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* Use the DeviceBase for the backendChris Robinson2021-04-2435-69/+69
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* Create a base the ALCdevice and ALCcontext structsChris Robinson2021-04-2422-263/+298
| | | | | A base that contains the API-agnostic data, with ALCdevice and ALCcontext being for AL-specific data.
* Move some more sources to coreChris Robinson2021-04-245-469/+2
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* Use the root mean square for the HRTF B-Format HF scaleChris Robinson2021-04-241-1/+3
| | | | | | | | For the second-order decoder. Since the delays are adjusted to avoid most high frequency phase cancelation, the energy-based scaling with the number of channels used creates a significant HF increase. Using the RMS-based scaling seems to create a more level response, though it's not perfect either. More testing and measurements may be needed.
* Move hrtf.cpp/h to coreChris Robinson2021-04-227-1564/+5
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* Avoid using config methods in hrtf.cppChris Robinson2021-04-225-40/+46
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* Move helpers.cpp to coreChris Robinson2021-04-2217-561/+15
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* Move declarations to a more appropriate headerChris Robinson2021-04-2210-14/+22
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* Print an error when starting the backend failsChris Robinson2021-04-201-0/+4
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* Lower RLIMIT_RTTIME to allow RTKit to give RT priorityChris Robinson2021-04-203-12/+60
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* Fill the PulseAudio buffer before uncorking playbackChris Robinson2021-04-191-14/+14
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* Optionally use RTKit/D-Bus to set elevated priorityChris Robinson2021-04-181-17/+60
| | | | If pthread_setschedparam fails or is unavailable.
* Make an inverted atomic flag type and use itChris Robinson2021-04-152-7/+7
| | | | | | | | The inverted atomic flag replaces test_and_set+clear with test_and_clear+set, essentially inverting the flag status. This makes more logical sense for flagging dirty state, which is less confusing than flagging clean state. The one caveat is ATOMIC_FLAG_INIT (or default construction in C++20) initializes the state to true rather than false.
* Use per-HRIR delay alignment for B-Format-to-HRTFChris Robinson2021-04-091-6/+11
| | | | | | | | | | | | | For HOA signals, the number of responses used with slightly varying delays causes noticeable attenuation in the higher frequencies because of destructive phase interference. This is not a result of minimum phase alignment (attempts to compensate for minimum phase had negligible results), nor does it affect first-order signals (which only has 4 unique responses on each side). This alternate alignment is only used when doing second-order rendering for HRTF output, which is not the default with HRTF. It's likely not very ideal, but it's necessary to prevent second-order rendering with HRTF from sounding muffled.
* Log the reset exception errorChris Robinson2021-04-071-0/+1
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* Advertise the in-progress AL_SOFT_UHJ extensionChris Robinson2021-04-011-1/+12
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* Rename Uhj2Encoder to UhjEncoderChris Robinson2021-04-014-7/+7
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* Avoid passing an array of pointersChris Robinson2021-04-011-7/+1
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* Add support for 4-channel UHJChris Robinson2021-03-316-14/+27
| | | | Also add the SOFT moniker to the new macros
* Handle 3-channel UHJ audio buffersChris Robinson2021-03-316-7/+15
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* Decode UHJ buffers to B-Format for mixingChris Robinson2021-03-313-69/+103
| | | | | This should also have an adjustment for the shelf filter. Although it's not clear what the appropriate adjustments should be.
* Start an interface for providing UHJ audioChris Robinson2021-03-315-1/+36
| | | | Currently only 2-channel UHJ, which gets treated as stereo.
* Combine some duplicate codeChris Robinson2021-03-303-48/+52
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* Move the UHJ phase shifter to a common headerChris Robinson2021-03-281-1/+1
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* Don't add the resampler prepadding to the source size to loadChris Robinson2021-03-231-18/+17
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* Load/convert samples from all channels at once for mixingChris Robinson2021-03-194-116/+139
| | | | | | | This uses a bit more memory (each voice needs to hold buffers for the deinterleaved samples of each channel, instead of just one buffer for the current channel being mixed on the device), but it will allow for handling formats that need or prefer their channels decoded together.
* Don't force application name with pulseaudioIlya Fedin2021-03-181-8/+1
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* Don't activate the WASAPI device when initially opening itChris Robinson2021-03-171-17/+2
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* Avoid returning objects with a reference parameterChris Robinson2021-03-161-13/+13
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* Check that CoCreateInstance succeedsChris Robinson2021-03-161-13/+17
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* Workaround mingw complaining about the %z formatterChris Robinson2021-03-123-2/+16
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* Avoid shadowing variable namesChris Robinson2021-03-111-6/+6
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* Declare the attribute counts closer to where they areChris Robinson2021-03-101-25/+30
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* Prevent querying the capture device name from a playback deviceChris Robinson2021-03-091-3/+16
| | | | And vice-versa.
* Don't verify and hold a device reference in alcRenderSamplesSOFTChris Robinson2021-03-091-5/+4
| | | | | | | | | | | | NULL devices are still checked, but invalid non-NULL device handles will invoke undefined behavior, as will attempting to close the device while the function is being executed (modifying the device state while the function is being called was inadvertently already UB, and will now remain so). This change is solely so alcRenderSamplesSOFT can be used in a buffer callback, and other places that need functions to be real-time safe. The verification requires locking to access the device list, which isn't allowed in a real-time callback.
* Add a function/extension to reopen a playback deviceChris Robinson2021-03-092-25/+112
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* Initialize the new audio unit before disposing the old oneChris Robinson2021-03-091-6/+6
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* Move the ComPtr wrapper to a common headerChris Robinson2021-03-082-126/+2
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* Allow calling BackendBase::open multiple times on playback devicesChris Robinson2021-03-0815-257/+360
| | | | | | | | | | | It will not be called while the device is running. If the first call succeeds, a subsequent call that happens to fail must leave the existing device state as it was so it can be resumed. This is a rough first pass. It will fail when trying to re-open the same device which can only be opened once (for instance, with direct hardware access, on hardware that doesn't do its own mixing). Some backends won't guarantee the new device is usable until the reset() or start() call.
* Avoid making BSincPointsMax publicChris Robinson2021-03-011-1/+0
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* Avoid trying to get the app path when it fails on Windows tooChris Robinson2021-02-271-9/+10
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* Avoid calling readlink on platforms that don't support itChris Robinson2021-02-271-11/+11
| | | | Also don't keep trying to find the path+name if it fails the first time.
* Avoiding cutting all bsinc resampler output at scale 0Chris Robinson2021-02-241-2/+2
| | | | | | | | | | | | This is mostly for the SampleConverter, used by some capture backends. When recording at really low rates, like 5512hz, with a device capturing at a higher rate like 44100hz or 48000hz, it hits the filter's downscaling limit and produces pure silence. In such cases, it's better to just accept some aliasing noise so that the app will still get some recognizable audio. The alternative would be to scale the desired rate by 2x, 3x, etc until it's above the bsinc limit, then take every 2nd, 3rd, etc sample of the result as if by an extra simpler resampler pass.
* Clean up some formattingChris Robinson2021-02-161-81/+71
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* Add enumeration to the JACK backendChris Robinson2021-02-131-35/+84
| | | | | | | | Port names seem to be structured as <device_name:channel_name> or <app_name:channel_name>. I'm not sure if this is always the case, but it seems some other apps expect something like this. Also fix the port selection to exclude MIDI ports and allow non-physical ports.
* Use spans instead of references to arraysChris Robinson2021-02-063-15/+16
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* Stop the Oboe recording stream when recording is stoppedChris Robinson2021-02-021-2/+2
| | | | Hopefully Oboe will retain any unread samples and let them be read.
* Implement Oboe captureChris Robinson2021-01-311-4/+135
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* Set the oboe stream buffer sizeChris Robinson2021-01-311-0/+2
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* Avoid copying buffers for aligning overlapped windowsChris Robinson2021-01-302-33/+42
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