Commit message (Collapse) | Author | Age | Files | Lines | |
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* | Use a separate LFO offset in the chorus effect | Chris Robinson | 2017-12-15 | 1 | -10/+19 |
| | | | | | Given that the LFO range is not a power-of-two, it won't correctly wrap on overflow. | ||||
* | Use linear interpolation for the chorus delay output | Chris Robinson | 2017-12-15 | 1 | -37/+39 |
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* | Add an option to ignore the app's speed of sound for reverb decay | Chris Robinson | 2017-09-22 | 1 | -2/+2 |
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* | Use the app-specified speed of sound for reverb decay | Chris Robinson | 2017-09-21 | 1 | -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 method | Chris Robinson | 2017-09-21 | 10 | -41/+50 |
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* | Use sqrtf for single-precision square roots | Chris Robinson | 2017-06-29 | 1 | -1/+1 |
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* | Remove the fastf2u conversion function | Chris Robinson | 2017-06-27 | 3 | -9/+9 |
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* | Calculate chorus and flanger mod delays separately from feedback | Chris Robinson | 2017-06-07 | 2 | -135/+118 |
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* | Make the late lines' delay the delay average for modulation | Chris Robinson | 2017-06-07 | 1 | -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. | ||||
* | Fix handling chorus and flanger LFO displacement offset | Chris Robinson | 2017-05-26 | 2 | -2/+8 |
| | | | | | The phase offset is modulo-wrapped rather than masked, so it's best to avoid negative offsets. | ||||
* | Properly handle the chorus and flanger LFOs | Chris Robinson | 2017-05-26 | 2 | -24/+30 |
| | | | | | The effects' specified delay is the average delay time, meaning the delay offset should move between -n and +n relative to the delay, where n <= delay. | ||||
* | Add a method to copy a filter's coefficients | Chris Robinson | 2017-05-21 | 2 | -39/+6 |
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* | Avoid unnecessary doubles | Chris Robinson | 2017-05-21 | 1 | -2/+2 |
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* | Use a macro to specify the decay target gain | Chris Robinson | 2017-05-21 | 1 | -2/+2 |
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* | Reduce the main reverb filter gain limit to match the rest | Chris Robinson | 2017-05-16 | 1 | -4/+4 |
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* | Make reverb delay line structs use interleaved floats | Chris Robinson | 2017-05-15 | 1 | -68/+73 |
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* | Make the reverb's early and late feedback lines interleaved | Chris Robinson | 2017-05-14 | 1 | -42/+47 |
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* | Update a couple comments about the reverb modulation | Chris Robinson | 2017-05-13 | 1 | -12/+13 |
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* | Replace 4 separate all-passes with one vector all-pass | Chris Robinson | 2017-05-13 | 1 | -53/+47 |
| | | | | | | Each 4 related all-passes now share a structure with one delay line, which uses an interleaved sample history. Also fixes some potential rounding problems for delay lines with interleaved samples. | ||||
* | Apply reverb modulation to the late feedback lines | Chris Robinson | 2017-05-12 | 1 | -104/+71 |
| | | | | | | | | This seems to be more in-line with the intended behavior, to allow build-up and overlap within the reverb decay, rather than a pitch-shift on input. Unfortunately there's no readily available implementation of this reverb model that includes modulation to compare with, so a low depth coefficient is used to keep it very subtle. | ||||
* | Remove the 0.5 gain on the reverb output | Chris Robinson | 2017-05-11 | 1 | -6/+2 |
| | | | | | Turns out to be unnecessary, as it reduced the volume below what other reverb implementations provide with the same presets. | ||||
* | Restore the previous reverb B2A and A2B matrices | Chris Robinson | 2017-05-11 | 1 | -12/+12 |
| | | | | Also, untranspose the A2B matrix. | ||||
* | Update reverb conversion matrices and output gain | Chris Robinson | 2017-05-08 | 1 | -10/+12 |
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* | Add a slight bit of attenuation to the reverb output | Chris Robinson | 2017-05-02 | 1 | -2/+4 |
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* | Change some ALuint parameters to ALsizei | Chris Robinson | 2017-05-02 | 10 | -75/+74 |
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* | Don't do more reverb samples than there are to fade. | Chris Robinson | 2017-04-28 | 1 | -2/+6 |
| | | | | This avoids having to clamp the fade value when incrementing it. | ||||
* | Combine mostly duplicate functions | Chris Robinson | 2017-04-27 | 1 | -121/+50 |
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* | Properly update the delay line offsets when fading is done | Chris Robinson | 2017-04-27 | 1 | -28/+44 |
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* | Reverb code update | Chris Robinson | 2017-04-26 | 1 | -824/+1213 |
| | | | | | | | | | | | | | | This update modifies the reverb in numerous ways. The 3-series, 4-parallel all-pass is replaced with a Gerzon vector all-pass. The vector all-pass is also applied to the early reflections, to help with the initial diffusion in the late reverb. The late reverb filter and feedback attenuation has been replaced with a dual first-order equalization filter, which controls the low and high frequencies with individual low-pass/high-shelf and high-pass/low-shelf filters with gain control. Additionally, delay lines now have the ability to blend the output between two offsets, to help smooth out the transition for when the delay changes (without such, it could result in undesirable clicks and pops). | ||||
* | Use ALsizei for more index lookups | Chris Robinson | 2017-03-24 | 1 | -15/+15 |
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* | Properly calculate the echo damping | Chris Robinson | 2017-03-24 | 1 | -1/+1 |
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* | Increase the filter slope to -12dB/octave | Chris Robinson | 2017-03-16 | 1 | -1/+1 |
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* | Remove CalcXYZCoeffs and inline CalcAngleCoeffs | Chris Robinson | 2017-02-23 | 4 | -10/+12 |
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* | Limit filter gains to -24dB | Chris Robinson | 2017-02-22 | 3 | -7/+7 |
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* | Fix late reverb low-pass filtering | Chris Robinson | 2017-01-29 | 1 | -1/+1 |
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* | Use an all-pass series on each late reverb line | Chris Robinson | 2017-01-28 | 1 | -261/+396 |
| | | | | | | | | | This attempts to improve the smoothness of the late reverb decay by passing each line through multiple all-pass filters. Some work is still needed to work better in high-density and not-so-high-diffusion environments. This also removes the decay from the early reflections, since it's no longer continuous feedback. | ||||
* | Avoid writing to the same buffer that's read from | Chris Robinson | 2016-12-23 | 1 | -36/+36 |
| | | | | Also clean up comment formatting a bit. | ||||
* | Reorder filter coefficients | Chris Robinson | 2016-12-21 | 2 | -10/+10 |
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* | Round the early and late delay tap sample offsets | Chris Robinson | 2016-10-26 | 1 | -4/+4 |
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* | Restore a comment that was accidentally deleted | Chris Robinson | 2016-10-10 | 1 | -0/+9 |
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* | Better sort the main delay line taps | Chris Robinson | 2016-10-06 | 1 | -60/+39 |
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* | Decorrelate the early reflection inputs | Chris Robinson | 2016-10-06 | 1 | -10/+19 |
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* | Pass current and target gains directly for mixing | Chris Robinson | 2016-10-05 | 1 | -45/+12 |
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* | Update a comment about using row mixers | Chris Robinson | 2016-10-05 | 1 | -3/+1 |
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* | Better pack the late reverb low- and all-pass variables | Chris Robinson | 2016-10-05 | 1 | -21/+25 |
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* | Use the row mixer functions for the B-to-A-Format conversion | Chris Robinson | 2016-10-04 | 1 | -16/+10 |
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* | Enhance reverb using B-Format processing | Chris Robinson | 2016-10-03 | 1 | -405/+386 |
| | | | | | | | | | | | | Technically it uses A-Format processing from the B-Format input and output. But this attempts to provide better spatial definition to the reverberation so that it can be used in a more generic fashion, allowing it to be decoded as any other B-Format signal to whatever output is needed, and also allowing for a bit of height information when the output is capable of such. There may still be some kinks to work out, such as properly decorrelating the early reflection taps and tweaking the late reverb density. But it seems to be a good enough start. | ||||
* | Fix EAX reverb effect output for HRTF and UHJ | Chris Robinson | 2016-09-13 | 1 | -2/+2 |
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* | Call ALfilterState_processC directly | Chris Robinson | 2016-09-12 | 2 | -5/+0 |
| | | | | | | It's the only implementation currently, so there's no point to having it stored as a function pointer in the filter struct. Even if there were SIMD versions, it'd be a global selection, not per-instance. | ||||
* | Write to the correct outputs for extra reverb channels | Chris Robinson | 2016-09-11 | 1 | -2/+2 |
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