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* Use a more specialized mixer function for B-Format to HRTFChris Robinson2016-08-121-0/+4
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* Move the input channel array out of the DirectParams and SendParamsChris Robinson2016-07-131-13/+9
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-111-6/+0
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* Use SSE for applying the HQ B-Format decoder matricesChris Robinson2016-05-311-0/+3
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* Use floats for the listener transformsChris Robinson2016-05-161-25/+0
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* Don't store the source's update method with the voiceChris Robinson2016-05-161-3/+0
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* Avoid separate updates to sources that should apply togetherChris Robinson2016-05-151-2/+0
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* Provide asynchronous property updates for sourcesChris Robinson2016-05-141-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | 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.
* Find a valid source buffer before updating the voiceChris Robinson2016-05-091-2/+3
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* Improve radius behavior with scaling of ambisonic coefficientsChris Robinson2016-04-241-5/+6
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* More directly map coefficients for ambisonic mixing buffersChris Robinson2016-04-151-3/+22
| | | | | | Instead of looping over all the coefficients for each channel with multiplies, when we know only one will have a non-0 factor for ambisonic mixing buffers, just index the one with a non-0 factor.
* Update some commentsChris Robinson2016-04-151-4/+5
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* Avoid mixing all coefficients together when only some are usedChris Robinson2016-04-151-1/+1
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* Avoid unnecessary loops for setting up effect slot b-format buffer mixingChris Robinson2016-04-141-0/+2
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* Move the InitRenderer method to panning.cChris Robinson2016-04-141-3/+13
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* Split aluInitPanning into separate functions for HRTF or UHJChris Robinson2016-04-141-0/+2
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* Add config options to enable the hq ambisonic decoderChris Robinson2016-03-161-2/+1
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* Add a dual-band ambisonic decoderChris Robinson2016-03-151-1/+2
| | | | | | | | | | This uses a virtual B-Format buffer for mixing, and then uses a dual-band decoder for improved positional quality. This currently only works with first- order output since first-order input (from the AL_EXT_BFROMAT extension) would not sound correct when fed through a second- or third-order decoder. This also does not currently implement near-field compensation since near-field rendering effects are not implemented.
* Use the real output's left and right channels with HRTFChris Robinson2016-03-111-2/+2
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* Calculate HRTF stepping params right before mixingChris Robinson2016-02-141-7/+11
| | | | | 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 Robinson2016-02-141-9/+11
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* Rename ComputeBFormatGains to ComputeFirstOrderGainsChris Robinson2016-01-311-5/+5
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* Mix to multichannel for effectsChris Robinson2016-01-281-4/+6
| | | | | | 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.
* Separate calculating ambisonic coefficients from the panning gainsChris Robinson2016-01-251-14/+34
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* Use doubles for the constructed listener matrixChris Robinson2015-11-111-12/+37
| | | | | | This helps the stability of transforms to local space for sources that are at or near the listener. With a single-precision matrix, even FLT_EPSILON might not be enough to detect matching positions.
* Implement a band-limited sinc resamplerChris Robinson2015-11-051-4/+30
| | | | | | | | 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.
* Pass in the Q parameter for setting the filter parametersChris Robinson2015-11-011-12/+1
| | | | Also better handle the peaking filter gain.
* Fix a commentChris Robinson2015-11-011-1/+1
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* 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 Robinson2015-10-151-2/+5
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* Store the source's previous samples with the voiceChris Robinson2015-10-151-0/+3
| | | | | | 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 Robinson2015-10-111-4/+5
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* Implement a 6-point sinc-lanczos filterChris Robinson2015-09-291-2/+11
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* Replace the cubic resampler with a 4-point sinc/lanczos filterChris Robinson2015-09-271-3/+3
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* Don't keep selecting the mixer to useChris Robinson2015-09-271-1/+1
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* Increase the max pitch to 255Chris Robinson2015-09-261-1/+1
| | | | | | | 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.
* Fix updating listener params when forcing updatesChris Robinson2015-09-181-0/+2
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* Rename F_2PI to F_TAUChris Robinson2015-09-131-1/+1
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* Move HRTF params and state closer togetherChris Robinson2015-02-091-3/+3
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* Add missing alignas to CubicLUT declarationChris Robinson2015-01-131-1/+1
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* Remove some unnecessary restrict usesChris Robinson2014-12-241-7/+6
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* Use aluVector and aluMatrix in a couple more placesChris Robinson2014-12-161-1/+1
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* Add explicit matrix and vector types to operate withChris Robinson2014-12-161-0/+38
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* Use a lookup table to do cubic resamplingChris Robinson2014-12-151-9/+9
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* Transpose the cubic matrix opChris Robinson2014-12-151-6/+6
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* Remove IrSize from DirectParamsChris Robinson2014-11-291-1/+0
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* Move the voice's last position and gain out of the Hrtf containerChris Robinson2014-11-241-2/+3
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* Partially revert "Use a different method for HRTF mixing"Chris Robinson2014-11-231-1/+9
| | | | | | | | | | | | The sound localization with virtual channel mixing was just too poor, so while it's more costly to do per-source HRTF mixing, it's unavoidable if you want good localization. This is only partially reverted because having the virtual channel is still beneficial, particularly with B-Format rendering and effect mixing which otherwise skip HRTF processing. As before, the number of virtual channels can potentially be customized, specifying more or less channels depending on the system's needs.
* Rename Voice's NumChannels to OutChannelsChris Robinson2014-11-221-1/+1
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* Store the number of output channels in the voiceChris Robinson2014-11-221-0/+1
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