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* Finalize AL_SOFT_source_spatializeChris Robinson2017-05-111-6/+0
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* Reduce LIMITER_VALUE_MAXChris Robinson2017-05-091-1/+1
| | | | | | 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 Robinson2017-05-081-2/+6
| | | | | This helps keep the squared sum stable over larger updates, also avoiding the need to keep recalculating it.
* Update AL_SOURCE_SPATIALIZE_SOFT valueChris Robinson2017-05-061-1/+1
| | | | | | Though it didn't strictly clash since it was for a different component (global state vs source property), 0x1213 was used by AL_RESAMPLER_NAME_SOFT. Probably best to avoid duplicate property values regardless.
* Calculate the output limiter gain using the RMSChris Robinson2017-05-052-1/+20
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* Rename RollOff to RolloffChris Robinson2017-05-052-2/+2
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* Start an extension to change the source's spatialize propertyChris Robinson2017-05-052-3/+9
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* Add a property to force source spatialization on or offChris Robinson2017-05-042-0/+8
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* Finalize AL_SOFT_source_resamplerChris Robinson2017-05-031-13/+0
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* Add a mixing function to blend HRIRsChris Robinson2017-05-031-0/+5
| | | | | | This is a bit more efficient than calling the normal HRTF mixing function twice, and helps solve the problem of the values generated from convolution not being consistent with the new HRIR.
* Change some ALuint parameters to ALsizeiChris Robinson2017-05-021-2/+2
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* Rename VOICE_IS_HRTF to VOICE_HAS_HRTFChris Robinson2017-05-021-1/+1
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* Rename 'moving' flag to 'fading'Chris Robinson2017-05-021-2/+2
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* Start an extension to toggle the output limiterChris Robinson2017-04-301-0/+5
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* Implement a limiter on the device outputChris Robinson2017-04-261-0/+2
| | | | | | 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 Robinson2017-04-211-0/+14
| | | | | | | | 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.
* Use more sensible values for the source resampler enumsChris Robinson2017-04-211-4/+4
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* Add a method to enumerate resamplersChris Robinson2017-04-212-0/+15
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* Store the resampler as part of the sourceChris Robinson2017-04-212-14/+17
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* Make the default resampler a variableChris Robinson2017-04-201-2/+1
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* Make the buffer list next pointer atomicChris Robinson2017-04-191-1/+2
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* Allocate a new context's voices after updating the device paramsChris Robinson2017-04-191-0/+1
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* Use a different way to get the size of structs with flexible array membersChris Robinson2017-04-181-0/+6
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* Store the source queue head in the voice to signify loopingChris Robinson2017-04-182-6/+10
| | | | | This removes the need to access a couple more source fields in the mixer, and also makes the looping and queue fields non-atomic.
* Store the source prop updates with the mixer voiceChris Robinson2017-04-172-59/+59
| | | | Also move its declaration and rename it for consistency.
* Close some gaps in enum valuesChris Robinson2017-04-161-2/+2
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* Remove some unnecessary parenthesisChris Robinson2017-04-161-1/+1
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* Use ALsizei for the fir4 resampler fractionChris Robinson2017-04-121-1/+1
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* Store the ambisonic order separate from the channel enumChris Robinson2017-04-121-8/+5
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* Add a sample converterChris Robinson2017-04-101-0/+10
| | | | | | | | | | 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 Robinson2017-04-091-3/+0
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* Handle the source offset fraction as an ALsizeiChris Robinson2017-04-081-2/+2
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* Pre-compute the sinc4 resampler coefficient tableChris Robinson2017-04-081-4/+3
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* Reference count HRTFs and unload them when unusedChris Robinson2017-04-061-1/+1
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* Load HRTF files as neededChris Robinson2017-04-051-1/+1
| | | | | Currently only applies to external files, rather than embedded datasets. Also, HRTFs aren't unloaded after being loaded, until library shutdown.
* Store the loaded hrtf entry container in the enumerated hrtf entryChris Robinson2017-04-051-5/+6
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* Remove the (u)byte3 sample formatsChris Robinson2017-03-311-2/+0
| | | | | They're not accessible since the removal of the buffer_samples extension, and were kind of clunky to work with as 24-bit packed values.
* Use an array of pointers for effects instead of a linked listChris Robinson2017-03-272-3/+7
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* Use an atomic flag to mark auxiliary effect slot updatesChris Robinson2017-03-231-2/+3
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* Use an atomic flag to test if a source needs to updateChris Robinson2017-03-201-1/+1
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* Don't defer source state or offset changesChris Robinson2017-03-192-11/+1
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* Store the HRIR coeff pointer and delays directly in MixHrtfParamsChris Robinson2017-03-122-4/+6
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* Rework HRTF coefficient fadingChris Robinson2017-03-112-11/+9
| | | | | | | | | | | | | | | 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.
* Make the voice's 'moving' state a bitflagChris Robinson2017-03-111-5/+4
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* Allocate as many channels for DirectHrtfState as neededChris Robinson2017-03-111-2/+4
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* Dynamically allocate the device's HRTF stateChris Robinson2017-03-101-13/+14
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* Implement NFC filters for Ambisonic renderingChris Robinson2017-03-102-1/+15
| | | | | | | | | | | | | | 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).
* Move ALvoice declaration to alu.hChris Robinson2017-03-092-55/+55
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* Remove unnecessary atomic membersChris Robinson2017-03-083-62/+62
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* Store the channel count and sample size in the voiceChris Robinson2017-03-071-4/+7
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