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path: root/Alc/effects/reverb.c
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* Avoid AL prefix on internal effect state factory typesChris Robinson2018-02-281-8/+8
| | | | Also avoid using the generic V/V0 macros for them
* Combine the vector reverse and partial scatter where they're togetherChris Robinson2018-02-181-21/+20
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* Remove unused reverb modulation codeChris Robinson2018-02-101-107/+2
| | | | | | Still unsure how to handle reverb modulation without some kind of reference output, so remove the related functions to not waste CPU time. It'll remain in the Git history should it ever need to be resurrected.
* Attempt to improve the reverb panning vectorsChris Robinson2018-02-081-20/+42
| | | | | This should now retain the original orientation of the soundfield and merely focus on the panning vector direction, as intended.
* Alter tha curve of the density-related delay scaleChris Robinson2018-02-081-22/+33
| | | | | | | | | | | | | | | | | | | The delay scale is roughly linear with respect to room size, however the density is not linear with room size. The density is calculated by taking the room size cubed, then normalized by some factor. Unnormalizing the density and taking the cube root restores the original room size to use as a delay scale. The patch also alters the delay and all-pass line lengths to be based on a 1 meter room size, so the the room size recovered from the density acts as a direct multiple for the desired target length. Note that the room scale range is unchanged (5m to 50m), so the minimum and maximum delays are the same. It should also be noted that 50m may not be the correct room size for a density value of 1. A density value of 1 corresponds to an environment size of roughly 2.52m when converted from EAX (DENSITY_SCALE should be 16 rather than 125000), but sizes that low result in undesirable resonance in the feedback, indicating other changes are necessary for that to work.
* Revert "Don't fade the all-pass delay changes"Chris Robinson2018-02-041-28/+48
| | | | This reverts commit 799dfb732b4f49198d72649e86955ea82f45f229.
* Construct error messages using parameterized valuesChris Robinson2018-01-251-44/+48
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* Provide messages for the remaining AL errorsChris Robinson2018-01-241-62/+50
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* Use a global RowMixerFuncChris Robinson2018-01-161-10/+0
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* Don't fade the all-pass delay changesChris Robinson2018-01-151-48/+28
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* Fix a comment regarding the T60 filter coefficientsChris Robinson2018-01-151-3/+2
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* Simplify preparing the T60 filtersChris Robinson2018-01-151-50/+11
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* Use separate input and outputs for the T60 and allpass processingChris Robinson2018-01-151-21/+22
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* Use a proper type for the T60 decay filterChris Robinson2018-01-151-25/+22
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* Separate some anonymous structs into defined structsChris Robinson2018-01-141-107/+109
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* Remove a redundant lower value clampChris Robinson2018-01-141-4/+2
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* Don't check the modulation depth for changes multiple timesChris Robinson2018-01-141-7/+8
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* Use separate arrays for the T60 filter statesChris Robinson2018-01-141-9/+10
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* Use a macro for the number of reverb linesChris Robinson2018-01-141-82/+86
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* Only pass the needed parameters to LateT60FilterChris Robinson2018-01-141-28/+21
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* Pre-apply the T60 filter midgain to the HF filter coeffsChris Robinson2018-01-131-38/+30
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* Replace some freq_mult variable names with f0normChris Robinson2018-01-131-6/+6
| | | | | | The latter is a bit more descriptive as f0 is often used to denote the reference frequency of a filter, so f0norm indicates the normalized reference frequency (ref_freq / sample_rate).
* Avoid using macros to access anonymous structuresChris Robinson2018-01-111-2/+4
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* Remove standard reverb specific processing functionsChris Robinson2018-01-071-114/+35
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* Move the UNEXPECTED macro to the main header and rename itChris Robinson2018-01-071-6/+1
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* Use separate outputs for the vector reverse and scatter functionsChris Robinson2018-01-041-29/+23
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* Avoid the modulated output tap for late reverbChris Robinson2018-01-041-28/+14
| | | | | | And fix the output filtering. The modulation code is still there since it's (probably) technically correct, but the interaction with the feedback loop and filtering on the output caused improper behavior which needs to be sorted out.
* Don't offset the reverb modulation sinusChris Robinson2018-01-021-1/+1
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* Use a float literal for float mathChris Robinson2018-01-021-2/+2
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* Fix up a commentChris Robinson2017-12-241-7/+5
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* Don't make a pass-through macro for FadedDelayLineOutChris Robinson2017-12-241-8/+7
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* Use linear interpolation for reverb modulationChris Robinson2017-12-241-76/+130
| | | | | | The core LateReverb_* functions are explicitly written out now, since the tapping and blending done by the Faded version is a bit more complex and it's not so easy to ensure proper optimizing on the Unfaded version.
* Fade between depths in the reverb modulatorChris Robinson2017-12-241-48/+26
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* Use a separate unmodulated feedback tap for reverbChris Robinson2017-12-241-18/+20
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* Clarify some reverb valuesChris Robinson2017-12-231-7/+6
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* Reorder some global variablesChris Robinson2017-12-231-188/+188
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* Cleanup reverb modulation scalingChris Robinson2017-12-231-14/+17
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* Make MixSamples non-static globalChris Robinson2017-12-171-2/+0
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* Add an option to ignore the app's speed of sound for reverb decayChris Robinson2017-09-221-2/+2
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* Use the app-specified speed of sound for reverb decayChris Robinson2017-09-211-4/+7
| | | | | Specifically, the initial reverb decay as determined by the source distance, and the reverb decayhf limit from air absorption.
* Pass the context to the auxiliary effect update methodChris Robinson2017-09-211-2/+3
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* Use sqrtf for single-precision square rootsChris Robinson2017-06-291-1/+1
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* Remove the fastf2u conversion functionChris Robinson2017-06-271-7/+7
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* Make the late lines' delay the delay average for modulationChris Robinson2017-06-071-59/+36
| | | | | | | | | Similar to the recent chorus and flanger changes, the modulation delay now swings between -n to +n, where n is less than the delay length. This brings up a slight issue with the linear interpolation, as modff doesn't produce the correct fraction value for interpolation (it's inverted, with 0 being closer to the next sample and 1 being closer to the base). So it's using nearest interpolation for now.
* Add a method to copy a filter's coefficientsChris Robinson2017-05-211-11/+2
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* Avoid unnecessary doublesChris Robinson2017-05-211-2/+2
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* Use a macro to specify the decay target gainChris Robinson2017-05-211-2/+2
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* Reduce the main reverb filter gain limit to match the restChris Robinson2017-05-161-4/+4
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* Make reverb delay line structs use interleaved floatsChris Robinson2017-05-151-68/+73
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* Make the reverb's early and late feedback lines interleavedChris Robinson2017-05-141-42/+47
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