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* Append the OpenSL capture device to the appropriate listChris Robinson2018-08-231-1/+1
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* Use second-order biquad filters for the reverb's T60 decayChris Robinson2018-08-121-309/+18
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* Store the reverb state offset locally while processingChris Robinson2018-08-111-16/+14
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* Correctly apply reverb coefficient fading over the entire fade lengthChris Robinson2018-08-091-23/+25
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* Crossfade the early reflections delay coefficientsChris Robinson2018-08-081-30/+50
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* Apply late reverb density gain on late reverb inputChris Robinson2018-08-081-15/+30
| | | | | | | | | | | This also uses gain stepping for changes in density or decay, so that the late reverb tap fades smoothly between delay offsets and density gain levels simultaneously. Now with these changes, it's preferrable to apply density gain adjustments on late reverb input instead of output so that samples currently in the feedback loop won't see a sudden increase or decrease in amplitude. A similar change can probably be made for the early reflection delays to further smooth out delay changes.
* Specify both scale values to FadedDelayLineOutChris Robinson2018-08-081-9/+17
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* Use the correct input channel for the compressor effectChris Robinson2018-08-081-1/+1
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* Properly set early reflection all-pass coefficientChris Robinson2018-08-061-3/+7
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* Make the all-pass coefficient part of the all-pass structureChris Robinson2018-08-061-22/+17
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* Prevent the autowah filter frequency from reaching nyquistChris Robinson2018-08-061-1/+1
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* Apply the vector all-pass separate on uninterleaved samplesChris Robinson2018-08-051-159/+310
| | | | | | | | | | With the vector all-pass applied in a self-contained function, the individual steps of the early and late reverb stages can be better optimized with tighter loops. That allows for more data to be held local, resulting in less thrashing from reloading the same values multiple times. There is room for further improvement, depending on the length of the early delay lines and all-pass delay lines allowing for bulk reads.
* Apply the reverb's T60 filter in groups of samplesChris Robinson2018-08-051-54/+78
| | | | | | | The late reverb line lengths are long enough to ensure a single process loop won't rely on reading samples it wrote in the same call. So we can safely read in all samples we need from the feedback buffer up front, then more efficiently filter them.
* Mix reverb samples right after generating themChris Robinson2018-08-041-31/+36
| | | | | | | Instead of generating both the early and late reverb samples first, then mixing them both to output, this now generates and mixes the early reflections then generates and mixes the late reverb. There's no reason to hold both at the same time so this reduces the amount of temporary storage needed.
* Scale the reverb decay weights to cover up to 20khzChris Robinson2018-08-031-11/+16
| | | | | | | Otherwise, using the device's maximum frequency will cause the weighting factors to shift for different sample rates, irrespective of the content being processed. 20khz is the maximum allowed reference frequency, and also acts as the upper limit of human hearing.
* Trace the NFC reference distance when setChris Robinson2018-08-031-0/+1
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* Apply late reverb density gain adjustment on outputChris Robinson2018-08-011-22/+17
| | | | | | | | Because density/late line length changes start affecting late reverb output right away, with samples that are still going through feedback decay and not just new input samples, it makes more sense to correct for it on output instead of input. This has the additional benefit of working with the output mixer's gain fading, avoiding discontinuities from significant density gain changes.
* Fix late reverb density gain blend weightsChris Robinson2018-08-011-4/+4
| | | | | | | | | | | | Now it only accounts for the representable frequency range (0.5 normalized, or 0...pi radians instead of tau). Previously, the bulk of the weighting factors was given to the HF decay (nearly 90%, given a 44.1khz sample rate and the default 5khz reference), with low- and mid-frequency decays splitting the remaining 10%. Now it's closer to 75%, matching the range of representable frequencies above the reference. This could probably be improved further due to human hearing being less sensitive to higher frequencies, but that is much more complicated.
* Use just the omnidirectional response for the compressor effectChris Robinson2018-07-291-64/+58
| | | | | | | | This is not the output compressor/limiter, but the EFX effect. Consequently, it simply compresses the dynamic range around 1.0 (boosting samples below it by up to double, reducing samples above it by as much as half). This is not intended to prevent clipping on the output, but to instead reduce the range between quiet sounds and loud sounds.
* Inline the autowah peaking filter processingChris Robinson2018-07-281-39/+74
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* Mark a couple parameters as unusedChris Robinson2018-07-251-1/+1
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* EFX: Enable 3D processingRaulshc2018-07-251-20/+17
| | | Use channel 0 envelope for calculate the frequency in all channels.
* EFX: Add 3D processing for autowahRaulshc2018-07-251-36/+55
| | | Add 3D processing code. It can be activated at compilation time.
* EFX: Autowah implementationRaulshc2018-07-252-2/+271
| | | Add autowah effect using biquad peaking filter and envelope follower
* Limit the normalized filter frequency to under halfChris Robinson2018-07-211-1/+1
| | | | | Nearing half, weird things can start happening with the filters' generated sine and cosine values.
* Improve handling of 0hz ring modulator frequencyChris Robinson2018-07-211-5/+13
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* Start an extension to allow source filter gains greater than 1Chris Robinson2018-07-151-0/+1
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* Prefer sndio over OSS when both are enabledTobias Kortkamp2018-06-221-3/+3
| | | | Signed-off-by: Tobias Kortkamp <[email protected]>
* Properly get the full executable and pathname on FreeBSDChris Robinson2018-06-081-4/+4
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* Always prepare the ALSA PCM handle before starting captureChris Robinson2018-06-081-2/+9
| | | | | | Draining the ALSA device via stopping puts it into a setup state, which requires re-preparing before playback can start again. Preparing it prior to the first start seems to cause no harm, so just always do it before starting.
* Stop capture devices while closingkdhp2018-06-081-0/+6
| | | | | | | In 'alcCaptureCloseDevice', check if the capture device is running and stop it if necessary. This fixes the case where the device data is deallocated while a background thread is still running (Issue #199)
* Clamp the maximum normalized reference frequency tooChris Robinson2018-06-031-1/+1
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* Use a higher normalized frequency limit for the ring modulatorChris Robinson2018-06-021-1/+1
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* Use the biquad high-pass in the ring modulatorChris Robinson2018-05-311-10/+6
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* Slightly simplify the modulator square wave generatorChris Robinson2018-05-311-3/+3
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* EFX: Ring modulator fixesRaulshc2018-05-311-3/+3
| | | Change from unipolar to bipolar carrier signal in the Ring modulator effect.
* Avoid extra sample copies and storage in the modulator effectChris Robinson2018-05-291-7/+6
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* Use fastf2i instead of manual rounding in another placeChris Robinson2018-05-291-1/+2
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* Use a macro to handle common case formattingChris Robinson2018-05-271-21/+10
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* Avoid setting unnecessary variablesChris Robinson2018-05-261-10/+0
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* Don't hardcode the max channels for HRTF B-Format decodingChris Robinson2018-05-263-8/+5
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* Improve formatting of the hilbert functionChris Robinson2018-05-241-1/+1
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* Clamp the dither depth between 2 and 20 bitsChris Robinson2018-05-231-3/+6
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* Add some LIKELY and ASSUME statementsChris Robinson2018-05-231-1/+8
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* Fix a function comment about a return valueChris Robinson2018-05-231-3/+1
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* Remove unused functionChris Robinson2018-05-231-10/+0
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* Use a tighter loop to handle the frequency shifter's fifoChris Robinson2018-05-221-33/+39
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* Use fixed point for the frequency shifter's phaseChris Robinson2018-05-221-16/+14
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* Fix a couple property variable namesChris Robinson2018-05-221-7/+7
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* Improve the frequency shifter output accum handlingChris Robinson2018-05-221-9/+8
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