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* Remove the fastf2u conversion functionChris Robinson2017-06-271-5/+0
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* Clean up some messy rounding codeChris Robinson2017-06-261-0/+18
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* Use the bsinc resampler for the converterChris Robinson2017-06-251-0/+2
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* "Convert" the QSA backend to the new APIChris Robinson2017-06-181-19/+0
| | | | | | | | | | I say "convert" because it takes the lazy way and essentially just embeds the wrappers into the backend. It's done this way because I lack the means to check any changes, even syntactically. This also means the device's ExtraData field is still needed. However, this does mean all the backends are now using the new API. Code related to the old interface can now be removed.
* Make the dithering depth configurableChris Robinson2017-06-171-1/+1
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* Fix source sends' initial HF absorption and decay calculationChris Robinson2017-05-271-1/+1
| | | | | | | | 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.
* Add a new compressor/limiterChris Robinson2017-05-272-24/+30
| | | | | | 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.
* Finalize ALC_SOFT_output_limiterChris Robinson2017-05-251-5/+0
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* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-232-0/+7
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* Reduce the amount of variables that hold the same valueChris Robinson2017-05-211-2/+2
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* Add a method to copy a filter's coefficientsChris Robinson2017-05-211-0/+9
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* Use a macro to specify the decay target gainChris Robinson2017-05-211-0/+3
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* Apply more proper air absorption to the wet pathChris Robinson2017-05-191-0/+2
| | | | | | | | 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.
* 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.
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