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* Better sort the main delay line tapsChris Robinson2016-10-061-60/+39
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* Decorrelate the early reflection inputsChris Robinson2016-10-061-10/+19
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* Add some more 'restrict' keywordsChris Robinson2016-10-067-43/+57
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* Pass current and target gains directly for mixingChris Robinson2016-10-056-155/+64
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* Update a comment about using row mixersChris Robinson2016-10-051-3/+1
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* Better pack the late reverb low- and all-pass variablesChris Robinson2016-10-051-21/+25
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* Use the row mixer functions for the B-to-A-Format conversionChris Robinson2016-10-041-16/+10
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* Make some pointer-to-array parameters constChris Robinson2016-10-048-45/+53
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* Include wtypes.h for defining Windows' property keysChris Robinson2016-10-042-0/+2
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* Enhance reverb using B-Format processingChris Robinson2016-10-031-405/+386
| | | | | | | | | | | | Technically it uses A-Format processing from the B-Format input and output. But this attempts to provide better spatial definition to the reverberation so that it can be used in a more generic fashion, allowing it to be decoded as any other B-Format signal to whatever output is needed, and also allowing for a bit of height information when the output is capable of such. There may still be some kinks to work out, such as properly decorrelating the early reflection taps and tweaking the late reverb density. But it seems to be a good enough start.
* Finalize AL_SOFT_gain_clamp_exChris Robinson2016-10-031-1/+1
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* Update the ambisonic coefficients for HRTFChris Robinson2016-09-261-15/+15
| | | | | | | This uses an AllRAD-derived decoder matrix for the high frequencies, which seems to improve positioning response. It also switches back to dual-band. The low frequencies appear to be unexpectedly quiet by comparison, but it's not that bad and can be tweaked later.
* Add a volume-adjust config option to adjust the source output volumeChris Robinson2016-09-242-1/+18
| | | | | | | | | Designed for apps that either don't change the listener's AL_GAIN, or don't allow the listener's AL_GAIN to go above 1. This allows the volume to still be increased further than such apps may allow, if users find it too quiet. Be aware that increasing this can easily cause clipping. The gain limit reported by AL_GAIN_LIMIT_SOFT is also affected by this.
* Recognize Headset formfactors as headphonesChris Robinson2016-09-211-1/+3
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* Fix EAX reverb effect output for HRTF and UHJChris Robinson2016-09-131-2/+2
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* Call ALfilterState_processC directlyChris Robinson2016-09-122-5/+0
| | | | | | It's the only implementation currently, so there's no point to having it stored as a function pointer in the filter struct. Even if there were SIMD versions, it'd be a global selection, not per-instance.
* Write to the correct outputs for extra reverb channelsChris Robinson2016-09-111-2/+2
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* Combine the reverb decorrelator delay line with the main delay lineChris Robinson2016-09-111-46/+26
| | | | | | | Since it was merely acting as an extension of it anyway, with the second delay line tap (for late reverb) copying attenuated samples to the decorrelator line that was being tapped off of. Just extend the delay line and offset the decorrelator taps to be relative to the late reverb tap.
* Don't warn if the desired default HRTF is already firstChris Robinson2016-09-101-4/+4
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* Reduce the volume for the HRTF ambisonic decoderChris Robinson2016-09-081-14/+14
| | | | | | | Since it's accumulating multiple HRIRs for two output speakers, it seems to be a better option to preserve the amplitude of the high-frequency decoder instead of increasing it, and reduce the amplitude of the low-frequency decoder to compensate.
* Only WARN if GetProcPath fails to find the binaryChris Robinson2016-09-081-1/+1
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* Properly check if /proc/cpuinfo openedChris Robinson2016-09-081-1/+1
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* Use a few more HRIRs for the HRTF B-Format decoderChris Robinson2016-09-071-19/+31
| | | | | 14 in total, an 8-point cube and a 6-point diamond shape, to help improve sound localization a bit. Incurs no real extra CPU cost once the IRs are built.
* Check for run-time NEON support by reading /proc/cpuinfoChris Robinson2016-09-071-2/+32
| | | | | | Less than ideal since documentations warn it may not list 'neon' even if it's really supported. However, the "proper" APIs to check for NEON extensions don't seem to exist in my toolchain.
* Simplify mmdevapi's device name searchChris Robinson2016-09-071-17/+33
| | | | | Avoids converting each enumerated devid from WSTR to UTF-8, and instead just converts the device name from UTF-8 to WSTR once if needed.
* Merge pull request #65 from Dmytry/dmytry_github_masterkcat2016-09-072-8/+46
|\ | | | | mmdevapi: Allow specifying output device by it's audio endpoint GUID …
| * mmdevapi: Allow specifying output device by it's audio endpoint GUID or by ↵Dmytry Lavrov2016-09-062-8/+46
| | | | | | | | the device id string (Oculus VR api requires you to play back on a specific device).
* | Properly align 16-bit fields in the Hrtf structChris Robinson2016-09-071-10/+20
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* Make the SelectMixer function sharableChris Robinson2016-09-063-26/+12
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* Use the optimized mixing functions for reverb outputChris Robinson2016-09-061-83/+117
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* Use deinterlaced buffers for the intermediate reverb storageChris Robinson2016-09-061-119/+137
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* Remove use of DECL_CONSTChris Robinson2016-09-062-5/+5
| | | | | No idea if it was really gaining us anything, but removing it fixes a crash I was getting with libs built with Clang.
* Do reverb modulation before band-pass filteringChris Robinson2016-09-061-7/+6
| | | | | | | Ideally the band-pass should probably happen closer to output, like gain is. However, doing that would require 16 filters (4 early + 4 late channels, each with a low-pass and high-pass filter), compared to the two needed to do it on input.
* Rename some variables for clarityChris Robinson2016-09-061-28/+30
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* Use more correct cube decoder matricesChris Robinson2016-09-052-16/+16
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* Do multiple samples at once for reverb modulationChris Robinson2016-09-051-38/+49
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* Use a predefined identity matrixChris Robinson2016-09-054-30/+13
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* Correct a comment about B-Format conversionChris Robinson2016-09-051-1/+1
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* Rename MatrixMixerFunc to RowMixerFuncChris Robinson2016-09-025-11/+11
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* Calculate a variable closer to where it's usedChris Robinson2016-09-011-5/+5
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* Use MixMatrixRow to upsample the split frequency bands to the outputChris Robinson2016-09-011-30/+12
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* Always load HRTF files through memory pointersChris Robinson2016-08-311-261/+134
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* Add some helper wrappers to mmap filesChris Robinson2016-08-312-0/+120
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* Add a compile-time macro to use dual-band ambisonic HRTF processingChris Robinson2016-08-301-12/+36
| | | | | Use single-band processing for now, to see if dual-band is causing a drop in quality at all.
* Simplify the ambisonic up-samplerChris Robinson2016-08-301-65/+99
| | | | | It still behaves the same, although now combines the separate decode+encode matrices into a transcode matrix (one per frequency band).
* Remove the upper limit from AL_MIN_GAIN and AL_MAX_GAINChris Robinson2016-08-291-2/+2
| | | | As per the current AL_SOFT_gain_clamp_ex proposal.
* Add a query for the maximum source gain limitChris Robinson2016-08-281-0/+1
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* Clamp the maximum mixing gain boost to 16Chris Robinson2016-08-271-10/+11
| | | | | | The combined source and listener gains now can't exceed a multiplier of 16 (~24dB). This is to avoid mixes getting out of control with large volume boosts, which reduces the effective precision given by floating-point.
* Allow sources to play while alcSuspendContext is in effectChris Robinson2016-08-261-3/+3
| | | | | | | | | | | | | | This appears to be how Creative's Windows drivers handle it, and is necessary for at least the Windows version of UT2k4 (otherwise it tries to play a source while suspended, checks and sees it's stopped, then kills it before it's given a chance to start playing). Consequently, the internal properties it gets mixed with are determined by what the source properties are at the time of the play call, and the listener properties at the time of the suspend call. This does not change alDeferUpdatesSOFT, which will still hold the play state change until alProcessUpdatesSOFT.
* Properly defer effect slot changesChris Robinson2016-08-252-3/+6
| | | | | | | | Note that this now also causes all playing sources to update when an effect slot is updated. This is a bit wasteful, as it should only need to re-update sources that are using the effect slot (and only when a relevant property is changed), but it's good enough. Especially with deferring since all playing sources are going to get updated on the process call anyway.
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