the most notable bugs are
1. osx_output_set_device_format should use the target asbd rather than AudioFormat. This is because asbd's sample rate calculation reflects the real dop target rate of the DAC, white AudioFormat's sample rate is the original DSD format rate.
2. the original code value the highest rate that's the multiple of the target rate. This cause DOP always have the wrong rate chosen. This is also not necessary for PCM playback --- MPD's goal is bit perfect, and it's meaningless to raise to two or four times the PCM sample rate.
3. if sample_rate cannot be synchronized, the test for falling back to PCM is wrong. If the file format is in DSD format such fallback is necessary, whatever the params.dop setting is.
the code here tried to guard DSD features behind ENABLE_DSD. However, the sample rate setting should be shared between two scenarios.
40a1ebee29 (diff-ce7ecec9ea9ca3df90d9c290cb3ef9d4R795)
The code runs fine if the dac supports the sample rate, as Mac OS will use the device rate if stream rate is 0.
However, when DAC is uncapable of processing the sample rate, a wrong rate (device rate) will be used for the stream rate.
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)
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.
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.
In e068d62 I added code that zeros the remainder of the output buffer
if there are not enough input frames available. I have now learned
that we can signal to the caller of the render callback how much data
is in the output buffers. In practice, the input buffer is so large
that it does not matter so much how we handle input buffer underruns,
but I suppose that saying 'there is no data' is better than 'here is
some silence for you'.
