some device seems to have issue with setting kAudioDevicePropertyVolumeScalar with kAudioObjectPropertyElementMaster. Use AudioToolbox 's kAudioHardwareServiceDeviceProperty_VirtualMasterVolume instead.
Ideally, we should get the steoro channels first, and set the kAudioDevicePropertyVolumeScalar for each channel, which is doable as presented in https://github.com/cmus/cmus/blob/master/op/coreaudio.c. I will do a follow up PR after refactor PR.
This PR will fix#271.
special thanks to @coroner21 who contributed a nice way to score hardware supported format in #292
Also, The DSD related code are all guarded with ENABLE_DSD flag.
- Update the mixer to set on device property instead of audio unit property. When user choose "hardware" as mixer type, they will be able to change the hardware device volume instead of the software (AudioUnit) volume.
- We don't use square root scale in volume calculation as previous code did. This will make the volume level in line with system volume meter --- That is, MPD will have the same percentage volume reading compared to System Setting (Either in "System Preference" or in "Audio Midi Setup" app)
This code was added in 21851c0673 but
looks completely broken:
- the status code is "206 OK" but "206" would be "Partial Content"
- the "Content-Length" header has a bogus value
- the "Content-RangeX" parameter has different bogus values (why
"Content-RangeX" anyway and not "Content-Range"?)
Apart from that, there are strange undocumented non-standard headers
which are probably there to work around bugs/expectations in one
broken proprietary client product. But these days, MPD doesn't bend
over to support broken clients. So let's kill this code.
Closes#304
Don't reactivate the PCM device immediately after Cancel() is
finished; if Cancel() gets called this may mean that new data may take
a while to produce, or no data at all will be produced because the
current song is being stopped.
Once new data is available, Play() will automatically reactivate the
PCM.
This fixes underruns when switching songs manually (closes#264).
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
[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.
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.
Coverity discovered that the Pulse plugin could throw exceptions from
Pause(), but that method was marked "noexcept" because its caller was
not designed to catch exceptions. So instead of avoiding exceptions
(by catching and logging them in each and every implementation), let's
allow them, and do the catch/log game in the MPD core.
Fixes build failure on OS X, closes#44. With the other plugins,
that's not critical, because those use the AudioOutputWrapper, which
hides this problem.
The "pure" and "const" attributes are not so well-defined, and a
recent clang version implements an optimization which pushes the
definition's boundary beyond what I believed it was. clang now
assumes that functions declared "pure" cannot throw exceptions, even
if they lack the "noexcept" specification.
When compiled with this new clang version, MPD will crash randomly if
an exception happens to get thrown by such as "pure" function
(https://github.com/MusicPlayerDaemon/MPD/issues/41).
This commit removes all such misplaced "pure" and "const" attributes,
closing #41.
Fixes another buffer overflow: if the stream has a very long title or
URL, resulting in a metadata string of more than 2 kB, icy_string[0]
is a negative value, which gets casted to size_t - ouch!
https://bugs.musicpd.org/view.php?id=4652
Fixes a buffer overflow due to the bad formula rounding the buffer
size up. At the same time, remove the "+1" from the meta_length
calculation, which takes the padding into account and at the same time
implements proper rounding.
