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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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* Remove an unnecessary union containerChris Robinson2014-11-221-3/+1
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* Use a different method for HRTF mixingChris Robinson2014-11-221-27/+1
| | | | | | | | | | | | | | | | | | | | | | | This new method mixes sources normally into a 14-channel buffer with the channels placed all around the listener. HRTF is then applied to the channels given their positions and written to a 2-channel buffer, which gets written out to the device. This method has the benefit that HRTF processing becomes more scalable. The costly HRTF filters are applied to the 14-channel buffer after the mix is done, turning it into a post-process with a fixed overhead. Mixing sources is done with normal non-HRTF methods, so increasing the number of playing sources only incurs normal mixing costs. Another benefit is that it improves B-Format playback since the soundfield gets mixed into speakers covering all three dimensions, which then get filtered based on their locations. The main downside to this is that the spatial resolution of the HRTF dataset does not play a big role anymore. However, the hope is that with ambisonics- based panning, the perceptual position of panned sounds will still be good. It is also an option to increase the number of virtual channels for systems that can handle it, or maybe even decrease it for weaker systems.
* Use a separate macro for the max output channel countChris Robinson2014-11-071-5/+5
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* Use a method to set omni-directional channel gainsChris Robinson2014-11-041-14/+7
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* Support B-Format source rotation with AL_ORIENTATIONChris Robinson2014-10-311-3/+4
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* Add preliminary AL_EXT_BFORMAT supportChris Robinson2014-10-311-0/+8
| | | | | Currently missing the AL_ORIENTATION source property. Gain stepping also does not work.
* Make ComputeAngleGains use ComputeDirectionalGainsChris Robinson2014-10-021-5/+5
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* Don't use ComputeAngleGains for SetGainsChris Robinson2014-10-021-1/+5
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* Use an ambisonics-based panning methodChris Robinson2014-09-301-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For mono sources, third-order ambisonics is utilized to generate panning gains. The general idea is that a panned mono sound can be encoded into b-format ambisonics as: w[i] = sample[i] * 0.7071; x[i] = sample[i] * dir[0]; y[i] = sample[i] * dir[1]; ... and subsequently rendered using: output[chan][i] = w[i] * w_coeffs[chan] + x[i] * x_coeffs[chan] + y[i] * y_coeffs[chan] + ...; By reordering the math, channel gains can be generated by doing: gain[chan] = 0.7071 * w_coeffs[chan] + dir[0] * x_coeffs[chan] + dir[1] * y_coeffs[chan] + ...; which then get applied as normal: output[chan][i] = sample[i] * gain[chan]; One of the reasons to use ambisonics for panning is that it provides arguably better reproduction for sounds emanating from between two speakers. As well, this makes it easier to pan in all 3 dimensions, with for instance a "3D7.1" or 8-channel cube speaker configuration by simply providing the necessary coefficients (this will need some work since some methods still use angle-based panpot, particularly multi-channel sources). Unfortunately, the math to reliably generate the coefficients for a given speaker configuration is too costly to do at run-time. They have to be pre- generated based on a pre-specified speaker arangement, which means the config options for tweaking speaker angles are no longer supportable. Eventually I hope to provide config options for custom coefficients, which can either be generated and written in manually, or via alsoft-config from user-specified speaker positions. The current default set of coefficients were generated using the MATLAB scripts (compatible with GNU Octave) from the excellent Ambisonic Decoder Toolbox, at https://bitbucket.org/ambidecodertoolbox/adt/
* Rename activesource to voiceChris Robinson2014-08-211-4/+4
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* Use a NULL source for inactive activesourcesChris Robinson2014-08-211-3/+7
| | | | Also only access the activesource's source field once per update.
* Combine the direct and send mixersChris Robinson2014-06-131-3/+3
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* Combine some dry and wet path typesChris Robinson2014-06-131-15/+6
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* Avoid a loop when updating the source position variablesChris Robinson2014-06-021-0/+2
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* Move the active source's offset out of the direct paramsChris Robinson2014-05-241-2/+0
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* Return a sample pointer from resamplersChris Robinson2014-05-191-2/+2
| | | | Both resampling and filtering now avoid copying samples when they no-op.
* Put per-channel filter properties togetherChris Robinson2014-05-191-8/+12
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* Move an HRTF mixer parameter and shorten a couple variable namesChris Robinson2014-05-181-3/+3
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* Don't pass the SendParams to the wet-path mixerChris Robinson2014-05-181-11/+13
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* Don't pass the DirectParams to the dry-path mixerChris Robinson2014-05-181-12/+13
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* Use different parameters for HRTF mixersChris Robinson2014-05-181-0/+4
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* Pass some DirectParams as function parametersChris Robinson2014-05-181-2/+3
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* Apply high-pass source filters as neededChris Robinson2014-05-171-0/+4
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* Add a flag to specify when the low-pass filter needs to applyChris Robinson2014-05-171-13/+21
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* Better pack HRTF mixing propertiesChris Robinson2014-05-031-11/+11
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* Clamp the current and target gain lower bound to epsilonChris Robinson2014-05-031-1/+1
| | | | | Should give a bit more wiggle room for the gain stepping to get lower than the silence threshold.
* Use C11 alignas when availableChris Robinson2014-04-191-4/+6
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* Make HRTF stepping values per-channelChris Robinson2014-04-051-2/+2
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* Remove the click removal buffers for auxiliary effect slotsChris Robinson2014-03-231-4/+1
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* Add gain stepping to the send mixersChris Robinson2014-03-231-1/+8
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* Remove the now-unneeded click removal buffers for the deviceChris Robinson2014-03-231-4/+1
| | | | | | They are still there for auxiliary sends. However, they should go away soon enough too, and then we won't have to mess around with calculating extra "predictive" samples in the mixer.
* Step mixing gains per-sample for non-HRTF mixingChris Robinson2014-03-231-2/+7
| | | | | | | | | | | | | | | | | | | | | | | | This fades the dry mixing gains using a logarithmic curve, which should produce a smoother transition than a linear one. It functions similarly to a linear fade except that step = (target - current) / numsteps; ... gain += step; becomes step = powf(target / current, 1.0f / numsteps); ... gain *= step; where 'target' and 'current' are clamped to a lower bound that is greater than 0 (which makes no sense on a logarithmic scale). Consequently, the non-HRTF direct mixers do not do not feed into the click removal and pending click buffers, as this per-sample fading would do an adequate job of stopping clicks and pops caused by extreme gain changes. These buffers should be removed shortly.
* Move the step counter and moving flag to DirectParamsChris Robinson2014-03-231-2/+4
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* Store the HrtfState directly in the DirectParamsChris Robinson2014-03-231-3/+3
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* Move some HRTF and mixer structs to alu.hChris Robinson2014-03-221-6/+60
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* Store some source mixing parameters in the active source structChris Robinson2014-03-191-3/+3
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* Add min/max/clamp methods for doublesChris Robinson2013-11-271-0/+7
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* Use C99 inline in more placesChris Robinson2013-11-041-18/+18
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* Use a helper macro for pi*2Chris Robinson2013-10-081-0/+1
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* Use helper macros to convert between degrees and radiansChris Robinson2013-10-081-2/+5
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* Use a macro for the silence thresholdChris Robinson2013-10-061-0/+2
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* Use helpers to set channel gain arraysChris Robinson2013-10-031-1/+18
| | | | Also avoid unnecessary clearing.
* Use C99's inline instead of __inlineChris Robinson2013-05-281-17/+17
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* Use restrict instead of RESTRICTChris Robinson2013-05-221-3/+3
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* Lock the device before calling aluHandleDisconnectChris Robinson2012-12-021-0/+1
| | | | | | PulseAudio causes an assert if being relocked inside a callback on the worker thread, where aluHandleDisconnect is called. We can assume it's already locked there, so just make sure the device is locked before being calling it.
* Move some math functions to where they're usedChris Robinson2012-10-251-26/+0
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* Remove the float math wrapper functionsChris Robinson2012-10-251-55/+0
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