aboutsummaryrefslogtreecommitdiffstats
path: root/Alc
Commit message (Collapse)AuthorAgeFilesLines
* Use the correct value for MAX_AMBI2D_CHANNELSChris Robinson2019-03-101-1/+1
|
* Don't copy old coeffs in MixHrtfBlendBaseChris Robinson2019-03-101-2/+2
|
* Avoid an extra level of indentationChris Robinson2019-03-101-38/+50
|
* Don't directly use a buffer for updating source parametersChris Robinson2019-03-102-25/+15
|
* Avoid excessive transformations of the source positionChris Robinson2019-03-101-44/+60
|
* Add a method to apply an HF scale without band-splittingChris Robinson2019-03-104-25/+52
|
* Fix for MSVC decaying arrays to pointers with ?:Chris Robinson2019-03-091-5/+5
|
* Fade out voices that end normallyChris Robinson2019-03-091-140/+180
| | | | | | Sometimes a sound may end with non-0 amplitude, particularly if a buffer queue underruns. This helps avoid clicks and pops for sources that don't already end in silence.
* Play dummy samples and force a fade out on stopping voicesChris Robinson2019-03-091-24/+42
|
* Add a Stopping state for voicesChris Robinson2019-03-093-26/+58
| | | | | | This currently doesn't do much, except have the mixer progress it to Stopped. It's valid to have without a source or buffers, and in the future will allow fading out when a source is paused or stopped.
* Clear the voice's buffer when detaching from sourceChris Robinson2019-03-091-0/+4
|
* Pass a reference to function for a template parameterChris Robinson2019-03-031-3/+3
|
* Use more specific names for temp buffer storageChris Robinson2019-03-021-18/+10
|
* Reverse the HRTF field arrayChris Robinson2019-02-273-15/+24
| | | | | | Most often a sound's distance will be beyond the farthest field measurement, so It's more efficient to have the farthest field first and avoid looping through the whole field array for them.
* Combine the reverb output mixes into a single callChris Robinson2019-02-251-28/+38
|
* Make sure the reverb fading completesChris Robinson2019-02-251-1/+1
| | | | | The processing loop doesn't depend on being aligned anymore, so it won't get stuck when only less than 4 samples can be done in a non-final update.
* Convert the device frequency to float just onceChris Robinson2019-02-241-9/+7
|
* Make sure the voice's direct buffer is always setChris Robinson2019-02-241-7/+5
|
* Remove a duplicate index arrayChris Robinson2019-02-241-2/+2
|
* Reduce BUFFERSIZE to match the default period sizeChris Robinson2019-02-241-9/+9
| | | | | Also adds a bit more space to the temp source data buffer, to avoid needing to loop on matching sample rates.
* Rework reverb A/B-Format conversion mixingChris Robinson2019-02-241-99/+77
| | | | | This should help improve performance using the optimized mixers, and fewer passes on the transforms, though at the cost of more memory.
* Change some functions to proper methodsChris Robinson2019-02-231-171/+171
|
* Avoid multiple int-to-float conversionsChris Robinson2019-02-231-12/+17
|
* Constify some parameters and remove an explicit loopChris Robinson2019-02-232-10/+23
|
* Remove the FOAOut mixing buffer and associated post-processesChris Robinson2019-02-226-289/+11
|
* Remove the unused FOAOut EffectTargetChris Robinson2019-02-221-2/+2
|
* Apply ambisonic upsampling on reverb output as neededChris Robinson2019-02-221-37/+110
| | | | | | | | | | This isn't the greatest thing since it splits the A-to-B-Format transform from the panning transform. The A-to-B and HF scale mixes are also not as optimal as they could be, since they can't use the main mixer functions (wrong buffer line length). It does, however, get rid of the final use of the FOAOut buffer, so the upsampling post-process is no longer needed.
* Ensure reverb fading doesn't end with less than 4 samplesChris Robinson2019-02-221-1/+1
|
* Avoid some unnecessary local variablesChris Robinson2019-02-221-13/+12
|
* Combine reverb transform matrices one column at a timeChris Robinson2019-02-211-18/+23
|
* Mix B-Format sources directly to the dry bufferChris Robinson2019-02-212-21/+34
| | | | Now the only thing that utilizes FOAOut is reverb output.
* Remove RESTRICT from the bandsplitter process methodChris Robinson2019-02-212-2/+2
| | | | | The compiler can see there's no aliasing with the local variables, and the input/output buffers are handled sequentially one element at a time anyway.
* Allow processing some effects in higher order ambisonicsChris Robinson2019-02-216-17/+17
| | | | | | Reverb notably is still only first-order (any higher order channels are dropped, and it writes to FOAOut). But others, like the equalizer, work on all available channels.
* Add helpers to get the channel count from an ambisonic orderChris Robinson2019-02-213-16/+20
|
* Fix unused parameter warningChris Robinson2019-02-211-1/+1
|
* Get rid of the MAX_EFFECT_CHANNELS macroChris Robinson2019-02-214-40/+39
|
* Make sure the B2A matrix has enough values for the input countChris Robinson2019-02-211-6/+6
|
* Remove some now-unnecessary ReverbState fieldsChris Robinson2019-02-211-26/+1
|
* Pass the number of input channels to EffectState::processChris Robinson2019-02-2113-107/+104
|
* Ensure the device's mAmbiOrder is always set appropriatelyChris Robinson2019-02-211-8/+23
| | | | The Dry target is always ambisonic, so set its order correctly.
* Store effect slots in groups of 64Chris Robinson2019-02-202-18/+41
| | | | | Now that their wet buffers are allocated dynamically, the ALeffectslot object itself is rather small.
* Allocate the effect slot wet buffer dynamicallyChris Robinson2019-02-203-27/+31
|
* Partially handle non-periphonic reverb inputChris Robinson2019-02-191-3/+26
|
* Use the right macro for the number of reverb panning gainsChris Robinson2019-02-191-2/+2
|
* Rename MAX_AMBI_COEFFS and MAX_AMBI2D_COEFFSChris Robinson2019-02-1914-56/+56
|
* Reduce some indentingChris Robinson2019-02-191-27/+25
|
* Apply the upsampler's all-pass when mixing the band-split samplesChris Robinson2019-02-191-10/+10
|
* Clean up some AmbiUpsampler initializationChris Robinson2019-02-191-17/+19
|
* Avoid a temp buffer for the distance compensation delayChris Robinson2019-02-171-23/+12
|
* Apply phase correction to the ambisonic decoder HRIRsChris Robinson2019-02-161-25/+60
| | | | | | This preserves the original phase of the HRIR frequencies for decoding the ambisonic signal. This should help prevent extra coloration from the band- splitter used to scale the HF response.
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-29 22:01:48 +0000