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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.
* Rename stereo-panning option to stereo-encodingChris Robinson2017-02-221-4/+4
| | | | Also rename the 'paired' value to 'panpot', and make it the default.
* Limit filter gains to -24dBChris Robinson2017-02-221-32/+26
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* Reduce some codeChris Robinson2017-02-211-61/+33
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* Make the voices' Send[] array dynamically sizedChris Robinson2017-02-211-11/+11
| | | | | The voices are still all allocated in one chunk to avoid memory fragmentation. But they're accessed as an array of pointers since the size isn't static.
* Apply distance compensation when writing to the outputChris Robinson2017-02-191-13/+38
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* Don't use periphonic FOA when the HOA decoder is not periphonicChris Robinson2017-02-191-4/+3
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* Always lock the device backend before calling aluMixDataChris Robinson2017-02-181-2/+0
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* Reorganize ALvoice membersChris Robinson2017-02-151-81/+87
| | | | | This places the Send[] array at the end of the struct, making it easier to handle dynamically.
* Make ALsourceProps' Send array dynamically sizedChris Robinson2017-02-141-11/+11
| | | | | | ALsourceProps' Send[] array is placed at the end of the struct, and given an indeterminate size. Extra space is allocated at the end of each struct given the number of auxiliary sends set for the device.
* Make the source state atomicChris Robinson2017-02-131-4/+6
| | | | | Since it's modified by the mixer when playback is ended, a plain struct member isn't safe.
* Put BsincState in a generic unionChris Robinson2017-02-131-2/+2
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* Fix more uses of unsigned sizes and offsetsChris Robinson2017-02-101-5/+5
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* Replace some ALvoid with voidChris Robinson2017-01-181-2/+2
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* Use ALsizei in more placesChris Robinson2017-01-181-1/+1
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* Pass the left and right buffers to the hrtf mixers directlyChris Robinson2017-01-171-14/+16
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* Use ALsizei for sizes and offsets with the mixerChris Robinson2017-01-161-1/+1
| | | | | | Unsigned 32-bit offsets actually have some potential overhead on 64-bit targets for pointer/array accesses due to rules on integer wrapping. No idea how much impact it has in practice, but it's nice to be correct about it.
* Use second-order ambisonics for basic HRTF renderingChris Robinson2017-01-151-0/+7
| | | | | | This should improve positional quality for relatively low cost. Full HRTF rendering still only uses first-order since the only use of the dry buffer there is for first-order content (B-Format buffers, effects).
* Avoid duplicating code using a macroChris Robinson2016-12-211-38/+3
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* Use separate macros for atomics that don't take a memory orderChris Robinson2016-12-201-7/+7
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* Update some atomic memory orderingChris Robinson2016-11-211-6/+9
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* Don't interpolate between nearest HRIRsChris Robinson2016-10-091-4/+4
| | | | | | | | It still fades between HRIRs when it changes, but now it selects the nearest one instead of blending the nearest four. Due to the minimum-phase nature of the HRIRs, interpolating between delays lead to some oddities which are exasperated by the fading (and the fading is needed to avoid clicks and pops, and smooth out changes).
* Make some pointer-to-array parameters constChris Robinson2016-10-041-7/+6
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* Add a volume-adjust config option to adjust the source output volumeChris Robinson2016-09-241-1/+1
| | | | | | | | | 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.
* Use a predefined identity matrixChris Robinson2016-09-051-0/+7
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