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* Separate calculating ambisonic coefficients from the panning gainsChris Robinson2016-01-2511-35/+47
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* Reorder filterstate propertiesChris Robinson2016-01-232-19/+18
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* Don't overwrite the output with the dedicated effectChris Robinson2015-12-121-1/+1
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* Simplify reverb panning rotationsChris Robinson2015-11-071-4/+4
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* Pan each early/late delay line separatelyChris Robinson2015-11-051-56/+70
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* Remove an unneeded memset that was overwriting memoryChris Robinson2015-11-051-2/+0
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* Pass in the Q parameter for setting the filter parametersChris Robinson2015-11-014-21/+36
| | | | Also better handle the peaking filter gain.
* Use modff to split the modulation delay componentsChris Robinson2015-10-311-4/+3
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* Mix reverb to output in the inner loopChris Robinson2015-10-291-41/+39
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* Include the echo's attenuation of the late reverb in the late reverb gainChris Robinson2015-10-291-24/+29
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* Do up to 256 samples per reverb inner loop iterationChris Robinson2015-10-291-2/+3
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* Always update all reverb propertiesChris Robinson2015-10-281-68/+44
| | | | | | The EAX-only effect properties will be set to compatible defaults when standard reverb is set, and the EAX-only effects will be skipped during sample processing.
* Do multiple samples at once in each reverb componentChris Robinson2015-10-281-245/+267
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* Rename F_2PI to F_TAUChris Robinson2015-09-135-7/+7
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* Do up to 256 samples at a time with multi-step loopsChris Robinson2014-12-188-23/+21
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* Improve ambient gain calculationsChris Robinson2014-11-251-4/+4
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* Fix reverb panning directionChris Robinson2014-11-251-2/+2
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* Interpolate directional reverb gains with ambient based on vector lengthChris Robinson2014-11-151-4/+23
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* Pas the output device channel count to ALeffectState::processChris Robinson2014-11-0711-48/+45
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* Use a separate macro for the max output channel countChris Robinson2014-11-0710-25/+25
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* Set gains using the device channel indexChris Robinson2014-11-051-4/+6
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* Use a method to set omni-directional channel gainsChris Robinson2014-11-046-19/+9
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* Avoid taking the square-root of the ambient gainChris Robinson2014-10-116-40/+20
| | | | | | Although it is more correct for preserving the apparent volume, the ambisonics- based panning does not work on the same power scale, making it louder by comparison.
* Add a helper to search for a channel index by nameChris Robinson2014-10-021-11/+7
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* Make ComputeAngleGains use ComputeDirectionalGainsChris Robinson2014-10-025-44/+74
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* Combine some fields into a structChris Robinson2014-09-106-7/+7
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* Remove a couple unnecessary typedefsChris Robinson2014-08-241-2/+2
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* Update COPYING to the latest ↵François Cami2014-08-1810-20/+20
| | | | https://www.gnu.org/licenses/old-licenses/lgpl-2.0.txt to fix the FSF' address Fix the FSF' address in the source
* Use logarithmic adjustment for the gain in the autowah effectChris Robinson2014-05-171-7/+7
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* Use an enum for the chorus and flanger waveformsChris Robinson2014-05-142-16/+54
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* Make LOWPASSFREQREF a float valueChris Robinson2014-05-112-2/+2
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* Use a void* for the effect state Delete method paramChris Robinson2014-03-2111-54/+34
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* Change the autowah to have a cutoff range of 20Hz to 20KHzChris Robinson2014-02-071-1/+1
| | | | | | This is the same as allowed in the SF2 spec for its filter cutoff generator, which can be used for a wah effect in MIDI. It makes sense to use a similar range here.
* Fix some typesChris Robinson2014-02-072-4/+4
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* Use a different autowah filter rangeChris Robinson2013-12-201-1/+1
| | | | Still just guesses, but this seems to avoid math problems with low peak gains.
* Fix numeric constant type (double->float)Chris Robinson2013-12-081-2/+2
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* Use one long buffer for the chorus and flanger delay linesChris Robinson2013-11-102-38/+26
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* Rename lfo_coeff to lfo_scaleChris Robinson2013-11-082-16/+16
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* Use integer modulo for chorus and flangerChris Robinson2013-11-082-58/+62
| | | | Also simplify LFO coefficient calculations.
* Only use the inner loop for the chorus/flanger template methodChris Robinson2013-11-082-112/+114
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* Redo the autowah effectChris Robinson2013-11-071-71/+71
| | | | | | | | It's now a low-pass filter with a cutoff that varies according to the input signal amplitude. There remains issues with apparent feedback in the resonant frequency with high resonance values. The actual cutoff range for the filter is also a guess.
* Reimplement the compressor to work per-sampleChris Robinson2013-11-061-37/+51
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* Use a helper macro for pi*2Chris Robinson2013-10-086-15/+12
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* Use inline initialization for effect state factory vtablesChris Robinson2013-10-0711-145/+66
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* Constify the effect parameter of effect gettersChris Robinson2013-10-0711-56/+49
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* Use the UNUSED macro in the effectsChris Robinson2013-10-0711-110/+60
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* Use a macro for the silence thresholdChris Robinson2013-10-0610-21/+22
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* Implement the Compressor effectChris Robinson2013-10-031-0/+216
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* Convert autowah.c line-endingsChris Robinson2013-10-031-284/+284
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* Use helpers to set channel gain arraysChris Robinson2013-10-039-81/+17
| | | | Also avoid unnecessary clearing.
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