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* Reimplement bilinear interpolation between HRIRsChris Robinson2017-05-011-25/+82
| | | | | | Some data sets are just too sparse, having noticeably few measurements to properly handle slowly panning sources. Although not perfect, bilinearly interpolating the HRIR measurements improves the positional accuracy.
* Allow querying the output limiter stateChris Robinson2017-04-301-2/+12
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* Start an extension to toggle the output limiterChris Robinson2017-04-301-5/+18
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* Don't do more reverb samples than there are to fade.Chris Robinson2017-04-281-2/+6
| | | | This avoids having to clamp the fade value when incrementing it.
* Fade HRTF coefficients over 64 samples at mostChris Robinson2017-04-281-40/+69
| | | | | | This greatly improves HRTF performance since the dual-mix only applies to the 64-sample coefficient transition. So rather than doubling the full mix, it only doubles 64 samples out of the full mix.
* Combine mostly duplicate functionsChris Robinson2017-04-271-121/+50
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* Properly update the delay line offsets when fading is doneChris Robinson2017-04-271-28/+44
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* Write directly to the output for HRTFChris Robinson2017-04-271-51/+26
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* Implement a limiter on the device outputChris Robinson2017-04-262-4/+70
| | | | | | This reduces the output volume when the mixed samples extend outside of -1,+1, to prevent excessive clipping. It can reduce the volume by -80dB in 50ms, and increase it by +80dB in 1s (it will not go below -80dB or above 0dB).
* Reverb code updateChris Robinson2017-04-261-824/+1213
| | | | | | | | | | | | | | This update modifies the reverb in numerous ways. The 3-series, 4-parallel all-pass is replaced with a Gerzon vector all-pass. The vector all-pass is also applied to the early reflections, to help with the initial diffusion in the late reverb. The late reverb filter and feedback attenuation has been replaced with a dual first-order equalization filter, which controls the low and high frequencies with individual low-pass/high-shelf and high-pass/low-shelf filters with gain control. Additionally, delay lines now have the ability to blend the output between two offsets, to help smooth out the transition for when the delay changes (without such, it could result in undesirable clicks and pops).
* Skip mixing the fade out step when starting silentChris Robinson2017-04-251-12/+19
| | | | | Unfortunately it can't skip mixing the fade in when going to silence because the history needs to be up to date.
* Remove const from _Atomic vars to make Clang happyChris Robinson2017-04-212-9/+13
| | | | | | | | Clang does not allow using C11's atomic_load on const _Atomic variables. Previously it just disabled use of C11 atomics if atomic_load didn't work on a const _Atomic variable, but I think I'd prefer to have Clang use C11 atomics for the added features (more explicit memory ordering) even if it means a few instances of breaking const.
* Add the ability to change the source resamplerChris Robinson2017-04-211-1/+2
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* Correctly enable the ambisonic upsampler for HOA outputChris Robinson2017-04-211-1/+1
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* Add a method to enumerate resamplersChris Robinson2017-04-211-0/+7
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* Store the resampler as part of the sourceChris Robinson2017-04-212-3/+3
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* Make the default resampler a variableChris Robinson2017-04-201-8/+9
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* Try NEON mixers before SSEChris Robinson2017-04-201-9/+8
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* Make the buffer list next pointer atomicChris Robinson2017-04-192-4/+6
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* Fix a mixed-sign-comparison warning on MSVCChris Robinson2017-04-191-1/+1
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* Allocate a new context's voices after updating the device paramsChris Robinson2017-04-191-26/+15
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* Check for the upsampler to determine if HRTF uses HOAChris Robinson2017-04-181-15/+12
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* Use a different way to get the size of structs with flexible array membersChris Robinson2017-04-185-10/+8
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* Store the source queue head in the voice to signify loopingChris Robinson2017-04-181-12/+9
| | | | | This removes the need to access a couple more source fields in the mixer, and also makes the looping and queue fields non-atomic.
* Remove unnecessary functions in the JACK backendChris Robinson2017-04-171-19/+8
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* Store the source prop updates with the mixer voiceChris Robinson2017-04-172-30/+70
| | | | Also move its declaration and rename it for consistency.
* Trace unhandled device reset attributesChris Robinson2017-04-161-111/+109
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* Correctly handle the attribute array size for alcGetInteger64vSOFTChris Robinson2017-04-151-2/+2
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* Implement capture support in the OpenSL backendChris Robinson2017-04-151-1/+379
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* Allow increasing the maximum source limitChris Robinson2017-04-141-15/+59
| | | | | | | | If the requested number of mono and stereo sources exceeds 256, the source limit will be expanded. Any config file setting overrides this. If the device is reset to have fewer sources than are currently allocated, excess sources will remain and be usable as normal, but no more can be generated until enough are delated to go back below the limit.
* Use separate atomic macros for pointersChris Robinson2017-04-146-32/+30
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* Mark some pointers with restrictChris Robinson2017-04-131-2/+4
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* Use ALsizei for the fir4 resampler fractionChris Robinson2017-04-121-1/+1
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* Select NEON when available before SSEChris Robinson2017-04-121-4/+4
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* Store the ambisonic order separate from the channel enumChris Robinson2017-04-1217-279/+280
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* Remove an unnecessary variableChris Robinson2017-04-111-6/+2
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* Use the correct channel conversion functionsChris Robinson2017-04-111-16/+16
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* Use the converters to enable mmdevapi captureChris Robinson2017-04-111-42/+171
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* Add a mono<->stereo converterChris Robinson2017-04-112-0/+134
| | | | This converter always outputs floats, and uses energy-preserving scaling.
* Update the given source pointer in the sample converterChris Robinson2017-04-102-5/+5
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* Reduce the size of the temp input bufferChris Robinson2017-04-101-1/+1
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* Trace the capture device formatChris Robinson2017-04-101-0/+4
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* Add a sample converterChris Robinson2017-04-103-10/+375
| | | | | | | | | | This is intended to do conversions for interleaved samples, and supports changing from one DevFmtType to another as well as resampling. It does not handle remixing channels. The mixer is more optimized to use the resampling functions directly. However, this should prove useful for recording with certain backends that won't do the conversion themselves.
* Convert the CoreAudio backend to the updated backend APIChris Robinson2017-04-093-179/+284
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* Handle the source offset fraction as an ALsizeiChris Robinson2017-04-088-53/+52
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* Pre-compute the sinc4 resampler coefficient tableChris Robinson2017-04-085-104/+4112
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* Clean up some formattingChris Robinson2017-04-081-5/+5
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* Try to write the full configured buffer length with PulseAudioChris Robinson2017-04-081-15/+37
| | | | | This basically ignores tlength even if it's smaller than what was requested. It keeps up-to-date with minreq changes too now, in case that happens.
* Overwrite the old search path with the new oneChris Robinson2017-04-081-1/+1
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* Combine a couple loopsChris Robinson2017-04-071-28/+24
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