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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.
* Implement the Autowah effect.Chris Robinson2013-10-031-0/+293
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* Avoid a divide-by-0 when distortion edge is 1Chris Robinson2013-09-041-0/+1
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* Implement master high-pass filter for EAXREVERBChris Robinson2013-07-211-9/+24
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* Remove an unused functionChris Robinson2013-07-211-7/+0
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* Fix input and feedback in the echo effectChris Robinson2013-06-101-3/+3
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* Use ALfilterState for the distortion effect filtersChris Robinson2013-06-061-70/+14
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* Silence some clang warningsChris Robinson2013-06-051-1/+8
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* Avoid an unnecessary loopChris Robinson2013-05-291-36/+19
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* Fix up the naming convention of effect methodsChris Robinson2013-05-299-148/+148
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* Use C99's inline instead of __inlineChris Robinson2013-05-284-28/+28
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* Use the high-shelf filter in place of low-passChris Robinson2013-05-282-2/+2
| | | | | They effectively both work to lower (or raise) high frequencies. However, the high-shelf performs better when gain=1.
* Simplify and fix some filter gain calculationsChris Robinson2013-05-271-5/+5
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* Use ALfilterState for the modulator high pass filterChris Robinson2013-05-271-25/+12
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* Use ALfilterState for the master echo and reverb filtersChris Robinson2013-05-273-21/+24
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* Move ALEQFilter to alFilter.c/h and rename it to ALfilterStateChris Robinson2013-05-271-107/+10
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* Cleanup the ALeffectStateFactory_create methodsChris Robinson2013-05-279-9/+18
| | | | | Get rid of the ALeffectStateFactory_create macro, and use the VCALL_NOARGS helper (requires adding the 'this' factory parameter).
* Rename a variableChris Robinson2013-05-261-8/+8
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* Pre-apply the a[0] EQ filter coefficientChris Robinson2013-05-261-5/+11
| | | | Saves 20 fp divisions per sample
* Use the helper macros in more placesChris Robinson2013-05-267-343/+237
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