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path: root/OpenAL32/Include/alu.h
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* Use a voice flag to indicate it being staticChris Robinson2018-01-161-5/+5
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* Use a global RowMixerFuncChris Robinson2018-01-161-1/+1
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* Add min/max/clamp functions for size_tChris Robinson2018-01-161-0/+7
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* Avoid using macros to access anonymous structuresChris Robinson2018-01-111-25/+30
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* Don't return whether the bsinc filter cuts or notChris Robinson2018-01-101-1/+1
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* Use one macro to handle both resample padding sizesChris Robinson2018-01-091-6/+5
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* Use a separate function to get the cubic valueChris Robinson2018-01-071-0/+9
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* Remove the sinc4 tableChris Robinson2018-01-071-5/+0
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* Replace the sinc4 resampler with cubicChris Robinson2018-01-071-4/+0
| | | | | | | Turns out the C version of the cubic resampler is just slightly faster than even the SSE3 version of the FIR4 resampler. This is likely due to not using a 64KB random-access lookup table along with unaligned loads, both offseting the gains from SSE.
* Make MixSamples non-static globalChris Robinson2017-12-171-1/+2
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* Re-update effect slots when context properties changeChris Robinson2017-09-271-1/+0
| | | | | Also keep all free property update structs together in the context instead of per-object.
* Update the context state properties separatelyChris Robinson2017-09-271-0/+2
| | | | | | | | | | | | | The context state properties are less likely to change compared to the listener state, and future changes may prefer more infrequent updates to the context state. Note that this puts the MetersPerUnit in as a context state, even though it's handled through the listener functions. Considering the infrequency that it's updated at (generally set just once for the context's lifetime), it makes more sense to put it there than with the more frequently updated listener properties. The aforementioned future changes would also prefer MetersPerUnit to not be updated unnecessarily.
* Add an option to ignore the app's speed of sound for reverb decayChris Robinson2017-09-221-0/+1
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* Add a higher quality bsinc resampler using 24 sample pointsChris Robinson2017-08-271-5/+10
| | | | | | | This improves the transition width, allowing more of the higher frequencies remain audible. It would be preferrable to have an upper limit of 32 points instead of 48, to reduce the overall table size and the CPU cost for down- sampling.
* Rename the bsinc resampler to bsinc12Chris Robinson2017-08-251-2/+2
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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 Robinson2017-08-161-7/+11
| | | | Also rename the resampler functions to remove the unnecessary '32' token.
* Simplify bsinc filter storage in the filter stateChris Robinson2017-08-161-6/+5
| | | | | | | 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.
* Keep bsinc info together in a structChris Robinson2017-08-151-2/+1
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* Avoid re-selecting the direct HRTF mix functionChris Robinson2017-08-071-0/+2
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* Apply the output buffer offset before writing to itChris Robinson2017-07-151-1/+1
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* Use the bsinc resampler for the converterChris Robinson2017-06-251-0/+2
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* Add a new compressor/limiterChris Robinson2017-05-271-22/+0
| | | | | | 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.
* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-231-0/+3
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* Reduce the amount of variables that hold the same valueChris Robinson2017-05-211-2/+2
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* Use a macro to specify the decay target gainChris Robinson2017-05-211-0/+3
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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.
* Calculate the output limiter gain using the RMSChris Robinson2017-05-051-0/+18
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* Rename RollOff to RolloffChris Robinson2017-05-051-1/+1
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* Start an extension to change the source's spatialize propertyChris Robinson2017-05-051-3/+3
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* Add a property to force source spatialization on or offChris Robinson2017-05-041-0/+7
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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.
* 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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* Add a method to enumerate resamplersChris Robinson2017-04-211-0/+2
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* Store the resampler as part of the sourceChris Robinson2017-04-211-14/+16
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* Make the default resampler a variableChris Robinson2017-04-201-2/+1
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* Store the source queue head in the voice to signify loopingChris Robinson2017-04-181-3/+8
| | | | | 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 Robinson2017-04-171-2/+59
| | | | Also move its declaration and rename it for consistency.
* Use ALsizei for the fir4 resampler fractionChris Robinson2017-04-121-1/+1
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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.
* 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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* Store the HRIR coeff pointer and delays directly in MixHrtfParamsChris Robinson2017-03-121-1/+2
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* Rework HRTF coefficient fadingChris Robinson2017-03-111-11/+8
| | | | | | | | | | | | | | | 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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* Implement NFC filters for Ambisonic renderingChris Robinson2017-03-101-1/+8
| | | | | | | | | | | | | | 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 Robinson2017-03-091-1/+55
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* Ensure a non-playing or -paused source does not use a mixing voiceChris Robinson2017-02-251-1/+1
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