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* Restrict ClampedDist to RefDistance for invalid distance attenuationChris Robinson2017-05-301-4/+10
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* Fix source sends' initial HF absorption and decay calculationChris Robinson2017-05-271-21/+15
| | | | | | | | The HF absorption is applied given the source distance, as relative to the source's immediate environment, with additional absorption being applied given the room/reverb environment. This does double up the amount of absorption compared to the dry path, but it can be assumed the initial reflections travel a longer distance.
* Use normal air absorption for the sendsChris Robinson2017-05-271-1/+1
| | | | | Applies just for the normal air absorption which uses the air absorption factor, not the automated decay applied when WetGainAuto is enabled.
* Add a new compressor/limiterChris Robinson2017-05-271-93/+2
| | | | | | This is just for the output limiter right now, but in the future can be used for the compressor EFX effect. The parameters are also hardcoded, but can be made configurable after 1.18.
* Apply distance compensation separatelyChris Robinson2017-05-251-43/+66
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* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-231-45/+138
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* Reduce the amount of variables that hold the same valueChris Robinson2017-05-211-8/+6
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* Avoid calculating the filter coefficients multiple timesChris Robinson2017-05-211-24/+36
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* Use a macro to specify the decay target gainChris Robinson2017-05-211-5/+5
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* Use more correct doppler shift calculationsChris Robinson2017-05-201-11/+24
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* Restore spec-defined cone behavior for auxiliary sendsChris Robinson2017-05-201-26/+23
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* Apply more proper air absorption to the wet pathChris Robinson2017-05-191-21/+53
| | | | | | | | This properly accounts for the room rolloff factor for normal air absorption (which makes it none by default, like distance attenuation), and uses the reverb's decay time, decay hf ratio, decay hf limit, and room air absorption properties to calculate an initial hf decay with the WetGainAuto flag. This mirrors the behavior of the initial distance decay.
* Reorganize some code to have fewer temporariesChris Robinson2017-05-171-39/+34
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* Limit the dry and wet path filter gains to -60dBChris Robinson2017-05-151-4/+4
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* Improve distance-related absorption and decay attenuationChris Robinson2017-05-151-15/+14
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* Reduce LIMITER_VALUE_MAXChris Robinson2017-05-091-6/+12
| | | | | | The previous value couldn't actually be expressed as a float and got rounded up to the next whole number value, leaving the potential for an overrun in the squared sum.
* Store the output limiter values as fixed-point integersChris Robinson2017-05-081-11/+6
| | | | | This helps keep the squared sum stable over larger updates, also avoiding the need to keep recalculating it.
* Apply attenuation when downmixing multi-channel sounds for panningChris Robinson2017-05-071-10/+26
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* Calculate the output limiter gain using the RMSChris Robinson2017-05-051-23/+59
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* Rename RollOff to RolloffChris Robinson2017-05-051-4/+4
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* Add a property to force source spatialization on or offChris Robinson2017-05-041-3/+5
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* Handle attenuated sources in CalcPanningAndFiltersChris Robinson2017-05-041-377/+442
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* Calculate the multi-channel panning in a separate functionChris Robinson2017-05-041-249/+249
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* Rename VOICE_IS_HRTF to VOICE_HAS_HRTFChris Robinson2017-05-021-4/+4
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* Implement a limiter on the device outputChris Robinson2017-04-261-1/+59
| | | | | | 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).
* Remove const from _Atomic vars to make Clang happyChris Robinson2017-04-211-1/+1
| | | | | | | | 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.
* Store the resampler as part of the sourceChris Robinson2017-04-211-0/+2
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* Make the buffer list next pointer atomicChris Robinson2017-04-191-1/+1
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* Use a different way to get the size of structs with flexible array membersChris Robinson2017-04-181-1/+1
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* Store the source prop updates with the mixer voiceChris Robinson2017-04-171-9/+34
| | | | Also move its declaration and rename it for consistency.
* Use separate atomic macros for pointersChris Robinson2017-04-141-6/+6
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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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* Constify a variableChris Robinson2017-04-011-3/+3
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* Make a pointer restrict and assume alignedChris Robinson2017-04-011-1/+1
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* Convert float samples to integer using a power-of-2 multipleChris Robinson2017-03-311-13/+9
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* Use an array of pointers for effects instead of a linked listChris Robinson2017-03-271-19/+15
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* Increase the filter slope to -12dB/octaveChris Robinson2017-03-161-8/+8
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* Store the HRIR coeff pointer and delays directly in MixHrtfParamsChris Robinson2017-03-121-1/+0
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* Rework HRTF coefficient fadingChris Robinson2017-03-111-3/+7
| | | | | | | | | | | | | | | This improves fading between HRIRs as sources pan around. In particular, it improves the issue with individual coefficients having various rounding errors in the stepping values, as well as issues with interpolating delay values. It does this by doing two mixing passes for each source. First using the last coefficients that fade to silence, and then again using the new coefficients that fade from silence. When added together, it creates a linear fade from one to the other. Additionally, the gain is applied separately so the individual coefficients don't step with rounding errors. Although this does increase CPU cost since it's doing two mixes per source, each mix is a bit cheaper now since the stepping is simplified to a single gain value, and the overall quality is improved.
* Allocate as many channels for DirectHrtfState as neededChris Robinson2017-03-111-1/+1
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* Dynamically allocate the device's HRTF stateChris Robinson2017-03-101-9/+8
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* Implement NFC filters for Ambisonic renderingChris Robinson2017-03-101-10/+54
| | | | | | | | | | | | | | NFC filters currently only work when rendering to ambisonic buffers, which includes HQ rendering and ambisonic output. There are two new config options: 'decoder/nfc' (default on) enables or disables use of NFC filters globally, and 'decoder/nfc-ref-delay' (default 0) specifies the reference delay parameter for NFC-HOA rendering with ambisonic output (a value of 0 disables NFC). Currently, NFC filters rely on having an appropriate value set for AL_METERS_PER_UNIT to get the correct scaling. HQ rendering uses the averaged speaker distances as a control/reference, and currently doesn't correct for individual speaker distances (if the speakers are all equidistant, this is fine, otherwise per-speaker correction should be done as well).
* Remove unnecessary atomic membersChris Robinson2017-03-081-109/+86
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* Make the voice's source pointer atomicChris Robinson2017-03-051-6/+6
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* Add a boolean to specify if a voice should be playingChris Robinson2017-03-021-10/+9
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* Dynamically allocate the channel delay buffersChris Robinson2017-02-281-6/+6
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* Move the current buffer queue entry and play position to the voiceChris Robinson2017-02-271-6/+3
| | | | | | | | | | | | | | This has a couple behavioral changes. First and biggest is that querying AL_BUFFERS_PROCESSED from a source will always return all buffers processed when in an AL_STOPPED state. Previously all buffers would be set as processed when first becoming stopped, but newly queued buffers would *not* be indicated as processed. That old behavior was not compliant with the spec, which unequivocally states "On a source in the AL_STOPPED state, all buffers are processed." Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return 0. Previously it would return the current "active" buffer in the queue, but there's no basis for that in the spec.
* Ensure a non-playing or -paused source does not use a mixing voiceChris Robinson2017-02-251-1/+4
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* Alter how panpot/pair-wise panning worksChris Robinson2017-02-231-28/+10
| | | | | | | | | | | | | | | | | | | This change allows pair-wise panning to mostly go through the normal ambisonic panning methods, with one special-case. First, a term is added to the stereo decoder matrix's X coefficient so that a centered sound is reduced by -3dB on each output channel. Panning in front creates a similar gain response to the typical L = sqrt(1-pan) R = sqrt(pan) for pan = [0,1]. Panning behind the listener can reduce (up to) an additional -10dB, creating a audible difference between front and back sounds as if simulating head obstruction. Secondly, as a special-case, the source positions are warped when calculating the ambisonic coefficients so that full left panning is reached at -30 degrees and full right at +30 degrees. This is to retain the expected 60-degree stereo width. This warping does not apply to B-Format buffer input, although it otherwise has the same gain responses.
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