From: Christian Kröner <ckroener@gmx.net>
This just copies the necessary bits and pieces from the ALSA plugin and applies them to OSXOutput based on dop config setting. It only changes the OSXOutput plugin as needed for DoP (further changes to support additionally e.g. integer mode or setting the physical device mode require rather a complete rewrite of the output plugin).
Fortunately the Core Audio API is by default bit perfect and supports DoP with minimal changes (setting the sampling rate accordingly after ensuring that the physical mode supports at least 24 bits per channel seems to be enough). This was tested on an Amanero Combo384 device hooked up to a ES9018 DAC.
USAGE (try only on DACs that support DoP):
- Add dop "yes" option to mpdconf
- Be sure to set at least 24bits per channel before playing some DSD file (using Audio-MIDI-Setup)
- Based on the dop setting, MPD will change the sample rate as required and output DoP signal to the DAC
- Hog mode is recommended to ensure that no other program will try to mix some output with the DoP stream (resulting in bad noise)
- Alternatively set the default output device to another device (e.g. the built-in output) to avoid having other audio interfere with DSD playback
The output plugin shall decide whether to insert silence or do nothing
at all. The ALSA output plugin has already implemented this.
Inserting silence is not necessary or helpful for some plugins, and
may even hurt them (e.g. "recorder").
These attributes are printed in the "outputs" response, and the new
command "outputset" allows setting new values.
No attributes are currently implemented.
[mk: the following text was copied from
https://github.com/MusicPlayerDaemon/MPD/pull/167]
For certain format (hi-res files) and normal buffer size hardware, The
hardware may at once consume most of the buffers. However, in Delay()
function, MPD is supposed to wait for 25 ms after the next try. it
will create a hiccup. The negative impact is much major than
increasing the latency.
I understand larger buffers come at a price. That's why in my earlier
commit last year I significantly reduced it. However, the buffer size
in CoreAudio is set according to the hardware, which is super small
latency. For instance, the system audio of 2015 generation of macbook
pro has maximum buffer size of 4096 samples, which is just 0.09s for
44.1k framerate, or 0.04s for 96k frames --- . compare to the 0.5 sec
latency alsa plugin has, even if we quadruple it, it's still super
tiny.
This fixes spurious replay gain logs when the player inserts silence
chunks, because those silence chunks had no replay gain attached,
resetting the ReplayGainFilter state, flipping it forth and back.
After UnlockActivate() returns, we not only need to check for errors,
but also for more room in the ring buffer. If we don't check the ring
buffer, it may be drained already, and the cond.wait() call will never
finish.
Closes#151
Without the flush, ReadPage() may not return any data, or not all
data. This may result in incomplete ddata the new "header" page,
corrupting streams with some encoders such as Vorbis.
Fixes#145
Allows defining a list of supported audio formats, and allows
switching on and off DoP with certain formats.
This is a first rough draft. The setting syntax and its semantics may
still be redesigned.
There is no documentation on whether calling shout_metadata_add()
multiple times on one instance is allowed. To be sure, let's allocate
the object on demand each time in SendTag().
Passing it by value is actually smaller (32 bit) than the rvalue
reference (64 bit pointer), and it ensures that the object is consumed
after the call returns, no matter how the methods are implemented.
This loop was introduced in commit
24c1f46353, but -EPIPE is not a possible
error condition for snd_pcm_hw_params(). This code does not appear to
make sense. Problems with a wrong period_time should be caught before
that by snd_pcm_hw_params_set_period_time_near().
This commit removes the last "goto" in MPD! Yay!
RoarAudio's sndio emulation has been a source for annoyances. First,
their headers turned out to be broken with C++, due to their use of
the "new" keyword. Then they used a preprocessor macro to rename
"sio_hdl" to something else, effectively disallowing the use of
forward declarations. Enough is enough, and I'm removing support for
it.
RoarAudio users should better use the RoarAudio output plugin.
