPD core will abort anyway.
Don't call AudioOutputUnitStart() in the play() method, do it after
the device has been opened. We can eliminate the "started" property
now, because the device is always started when it's open.
The MPD core guarantees that the audio_output object is always
consistent, and our pa_simple!=NULL checks are superfluous. Also
don't manually close the device on error in pulse_play(), since the
MPD core does this automatically when the play() method returns 0.
The MPD core guarantees that the audio_output object is always in a
consistent state: either open or closed. When open, it will not call
the open() method again, and when closed, it will not call play().
Removed several checks and the NULL initialization.
audio_output_get_name() has been removed, which was the only function
left in output_api.h. The output plugin doesn't need the audio_output
object at all, remove the parameter from the init() method.
The old API required an output plugin to not return until all data
passed to the play() method is consumed. Some output plugins have to
loop to fulfill that requirement, and may block during that. Simplify
these, by letting them consume only part of the buffer: make play()
return the length of the consumed data.
Now that I've found this nice function in the GLib docs, we can
finally remove our custom sleep function. Still all those callers of
g_usleep() have to be migrated one day to use events, instead of
regular polling.
Pass the music chunk as a "const void *" to the encoder, instead of a
"const char *". Actually, both encoders currently expect 16 bit
samples, passing a 8-bit character is rather pointless.
Always assume the buffer is empty before calling the encoder. Always
flush the buffer immediately after there has been added something.
This reduces the risk of buffer overruns, because there will never be
a "rest" in the current buffer.
The non-blocking mode of libshout is sparsely documented, and MPD's
implementation had several bugs. Also removed connect throttling
code, that is done by the MPD core since 0.14.
The shout_mp3 encoder had two bugs: when no song was ever played, MPD
segfaulted during cleanup. Second bug: memory leak, each time the
shout device was opened, lame_init() was called again, and
lame_close() is only called once during shutdown.
Fix this by shutting down LAME each time the clear_encoder() method is
called.
Move the "while" loop which checks for commands to the caller
ao_pause(). This simplifies the pause() method, and lets us remove
audio_output_is_pending().
If no ports are configured, don't overwrite the (NULL) configuration
with the port names of the first JACK server. If the server changes
after a JACK reconnect, MPD won't attempt to auto-detect again.
Currently, the JACK plugin manipulates the audio_format struct which
was passed to the open() method. This is very likely to break,
because the plugin must not permanently store this pointer. After
this patch, MPD ignores sample rate changes. It looks like other
software is doing the same, and I guess this is a non-issue.
This patch converts the audio_format pointer within jack_data into a
static audio_format struct.
Preparation for supporting other channel numbers than stereo: use
loops instead of duplicating code for the second channel. Most
likely, gcc will unroll these loops, so the binary won't be any
different.
When waiting for free space in the ring buffer, the JACK plugin
sleeped 10ms until there is enough space. This delay was too large
for low-latency setups (<10ms), and created a lot of xruns. Work
around that by reducing the sleep time to 1ms.
A proper solution for this would be to use an event based approach,
and we will do it, just not now.
When the connection failed once, you had to restart MPD, because it
never cleared the jack_data.shutdown flag. Instead, it refused to
play anything "because there is no client thread" (which is wrong at
that point).
On some platforms, g_free() must be used for memory allocated by
GLib. This patch intends to correct a lot of occurrences, but is
probably not complete.
Return the default value in the conf_get_block_*() functions when
param==NULL was passed.
This simplifies a lot of code, because all initialization can be done
in one code path, regardless whether configuration is present.
Two bugs here led to a large number of interrupts being generated on the
sound card when ALSA output is being used. Because we specify no default
period_time, the sound card gives us 3000 interrupts/sec rather than a more
sane 20 or 30. This completes the revert of dd7711 already started by
4ca24f.
The larger bug was in the change to config_get_block_unsigned() and using 0
as the default value for both 'buffer_time' and 'period_time'. This means
any pre-setting of these options in newAlsaData() gets wiped out. Add a new
default for period_time, and ensure default values for buffer_time and
period_time are used if none are provided by the user.
Signed-off-by: Dan McGee <dan@archlinux.org>
[mk: set defaults in newAlsaData() to fix auto-configuration; renamed
"_MS" back to "_US" because ALSA expects microseconds, not milliseconds]
Signed-off-by: Max Kellermann <max@duempel.org>
The null plugin synchronizes the playback so it will happen in real
time. This patch adds a configuration option which disables this: the
playback will then be as fast as possible. This can be useful to
profile MPD.
I was having problems with shoutcast stream outputs before applying
the attached patch, which enlarges the shoutcast output
buffer. Ideally, this should be configurable, but this resolves the
issue for my needs.
This patch tryes to introduce pluggable mixer (struct mixer_plugin) along with some basic infrastructure (mixer_* functions). Instance of mixer (struct mixer) is used in
alsa and oss output plugin
JACK documentation states: "The caller is responsible for calling
free(3) any non-NULL returned value."
This does not seem to include the array elements. Duplicate them
after jack_get_ports(), and free only the array. Convert
JackData.output_ports to non-const.
There have been bug reports on MPD regarding 24 bit output via
libao/esd. The "ao" plugin does not attempt fall back to 16 bit
currently, and thus fails to play 24 bit audio (i.e. all mp3 files).
Make it always use 16 bit samples for now, until more bits are
well-tested.
The OS X output does not seem to support 24 bit audio in the way MPD
implements it currently. Fall back to 16 bit for now, and schedule
24 bit support on OS X for MPD 0.15.
Commit dd7711d8 removed MPD's default ALSA buffer_time. The result
was a buffer size which was way too small for playing streams on some
sound hardware, and caused skips and distorted sound. Revert the
default to 500 ms.
"float (*lamebuf)[2] = g_malloc()" does NOT allocate two float*
buffers. The formula is even wrong: it should be applied to LAME's
output buffer, not its input buffer.
Converted "lamebuf" to the two variables "left" and "right", and
allocate them independently with the exact buffer size. Set
right=left if mono output is configured.
"LOG_H" is a macro which is also used by ffmpeg/log.h. This is
ffmpeg's fault, because short macros should be reserved for
applications, but since it's always a good idea to choose prefixed
macro names, even for applications, we are going to do that in MPD.
