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* Replace some ALvoid with voidChris Robinson2017-01-181-1/+1
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* Use ALsizei in more placesChris Robinson2017-01-181-33/+33
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* Pass the left and right buffers to the hrtf mixers directlyChris Robinson2017-01-171-4/+6
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* Use ALsizei and ALint for sizes and offsets with resamplers and filtersChris Robinson2017-01-161-1/+1
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* Use ALsizei for sizes and offsets with the mixerChris Robinson2017-01-161-3/+3
| | | | | | 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 separate macros for atomics that don't take a memory orderChris Robinson2016-12-201-2/+2
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* Update some atomic memory orderingChris Robinson2016-11-211-2/+2
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* Remove an unnecessary intermediate variableChris Robinson2016-11-021-7/+5
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* Add some more 'restrict' keywordsChris Robinson2016-10-061-1/+1
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* Pass current and target gains directly for mixingChris Robinson2016-10-051-86/+16
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* Make some pointer-to-array parameters constChris Robinson2016-10-041-0/+13
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* Make the SelectMixer function sharableChris Robinson2016-09-061-9/+9
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* Combine related members into a structChris Robinson2016-08-241-1/+1
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* Don't store the looping state in the voiceChris Robinson2016-07-311-1/+1
| | | | | Certain operations on the buffer queue depend on the loop state to behave properly, so it should not be deferred until the async voice update occurs.
* Fix use of a loop varChris Robinson2016-07-251-9/+11
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* Move the input channel array out of the DirectParams and SendParamsChris Robinson2016-07-131-20/+18
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-111-14/+14
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* Avoid function calls to get the HRTF sample rate and IR sizeChris Robinson2016-07-071-1/+1
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* Make the source position calues atomicChris Robinson2016-05-191-7/+7
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* Provide asynchronous property updates for sourcesChris Robinson2016-05-141-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | This necessitates a change in how source updates are handled. Rather than just being able to update sources when a dependent object state is changed (e.g. a listener gain change), now all source updates must be proactively provided. Consequently, apps that do not utilize any deferring (AL_SOFT_defer_updates or alcSuspendContext/alcProcessContext) may utilize more CPU since it'll be filling out more update containers for the mixer thread to use. The upside is that there's less blocking between the app's calling thread and the mixer thread, particularly for vectors and other multi-value properties (filters and sends). Deferring behavior when used is also improved, since updates that shouldn't be applied yet are simply not provided. And when they are provided, the mixer doesn't have to ignore them, meaning the actual deferring of a context doesn't have to synchrnously force an update -- the process call will send any pending updates, which the mixer will apply even if another deferral occurs before the mixer runs, because it'll still be there waiting on the next mixer invocation. There is one slight bug introduced by this commit. When a listener change is made, or changes to multiple sources while updates are being deferred, it is possible for the mixer to run while the sources are prepping their updates, causing some of the source updates to be seen before the other. This will be fixed in short order.
* Use the real output's left and right channels with HRTFChris Robinson2016-03-111-1/+7
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* Only calculate steps for the used coefficientsChris Robinson2016-02-141-1/+1
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* Calculate HRTF stepping params right before mixingChris Robinson2016-02-141-4/+36
| | | | | This means we track the current params and the target params, rather than the target params and the stepping. This closer matches the non-HRTF mixers.
* Calculate channel gain stepping just before mixingChris Robinson2016-02-141-8/+91
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* Mix to multichannel for effectsChris Robinson2016-01-281-1/+1
| | | | | | This mixes to a 4-channel first-order ambisonics buffer. With ACN ordering and N3D scaling, this makes it easy to remain compatible with effects that only care about mono input since channel 0 is an unattenuated mono signal.
* Move the bsincTable to a separate fileChris Robinson2015-11-101-988/+0
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* Update the bsinc tableChris Robinson2015-11-101-966/+967
| | | | Precision is increased to cover the full 32-bit float range.
* Cast a double->float return to silence MSVCChris Robinson2015-11-061-1/+1
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* Implement a band-limited sinc resamplerChris Robinson2015-11-051-1/+997
| | | | | | | | This is essentially a 12-point sinc resampler, unless it's resampling to a rate higher than the output, at which point it will vary between 12 and 24 points and do anti-aliasing to avoid/reduce frequencies going over nyquist. Code provided by Christopher Fitzgerald.
* Change the Kaiser rippling limit to -60dBChris Robinson2015-11-041-2/+2
| | | | | This improves the transition cutoff, shortening its width and reducing the amount of error.
* Replace the Lanczos window with Kaiser for the sinc resamplerChris Robinson2015-11-041-17/+87
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* Update filter histories even when they're not usedChris Robinson2015-10-241-0/+4
| | | | | | If the filter properties are continually updated, and the HF or LF gain goes from <1, to 1, and later back to <1, the history shouldn't hold stale values from before it was at 1.
* Set XYZ channel gains for source sends to 0Chris Robinson2015-10-231-18/+0
| | | | | It's cleaner to just set the gains to 0 rather than to special-case B-Format in the mixer.
* Use one send gain per buffer channelChris Robinson2015-10-231-1/+1
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* Use a constant value for the post-position paddingChris Robinson2015-10-151-33/+20
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* Store the source's previous samples with the voiceChris Robinson2015-10-151-92/+32
| | | | | | This helps avoid different results when looping is toggled within a couple samples of the loop point, or when a processed buffer is removed while the source is only a couple samples into the next buffer.
* Replace the sinc6 resampler with sinc8, and make SSE versionsChris Robinson2015-10-111-13/+23
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* Move the FIR4 from SSE2 to SSE3Chris Robinson2015-10-111-3/+3
| | | | | SSE3 can avoid the slow _MM_TRANSPOSE_PS4 call thanks to the inclusion of horizontal adds.
* Use doubles to calculate the Lanczos coefficientsChris Robinson2015-10-091-20/+20
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* Combine two arraysChris Robinson2015-10-011-13/+11
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* Move the resampler stuff to mixer.c where it's usedChris Robinson2015-10-011-0/+53
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* Implement a 6-point sinc-lanczos filterChris Robinson2015-09-291-12/+26
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* Replace the cubic resampler with a 4-point sinc/lanczos filterChris Robinson2015-09-271-10/+19
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* Don't keep selecting the mixer to useChris Robinson2015-09-271-30/+32
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* Increase the max pitch to 255Chris Robinson2015-09-261-0/+3
| | | | | | | Note that this is the multiple above the device sample rate, rather than the source property limit. It could theoretically be increased to 511 by testing against UINT_MAX instead of INT_MAX, since the increment and positions are using unsigned integers. I'm just being paranoid about overflows.
* Move HRTF params and state closer togetherChris Robinson2015-02-091-2/+2
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* Do up to 256 samples at a time with multi-step loopsChris Robinson2014-12-181-2/+2
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* Inline a couple functionsChris Robinson2014-12-181-2/+2
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* Offset to the buffer's channel start firstChris Robinson2014-12-181-3/+6
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* Assert that there's a buffer for mixingChris Robinson2014-12-171-0/+1
| | | | For Clang's static analysis.
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