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* Meters per unit can't be 0Chris Robinson2017-08-211-2/+2
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* Properly clear the extra ChannelsPerOrder entriesChris Robinson2017-08-191-1/+1
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* Pass the filter entry to apply to resample_fir4Chris Robinson2017-08-181-3/+2
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* Store the sinc4 table in the filter stateChris Robinson2017-08-161-3/+11
| | | | Also rename the resampler functions to remove the unnecessary '32' token.
* Simplify bsinc filter storage in the filter stateChris Robinson2017-08-161-12/+3
| | | | | | | Rather than storing individual pointers to filter, scale delta, phase delta, and scale phase delta entries, per phase index, the new table layout makes it trivial to access the per-phase filter and delta entries given the base offset and coefficient count.
* Make the bsinc table layout more efficientChris Robinson2017-08-161-4/+4
| | | | | | | | | | | | | | | | The old layout separated filters, scale deltas, phase deltas, and scale phase deltas into separate segments that each contained a numbers of scale and phase entries, Since processing a sample needed a filter and one of each delta entry relating to a particular scale and phase, the memory needed would be spread across the whole table. And since subsequent samples would use a different phase, it would jump around the table a whole lot as well. The new layout packs the data in a way more consistent with its use. The filters, scale deltas, phase deltas, and scale phase deltas are interleaved, such that for a particular scale and phase, the filter and delta entries used are contiguous. And the phase entries for a particular scale are kept together, so the ~500 to ~1000 samples processed per source update stay within the same 3KB to 6KB area of the 70+KB table, which is much more cache friendly.
* Keep bsinc info together in a structChris Robinson2017-08-151-22/+9
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* Avoid re-selecting the direct HRTF mix functionChris Robinson2017-08-071-15/+22
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* Add a front-stablizer config option for surround sound modesChris Robinson2017-07-311-0/+59
| | | | | | | | | | | | This improves a stereo (front-left + front-right) sound "image" by generating a front-center channel signal. Done correctly, it helps reduce the comb effects and phase errors associated with using only two speakers to simulate center sounds. Note that it shouldn't be used if the front-center channel is already included in the positional audio mix (the dialog effect is okay). In general, it may actually be better to exclude the front-center channel from the positional audio mix and use this to generate front-center output.
* Cleanup output write functionsChris Robinson2017-07-271-32/+26
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* Apply the output buffer offset before writing to itChris Robinson2017-07-151-24/+24
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* Add the default auxiliary slot to the active slot arrayChris Robinson2017-07-131-16/+0
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* Use macros to set and restore the mixer FPU modeChris Robinson2017-07-131-5/+2
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* Store the default effect slot in the contextChris Robinson2017-07-131-18/+18
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* Remove the fastf2u conversion functionChris Robinson2017-06-271-2/+2
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* Clean up some loop variablesChris Robinson2017-06-251-29/+23
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* Use the bsinc resampler for the converterChris Robinson2017-06-251-1/+1
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* Remove an unnecessary variableChris Robinson2017-06-211-3/+2
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* Make the dithering depth configurableChris Robinson2017-06-171-14/+4
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* Apply dither separately from outputChris Robinson2017-06-171-98/+68
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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.
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